| 00:01.32 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 00:05.19 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 00:31.12 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 00:31.59 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 00:41.34 | atan2 | drmessano, "<drmessano> atan, yeah.. it does exist and is supported by a lot of phones" oh really? =) what is this 'feature' called I would be looking for? |
| 00:41.40 | atan2 | I assume it would just enable speakerphone |
| 00:50.16 | *** join/#asterisk Dovid (42570825@gateway/web/freenode/ip.66.87.8.37) |
| 00:59.45 | ChannelZ | auto answer |
| 01:00.07 | ChannelZ | May or may not involve adding SIP headers to trigger it to do so |
| 01:00.59 | ChannelZ | SIPAddHeader(Call-Info: \;answer-after=0) works on a lot of Linksys/Cisco SPA phones |
| 01:02.22 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 01:03.18 | *** join/#asterisk candrews (~candrews@fsf/member/candrews) |
| 01:05.03 | candrews | chan_gtalk.c: Could not find recipient. |
| 01:05.05 | candrews | what does that mean? |
| 01:06.46 | atan | Interesting ChannelZ. Thank you! |
| 01:18.41 | *** join/#asterisk meatbun (~wafers4@cpe-98-155-139-88.hawaii.res.rr.com) |
| 01:18.52 | meatbun | is there a web interface to manage asterisk? |
| 01:19.40 | WIMPy | Several. |
| 01:20.50 | drmessano | Yes, some good, some like a bun full of meat |
| 01:21.09 | WIMPy | Which one is the good one? |
| 01:21.36 | *** join/#asterisk fazendeiro (~chatzilla@200.147.129.15) |
| 01:24.17 | [TK]D-Fender | ~toywy |
| 01:24.18 | infobot | extra, extra, read all about it, toywy is The one you write yourself. |
| 01:24.21 | drmessano | FreePBX for one |
| 01:24.26 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 01:24.53 | meatbun | any pictures? |
| 01:25.33 | [TK]D-Fender | meatbun: www.freepbx.org |
| 01:32.53 | candrews | Is there a max length for a context or connection's name? |
| 01:33.10 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 01:34.15 | [TK]D-Fender | candrews: Probably.how far do you really feel like pushing it? |
| 01:34.20 | *** join/#asterisk hudony (~chatzilla@modemcable102.166-201-24.mc.videotron.ca) |
| 01:34.25 | hudony | hi |
| 01:34.27 | candrews | 8 characters |
| 01:34.45 | [TK]D-Fender | candrews: I've seen over 20 |
| 01:34.51 | candrews | well, I have ~30 character names, and I'm seeing a log entry which is leading me to believe it might be limited to 8, but that seems crazy |
| 01:34.55 | hudony | I have a cisco 7940 phone and I just reset it to factory default. Now, I cannot unlock it anymore, tried cisco as password and **# |
| 01:35.00 | hudony | Any other idea anyone^ |
| 01:35.02 | [TK]D-Fender | candrews: I'm sure it can rather longer stiff if needed |
| 01:38.54 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
| 01:48.31 | *** join/#asterisk simplydrew (~simplydre@pool-96-238-59-82.prvdri.fios.verizon.net) |
| 01:54.22 | *** join/#asterisk will_ (~wfong@pdpc/supporter/professional/will) |
| 01:54.54 | will_ | Hello. I just installed AsteriskNow. I cant' seem to find the doc explaining this set up, just the Asterisk reference manual. |
| 01:55.40 | [TK]D-Fender | will_: What part of the setup? |
| 01:56.06 | will_ | hehe, What's the default username/password to sign in via the webbrowser :) |
| 01:56.38 | [TK]D-Fender | will_: GOOGLE can answer that in under a minute. |
| 01:56.54 | [TK]D-Fender | will_: and once you even get in, FreePBX is NOT supported here. |
| 01:56.57 | [TK]D-Fender | ~freepbx |
| 01:56.57 | 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 |
| 01:57.02 | will_ | ah sorry |
| 01:57.17 | will_ | I'm not exactly sure what I'm doing :D |
| 01:58.25 | will_ | I did find what I needed. thanks :) |
| 02:02.12 | *** join/#asterisk Queka (~ZiKi@108.25.145.232) |
| 02:03.38 | Queka | hello |
| 02:03.48 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 02:04.35 | ChannelZ | OH HAI! |
| 02:09.32 | *** join/#asterisk MatBoy (~MatBoy@wiljewelwetenhe.xs4all.nl) |
| 02:13.07 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 02:17.33 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 02:21.54 | *** join/#asterisk Pan3D (~Pan3D@63.208.160.190) |
| 02:23.59 | candrews | I have a SIP device in a context called "100-device" |
| 02:24.21 | candrews | the context (in extensions.conf) for 100-device simply says "include => candrewsintegralblue-google-out" |
| 02:24.31 | candrews | that context says: |
| 02:24.33 | candrews | [candrewsintegralblue-google-out] |
| 02:24.33 | candrews | ;append a 1 if necessary |
| 02:24.33 | candrews | exten => _NXXNXXXXXX,1,Set(CALLERID(dnid)=1${CALLERID(dnid)}) |
| 02:24.33 | candrews | exten => _NXXNXXXXXX,n,Goto(1${EXTEN},1) |
| 02:24.33 | candrews | ;do our real dialing |
| 02:24.34 | candrews | exten => _1XXXXXXXXXX,1,dial(gtalk/candrewsintegralblue/+${EXTEN}@voice.google.com,,) |
| 02:24.41 | candrews | Seems like it should work... |
| 02:24.56 | candrews | But I get a "cancel" tone when I try to dial a phone number |
| 02:24.59 | candrews | and this in the log: |
| 02:26.23 | candrews | [Nov 14 21:25:24] DEBUG[22215] devicestate.c: No provider found, checking channel drivers for Gtalk - +18002464411@voice.google.com |
| 02:26.23 | candrews | [Nov 14 21:25:24] DEBUG[29612] channel.c: Soft-Hanging up channel 'SIP/101-0000000c' |
| 02:26.25 | candrews | I have the full logs... |
| 02:26.30 | candrews | Any idea why it's not working? |
| 02:27.13 | *** join/#asterisk N3tw0rK (~N3tw0rK@74.197.192.192) |
| 02:27.22 | [TK]D-Fender | candrews: PASTEBIN from now on. And please include the COMPLETE call attempt |
| 02:27.24 | [TK]D-Fender | ~pb |
| 02:27.25 | 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. |
| 02:27.26 | [TK]D-Fender | ^^^ |
| 02:28.18 | candrews | here's the complete log: |
| 02:28.18 | candrews | http://pastebin.ca/1991791 |
| 02:28.19 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 02:28.21 | candrews | (with debug on) |
| 02:29.23 | candrews | gtalk.conf: http://pastebin.ca/1991795 |
| 02:29.57 | [TK]D-Fender | candrews: disable that debug, set verbose to 10, enable SIP DEBUG, and retry |
| 02:30.36 | candrews | jabber.conf: http://pastebin.ca/1991796 |
| 02:30.38 | candrews | okay, doing so now.. |
| 02:32.53 | candrews | [TK]D-Fender, when I unset debug, and set verbose=10 and enabled sip debug, there is no logging done when I try to place that call |
| 02:34.21 | [TK]D-Fender | candrews: There should be all sorts of things to pastebin..... |
| 02:35.33 | N3tw0rK | when using * as a 3rd party VM system (for CUCME) do i need to have the CUCME instance registered as a user for MWI to work? |
| 02:35.53 | candrews | Only one line, which doesn't seem worth pastebin, in /var/log/asterisk/messages: Nov 14 21:35:29] WARNING[30561] chan_sip.c: Address remapping activated in sip.conf but we're using IPv6, which doesn't need it. Please remove "localnet" and/or "externaddr" settings. |
| 02:36.31 | candrews | I don't think that's important - I have GV calling working earlier with that error message, and it didn't hurt anything. BTW, I had GV working for in and out earlier, but want to use 2 GV's to 2 extensions, and I can't get it to work. |
| 02:37.45 | [TK]D-Fender | candrews: You've clearly done something wrong as you HAVE a SIP call coming in and DOES hit dialplan. Now try AGAIN. |
| 02:38.31 | [TK]D-Fender | N3tw0rK: Normally yes |
| 02:38.58 | [TK]D-Fender | N3tw0rK: You could craft your own packet sender and poll for VM's yourself I suppose |
| 02:39.40 | [TK]D-Fender | N3tw0rK: Actually... that wouldn't need to b "registered", it would simply have to have an IP to contact |
| 02:39.45 | N3tw0rK | [TK]D-Fender: i had this working in 1.4 but cant seem to get it to work correctly and im lacking any debug on the cisco side |
| 02:39.47 | [TK]D-Fender | N3tw0rK: Which could be a fixed host. |
| 02:40.06 | [TK]D-Fender | N3tw0rK: Where does "cisco" factor into this? |
| 02:41.21 | N3tw0rK | * doesnt seem to be sending out the correct MWI to the CUCME server to turn the lights on on the phone. Thanks |
| 02:41.27 | Queka | Where can I find a writeup on getting gvoice in and out working? |
| 02:43.00 | candrews | I don't know what to say... verbose = 10 in asterisk.conf, and sipdebug=yes in sip.conf. I've reloaded asterisk, and am tailing messages. |
| 02:43.35 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 02:44.32 | [TK]D-Fender | [21:42]<candrews>I don't know what to say... verbose = 10 in asterisk.conf, and sipdebug=yes in sip.conf. I've reloaded asterisk, and am tailing messages. <- NEITHER of those are CONFIG FILE parameters. Those are things to set in * CLI |
| 02:45.04 | [TK]D-Fender | N3tw0rK: Again, what is this "cisco", and where do I see anything that would indicate that * SHOULD send some sort of MWI? |
| 02:45.52 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 02:46.47 | N3tw0rK | [TK]D-Fender: shouldnt * send a sip message back to CUCME once a voicemail is left for an exten. At least thats how it worked 3 years ago when i last set it up |
| 02:47.29 | [TK]D-Fender | N3tw0rK: I see nothing. |
| 03:02.11 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 03:06.29 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 03:07.35 | *** join/#asterisk Defraz (~Defraz@gump.fuzecore.com) |
| 03:14.03 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 03:23.54 | [TK]D-Fender | ~seen titter |
| 03:23.56 | infobot | titter is currently on #asterisk (9m 53s), last said: 'Is it possible to set the cdr(userfield) from sip.conf per user?'. |
| 03:24.59 | pabelanger | candrews: PB your extensions.conf |
| 03:25.43 | [TK]D-Fender | pabelanger: No need to care about that so far |
| 03:26.21 | pabelanger | sure their is, I suspect is dial syntax is wrong |
| 03:26.32 | pabelanger | there* |
| 03:28.48 | pabelanger | candrews: exten => _1XXXXXXXXXX,1,dial(gtalk/candrewsintegralblue-google-in/+${EXTEN}@voice.google.com,,) |
| 03:29.57 | pabelanger | forget that |
| 03:30.39 | pabelanger | my brain is fried |
| 03:31.15 | WIMPy | Any dips around? |
| 03:31.22 | *** join/#asterisk moy_ (~moy@64.231.53.158) |
| 03:36.02 | [TK]D-Fender | thought that was only half-baked... |
| 03:43.46 | titter | mer |
| 03:47.59 | [TK]D-Fender | titter: Whats with the 1/2 join+quit all the time? |
| 03:48.13 | [TK]D-Fender | 1/2 hour |
| 03:54.44 | candrews | pabelanger, here's my extensions.conf: http://www.pastebin.ca/1991827 |
| 03:54.59 | candrews | I'm using exten => _1XXXXXXXXXX,1,dial(gtalk/candrewsintegralblue/+${EXTEN}@voice.google.com,,) - I think that's correct |
| 03:56.48 | candrews | irrational*CLI> sip set debug on |
| 03:56.49 | candrews | SIP Debugging re-enabled |
| 03:56.59 | candrews | irrational*CLI> core set verbose 10 |
| 03:56.59 | candrews | Verbosity is at least 10 |
| 03:57.21 | candrews | still nothing in my logs... am I setting those options right? |
| 03:59.54 | [TK]D-Fender | candrews: I want CLI OUTPUT. logs = trash |
| 04:00.29 | candrews | ah! I'm sorry! |
| 04:01.40 | candrews | http://pastebin.ca/1991830 CLI output |
| 04:01.43 | candrews | I'm reading this now.. |
| 04:02.39 | candrews | nothing sticks out at me |
| 04:03.15 | [TK]D-Fender | # <--- SIP read from UDP:192.168.0.231:5060 ---> CANCEL sip:18002464411@www.integralblue.com SIP/2.0 <--- ****EKIGA**** aborted the call |
| 04:03.33 | [TK]D-Fender | That sticks out for me. |
| 04:05.04 | candrews | you know what you're doing :-) Why would it abort? Here's another sequence - I captured for longer this time, until it stopped saying anything. http://pastebin.ca/1991831 |
| 04:05.24 | [TK]D-Fender | candrews: Try another client |
| 04:05.32 | [TK]D-Fender | candrews: it offers no reason. Test something else |
| 04:05.42 | candrews | trying csipsimple on android now.. |
| 04:07.48 | candrews | http://pastebin.ca/1991837 |
| 04:08.21 | candrews | Everyone is busy/congested at this time (1:0/0/1) Auto fallthrough, channel 'SIP/100-00000001' status is 'CHANUNAVAIL' |
| 04:08.25 | candrews | that seems suspicious |
| 04:09.55 | titter | [TK]D-Fender: No clue just saw the nickserv messages ... I am using Pidgin because I am lazy |
| 04:11.34 | [TK]D-Fender | titter: its been going on for a LONG time |
| 04:11.39 | [TK]D-Fender | titter: please fix your client |
| 04:13.41 | [TK]D-Fender | candrews: Garbage SIP packets.. lovely. |
| 04:13.59 | [TK]D-Fender | candrews: And what is this address remapping its referring to? |
| 04:15.12 | candrews | I have localnet set... I'm planning to use this outside my firewall, and the asterisk server is behind an ipv4 nat |
| 04:15.14 | titter | [TK]D-Fender: Actually it looks like an issue with my dd-wrt firmware, will see if I can figure out whats going on. If it doesn't get better I will just start disconnecting from irc |
| 04:20.26 | candrews | Is there something wrong with my dial plan? |
| 04:21.38 | titter | [TK]D-Fender: Made some changes, will see if things improve. |
| 04:22.42 | [TK]D-Fender | candrews: I'm wondering about your new garbage SIP packets, but I don't see GV "answering" anywhere |
| 04:24.10 | candrews | that's why I'm wondering if I screwed up the dial plan somehow |
| 04:28.16 | [TK]D-Fender | candrews: No |
| 04:28.51 | [TK]D-Fender | candrews: I'd se if you can enable a debug for gtalk... I don't know that protocol really.. |
| 04:28.52 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 04:32.36 | *** join/#asterisk coppice (~chatzilla@116.92.195.24) |
| 04:32.37 | candrews | turned on jabber debugging: http://pastebin.ca/1991851 |
| 04:32.47 | candrews | # |
| 04:32.47 | candrews | JABBER: candrewsintegralblue INCOMING: <iq from="+18002464411@voice.google.com/srvres" to="candrewsintegralblue@gmail.com/asterisk45FB52F4" id="jingle:10.218.11.68-30730860:11:3436B42B" type="set"><ses:session type="terminate" id="6653760f377459b0" initiator="candrewsintegralblue@gmail.com/asterisk45FB52F4" xmlns:ses="http://www.google.com/session"><pho:recipient-unavailable xmlns:pho="http://www.google.com/session/phone">Session timed out</pho: |
| 04:32.47 | candrews | recipient-unavailable></ses:session></iq> |
| 04:32.59 | candrews | wonder why my session timed out |
| 04:33.51 | [TK]D-Fender | candrews: More private IP's.. doesn't look legit |
| 04:38.38 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:40.12 | candrews | [TK]D-Fender, which private IPs do you see that seems suspicious? |
| 04:40.57 | [TK]D-Fender | ingle:10.218.11.68 |
| 04:42.27 | candrews | looking at a different problem, which is also annoying me... when I make a jingle call to my gtalk address, asterisk routes it to the default context. How can I route it to a different context? |
| 04:48.21 | *** join/#asterisk knot (~knotsucke@unaffiliated/devemo) |
| 04:53.49 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 04:53.55 | [TK]D-Fender | titter: GAH |
| 04:56.11 | p3nguin | Is he still joining and quitting all the damn time? I had to ignore him DAYS ago because of that. |
| 04:58.45 | [TK]D-Fender | p3nguin: Ignore on most clients doesn't seem to stop join/quit |
| 04:59.35 | p3nguin | I guess someone needs to make a better client, then. |
| 05:02.20 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 05:02.55 | shamelessn00b | Hi all |
| 05:24.19 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 05:25.54 | [TK]D-Fender | titter: Right on schedule... |
| 05:26.55 | shamelessn00b | http://www.youtube.com/watch?v=ASusmLRreXw |
| 05:27.58 | *** join/#asterisk costal (~ivan@corpnat.comindico.com.au) |
| 05:28.08 | costal | Hello All |
| 05:28.42 | costal | I'm trying to understand a legacy configuration which is working at the moment |
| 05:28.52 | costal | we have the mysql addon and using res_mysql.conf |
| 05:29.48 | costal | in the extconfig we have sipusers => mysql,database, sip_buddies |
| 05:30.30 | costal | I can see the sip_buddies table with a lot of usernames some with md5secret enable |
| 05:30.40 | costal | but some of them they don't have any on md5secret or secret |
| 05:30.44 | costal | my questio is |
| 05:31.09 | costal | how this could be posible if insecure is set to "no" ? |
| 05:31.14 | costal | any ideas ? |
| 05:31.36 | [TK]D-Fender | costal: that option is completely unrelated |
| 05:32.10 | costal | mmm |
| 05:32.46 | costal | any ideas how can sip user can access asterisk without secret or md5secret ? |
| 05:33.36 | [TK]D-Fender | costal: By not having a secret set against a peer at all, or allowing unauthed calls at all |
| 05:35.27 | costal | thanks Fender can you please point me in the right direction ? what settings should I look for ? |
| 05:39.46 | atan2 | costal, isn't there a static IP auth setup there somewhere? |
| 05:40.44 | [TK]D-Fender | costal: I think you should actually LOOK at your calls |
| 05:41.09 | *** join/#asterisk knot (~knotsucke@unaffiliated/devemo) |
| 05:41.39 | costal | mmmm yeah I can't find it |
| 05:42.08 | costal | I can see users with md5secret connecting to the asterisk box |
| 05:42.18 | [TK]D-Fender | costal: And you're showing nothing. Expect proportionate results. |
| 05:42.40 | [TK]D-Fender | costal: Yes, all passwords are normally Md5 |
| 05:43.24 | costal | but some users they don't have md5secret and I can see connecting to the asterisk all of them coming with different ips |
| 05:44.18 | [TK]D-Fender | costal: md5secret is FIELD, not how it is transmitted |
| 05:44.39 | costal | I mean in the Database sip_buddies |
| 05:44.41 | [TK]D-Fender | costal: this tells * that the field youfilled in with the password is ALREADY MD% hased so it doesn't DOUBLE has it. |
| 05:44.47 | [TK]D-Fender | hash* |
| 05:45.13 | costal | I mean in the Database sip_buddies I have some users with md5secret field empty and still connecting |
| 05:45.37 | [TK]D-Fender | costal: You are showing no logs, no SIP debug. There is nothing for us to do here. |
| 05:45.43 | costal | ok |
| 05:45.53 | costal | Cheers |
| 05:45.57 | [TK]D-Fender | [00:45]<costal>I mean in the Database sip_buddies I have some users with md5secret field empty and still connecting <- this field does not matter at all if it is blank |
| 05:47.03 | [TK]D-Fender | costal: it **IS** the password if you fill it in. If not then it is not used. SECRET is the "normal" password field, and it is in PLAIN TEXT. this will GET MD5 hased for transmission. The other one already IS so it doesn't need to be re-hased |
| 05:47.42 | costal | my problem is secret is empty no value at all |
| 05:47.56 | costal | I don't understand how this users are being authenticated thats all |
| 05:48.40 | [TK]D-Fender | costal: Why do you have peers configured with NO passwords? |
| 05:49.27 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 05:49.34 | [TK]D-Fender | costal: You seem to have set up a completely insecure peer |
| 05:54.39 | [TK]D-Fender | waits 30s |
| 05:54.47 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 05:56.24 | [TK]D-Fender | resets his stop-watch for 30 mins |
| 05:56.43 | *** join/#asterisk mintos (~mvaliyav@114.143.165.244) |
| 06:02.42 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 06:20.12 | *** part/#asterisk will_ (~wfong@pdpc/supporter/professional/will) |
| 06:24.52 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 06:32.50 | *** join/#asterisk raddy (~raddy@117.192.234.163) |
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| 06:47.06 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 06:52.42 | *** join/#asterisk brut- (~brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 06:55.50 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 07:02.14 | *** join/#asterisk Tim_Toady (~moi@77.49.252.191.dsl.dyn.forthnet.gr) |
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| 07:22.04 | *** join/#asterisk schmidts (~schmidts@lmlo.sil.at) |
| 07:22.06 | schmidts | good morning |
| 07:24.44 | *** join/#asterisk Weazel (~Weazel-@213.8.83.6) |
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| 07:30.21 | ChannelZ | Good? monday.. BAH! |
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| 07:39.53 | schmidts | hey atleast my working just start, the good was before i read my mails. You can cancel this by now |
| 07:41.19 | raddy | Hello Everybody |
| 07:41.26 | raddy | Can anybody help me |
| 07:41.40 | raddy | I have Voxzone x100p device. |
| 07:42.04 | schmidts | hello raddy, as usual the ability to help depends on your problem, but we can try it ;) |
| 07:42.09 | raddy | wxcfxo is not detecting the hardware. |
| 07:42.46 | raddy | says, "Failed to initialize DAA," |
| 07:43.42 | raddy | My system supports IO-APIC |
| 07:46.32 | raddy | schmidts : Any more information required? |
| 07:46.54 | schmidts | raddy to you use dahdi or zaptel? |
| 07:47.21 | raddy | schmidts : both are same right? |
| 07:47.37 | schmidts | raddy to be honest i dont have an idea about interface cards and there problems ;) |
| 07:47.58 | schmidts | not really the same but both do the same thing. zaptel is the old version and dahdi is the new one |
| 07:48.22 | raddy | schmidts : I am using Dahdi |
| 07:48.28 | schmidts | ok |
| 07:50.44 | *** join/#asterisk _zoom_ (~user@196.1.250.25) |
| 07:50.51 | _zoom_ | hi, |
| 07:52.05 | _zoom_ | I have two gsm gateways to terminate asterisk traffic, how to make them failover each other? |
| 07:52.44 | schmidts | raddy what do you see when you use lspci is your card recognized by your system at all? |
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| 08:05.01 | raddy | schmidts : Please check this http://pastebin.com/k62Hg9Ff |
| 08:05.26 | raddy | schmidts : the card is detected well by the OS. |
| 08:11.48 | raddy | schmidts : Are you there? |
| 08:12.05 | schmidts | raddy yes still here but not really any other idea |
| 08:12.15 | schmidts | what does dahdi tells you when you restart it? |
| 08:12.16 | raddy | okkkkk |
| 08:12.36 | raddy | schmidts : How do i start / restart it? |
| 08:13.03 | schmidts | /etc/init.d/dahdi stop and then start |
| 08:13.27 | *** join/#asterisk festr_ (~festr@nostromo.flh.cz) |
| 08:13.35 | festr_ | hi anyone on astertisk 1.4.37? |
| 08:13.45 | festr_ | I cannot blind transfer anymore |
| 08:13.51 | festr_ | transfering to own context |
| 08:14.10 | raddy | schmidts : It shows, lot of modukes with OK, then shows "No Hardwar timing source found in /proc/dahdi" |
| 08:15.14 | schmidts | raddy ok then dahdi doesnt recognize your card |
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| 08:19.38 | kaldemar | does x100p provide timing? if /proc/dahdi/* lists the card, it is recognized. |
| 08:20.23 | raddy | kaldemar : But, not by dahdi or asterisks |
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| 08:30.59 | ChannelZ | festr_: cannot how? |
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| 08:54.01 | hay | hi all... we are using siemens c470IP base station and I can see in logs that it happens sometimes (looks random to me) that all the extensions are unregistered (Unregistered SIP ext_number) and after about 5 seconds registered again... is this unregistration caused automatically for example when base station isn't responding / is this a standard procedure (I suppose it isn't) / could any of |
| 08:54.01 | hay | asterisk's settings be causing this? Of course if there is a call when this happens, the line disconnects... TIA |
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| 08:58.34 | ChannelZ | If you're actually seeing them UN-register and not just timing out (becoming "unreachable") then they're doing it on purpose |
| 09:00.53 | hay | ChannelZ, all the extensions on that base unit are un-registered at the same second... and then after about 4 seconds they all register again |
| 09:13.40 | *** join/#asterisk mintos (~mvaliyav@114.143.164.92) |
| 09:21.13 | mark22 | hello, what is the option for 1.8.x that is almost the same as nat=route for 1.6.x? that way if someone did give some internal (eg 1925. |
| 09:21.31 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 09:21.48 | mark22 | eg 192.168.x.x) ip it still could register with asterisk (while asterisk is not behind the same nat) |
| 09:22.05 | raddy | Hello Everybody |
| 09:22.41 | raddy | Anybody using Motorola based x100p card here? |
| 09:26.19 | *** join/#asterisk tyman_ (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
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| 09:37.40 | MCIML | has anyone gotten gtalk working with a google apps account? |
| 09:37.47 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 09:37.54 | MCIML | a non gmail.com address? |
| 09:41.30 | schmidts | MCIML i am not sure but i think this didnt work |
| 09:41.32 | hay | MCIML, yup... but on a domain that is using Google Apps so this is probably not the best answer :-) |
| 09:42.03 | MCIML | yeah thats what i have too |
| 09:42.14 | hay | MCIML, I see.. missed the first line :) |
| 09:44.24 | MCIML | how? |
| 09:45.33 | hay | I don't remember I did anything special... just download Google Talk and sign it with your user credentials... what error do you see? |
| 09:46.41 | hay | I still see this annoying un-registering with C470IP base station... http://pastebin.com/4VGs6JtH ... I can't find any reason or setting in the configuration of the base station for it... any help appreciated |
| 09:47.28 | MCIML | will jabber.conf work if you set username=uname@googleappsdomain.com instead of gmail.com? |
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| 10:05.07 | Diffen2 | Hello, is it possible to have one DID and on that DID check incoming number and then send the calls to two different IVR? We are planning on having one IVR and do a number check to see if the call is from sweden we send it to the swedish IVR if not we send it to the English IVR. |
| 10:25.45 | *** join/#asterisk whtsup (~whtsup@WimaxUser379-239.wateen.net) |
| 10:25.47 | whtsup | hello |
| 10:26.08 | whtsup | i m not getting ring back tone when i dial from dhadi channel |
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| 10:39.52 | *** join/#asterisk whtsup (~whtsup@WimaxUser378-161.wateen.net) |
| 10:39.56 | whtsup | hello |
| 10:39.59 | whtsup | anyone there ? |
| 10:54.09 | *** join/#asterisk DanFromUK (~DanFromUK@91.109.104.77) |
| 10:54.26 | DanFromUK | Hello, is anything free to assist with the following error: |
| 10:54.29 | DanFromUK | "username mismatch, have <silverstone_202>, digest has <silverstone_201>" |
| 10:54.36 | DanFromUK | anyone* |
| 10:54.51 | whtsup | chan_sip.c:4034 sip_setoption: Unknown option: 9 |
| 10:54.59 | whtsup | i m using asterisk 1.8 |
| 10:55.04 | whtsup | getting this error |
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| 11:01.12 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 11:02.49 | krion | hi |
| 11:03.11 | *** join/#asterisk mintos (~mvaliyav@114.143.164.92) |
| 11:03.48 | whtsup | hi i want to ask one thing |
| 11:04.08 | whtsup | i want to remove asterisk 1.8 and recompile to asterisk 1.6 |
| 11:04.15 | whtsup | wht will be best solution |
| 11:05.36 | krion | sorry if this is a noob question |
| 11:05.54 | krion | my sip_peers (and other stuff) are stored in mysql |
| 11:06.55 | krion | find it |
| 11:07.02 | krion | i've to do a sip prune <peer> |
| 11:07.19 | krion | correction: sip prune realtime <peer> |
| 11:13.30 | DanFromUK | i've got a problem registering two lines on a polycom phone, with asterisk. |
| 11:13.54 | DanFromUK | asterisk gets confused as to which line is making the call, and therefore cant authenticated. |
| 11:15.17 | kaldemar | DanFromUK: you need to tell asterisk not to match the devices by ip address, i.e. have user as the type. |
| 11:15.35 | kaldemar | whtsup: how did you install the 1.8.0? |
| 11:16.37 | kaldemar | Diffen2: yes, that is possible. you do it in the dialplan. |
| 11:18.52 | *** join/#asterisk Lipsum (~sengebret@77.40.154.242) |
| 11:20.30 | DanFromUK | kaldemar: if I change all my peers to users, does it affect anything else? |
| 11:20.48 | DanFromUK | kaldemar: will anything stop working or change? |
| 11:24.28 | kaldemar | depends on rest of your configuration. try it. |
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| 11:30.41 | whtsup | hello |
| 11:30.58 | whtsup | i m using asterisk 1.8 i want to remove 1.8 and install 1.4 |
| 11:31.13 | whtsup | i have remove completely asterisk 1.8 or i can just recompile |
| 11:31.14 | whtsup | ? |
| 11:31.30 | kaldemar | whtsup: did you see my question? |
| 11:38.33 | Diffen2 | kaldemar cool i will try to google it then |
| 11:40.41 | fauxalliance | Good Day Everyone. |
| 11:41.55 | DanFromUK | kaldemar: when i do that, i get "No matching peer found" when it tries to register. |
| 11:42.49 | DanFromUK | name and username both match and correct. |
| 11:45.02 | kaldemar | and host=dynamic? |
| 11:45.41 | DanFromUK | yep. |
| 11:45.58 | DanFromUK | p.s. im using realtime if that makes a difference. |
| 11:46.16 | *** join/#asterisk knot (~knotsucke@unaffiliated/devemo) |
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| 12:02.17 | DanFromUK | kaldemar: are you still available? |
| 12:08.47 | *** join/#asterisk raddy (~raddy@117.192.231.16) |
| 12:08.54 | raddy | Hello Everybody |
| 12:10.12 | shamelessn00b | hi raddy |
| 12:13.31 | raddy | shamelessn00b : I have a Motorola Wildcard based x100p fxo card. |
| 12:14.38 | raddy | shamelessn00b : I am experiencing irq conflicts with IDE controller or SMBus controller |
| 12:15.30 | raddy | shamelessn00b : Is it ok, if the IRQ be shared with IDE Controller? |
| 12:16.10 | raddy | shamelessn00b : i have SMP Processor and Kernel, would offer any workarounds? |
| 12:20.25 | raddy | Can any give a suggesion? |
| 12:20.54 | fauxalliance | raddy, don't share IRQ's.... use a $10 clone, get a $10 PBX. |
| 12:21.28 | raddy | fauxalliance : I don't understand |
| 12:22.12 | raddy | fauxalliance : I am using $10 clone x100p card, it is not original. |
| 12:22.37 | raddy | fauxalliance : It voxzone x100p card. |
| 12:23.46 | fauxalliance | raddy, wonderful... an expensive $10 pbx :P DONT SHARE IRQ'S! |
| 12:24.30 | raddy | fauxalliance : Ohh :( |
| 12:25.19 | raddy | fauxalliance : that appears to be difficult beast, the card shares irqs either with smbus controller or ide controller :(( |
| 12:25.37 | fauxalliance | "Interrupt Sharing" or an "Interrupt Conflict". I'd bet good money on the latter. |
| 12:26.18 | DanFromUK | I'm unable to use type=user. Friend and peer works fine. Any ideas? thnaks |
| 12:26.30 | fauxalliance | raddy, If too many high-irq-issuing devices share the same IRQ, it may cause delays in the IRQs getting serviced and can't even result in buffer overruns and other errors! |
| 12:27.17 | fauxalliance | DanFromUK, The difference between friend and peer is the same as defining _both_ a user and peer, since that is what 'type=friend' does internally. |
| 12:27.37 | fauxalliance | DanFromUK, capiche? |
| 12:28.35 | raddy | fauxalliance : there is no "too many", i have disabled most of the optional components in my motherboard, moreover the irq conflict may the prominent reason for the wcfxo driver to not detect the device, |
| 12:29.14 | fauxalliance | \o/ |
| 12:29.19 | raddy | fauxalliance : should i still try this solution ? http://www.asteriskguru.com/board/zaptel-amp-x100pcom-fxo-vt3239.html |
| 12:29.33 | raddy | even before resolving irq conflicts? |
| 12:29.40 | fauxalliance | raddy, sure, just don't confuse effort with success |
| 12:30.18 | fauxalliance | raddy, The problem was - IRQ. Changed motherboard! <try that |
| 12:30.18 | raddy | or resolving irq conflicts is mandatory to proceed further? |
| 12:30.37 | fauxalliance | raddy, YES! |
| 12:30.51 | fauxalliance | DanFromUK, ACK |
| 12:31.25 | raddy | fauxalliance : But, changing the motherboard is an expensing proposition. |
| 12:31.31 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 12:31.44 | fauxalliance | raddy, is getting a sensible card a better option? |
| 12:31.50 | hrhrhr | what command you using to show irq alloc |
| 12:32.20 | raddy | hrhrhr : lspci -v |
| 12:33.13 | fauxalliance | raddy, 'cat /proc/interrupts' |
| 12:34.01 | raddy | fauxalliance : I tried that command too, but can't interpret the results. |
| 12:34.16 | fauxalliance | raddy, looks like counting from zero to me. |
| 12:34.44 | hrhrhr | i woulda though proc/interrupts was infinitely easier to read |
| 12:35.11 | raddy | hrhrhr : fauxalliance : please see http://pastebin.com/sqh5h3bc |
| 12:35.28 | fauxalliance | raddy, this is also _way_ outside the scope of the channel IMHO... you'd be more likely to avail of (handholding) support with the purchase of a Digium (tm) card. |
| 12:37.01 | hrhrhr | im not sure which one is your x100p? shared are usually comma separated tho...? |
| 12:37.53 | fauxalliance | s/usually/are/ |
| 12:38.19 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 12:40.52 | hrhrhr | i dont see any comma separated on that pb, do you fauxalliance? |
| 12:41.00 | fauxalliance | hrhrhr, not a one |
| 12:41.33 | hrhrhr | you may as well pb your lspci output raddy |
| 12:41.37 | hrhrhr | not that i'll be able to help you |
| 12:41.39 | hrhrhr | because im a noob |
| 12:41.44 | hrhrhr | im just interested :P |
| 12:42.54 | tzafrir | raddy, how can you tell you have a problem? |
| 12:44.06 | tzafrir | raddy, from /proc/interrupts above, the driver for your card does not seem to be loaded |
| 12:44.21 | tzafrir | What's the output of dahdi_hardware ? |
| 12:44.37 | raddy | hrhrhr : fauxalliance : It appears something positive is happening, when rmmod wxfxo and modprobe wcfxo, |
| 12:45.46 | raddy | hrhrhr: fauxalliance : but while modprobing, the processes stalled :(( |
| 12:45.57 | fauxalliance | raddy, reboot... load the module automatically |
| 12:46.19 | tzafrir | raddy, also: be sure to use dahdi and not zaptel |
| 12:46.39 | raddy | tzafrir : i saw in, lsmod that my ide controller and fxo card use irq. |
| 12:47.43 | DanFromUK | fauxalliance: i knew that. I currently use "friend" but need asterisk to match based on username, and no IP. so i need to change it to "user". but when i do that, it cant register any more. |
| 12:48.01 | *** join/#asterisk netadmin (~toor@fsf/member/okaratas) |
| 12:48.36 | raddy | tzafrir : I am using dahdi it self. |
| 12:49.42 | DanFromUK | i get peer not found |
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| 13:02.35 | fauxalliance | DanFromUK, want to show the registration attempt in the CLI, pastebin it... smells like DNS |
| 13:04.38 | z4nD4R | hi all..i need make peering between asterisk and sipxec.. somebody any ideas???... i can find any how to... |
| 13:05.22 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:05.37 | raddy | tzafrir : fauxalliance : hrhrhr : after a reboot, when do rmmod and modprobe the wxfxo module, nothing happening, |
| 13:05.50 | tzafrir | because it's not loaded? |
| 13:06.53 | raddy | here is the error |
| 13:06.54 | raddy | http://pastebin.com/P5GxqTEM |
| 13:09.04 | raddy | tzafrir : hrhrhr : here is the lspci output. |
| 13:09.32 | *** join/#asterisk dapip (~IceChat77@64.85.142.34) |
| 13:09.55 | *** join/#asterisk delvin (~delvin@200-251-138-94.poolip.BHE.embratel.net.br) |
| 13:10.03 | tzafrir | raddy, what version of dahdi? |
| 13:10.06 | dapip | Hello |
| 13:10.57 | dapip | Wondering if someone could help me... I'm trying to find the setting in Asterisk that disconnects a call if it's inactive for xx time... Google hasn't been much of a friend lately for that information :\ |
| 13:11.03 | dapip | thanks in advance! :) |
| 13:11.20 | wdoekes2 | sip session timers? |
| 13:11.54 | raddy | tzafrir : i think 2.3 |
| 13:12.02 | dapip | Possibly... |
| 13:14.05 | [TK]D-Fender | dapip: You WANT it to disconnect in XX time, it its happening NOW and you don't want it to? |
| 13:15.48 | dapip | Fender: It's currently happening, I think, and I want to see the setting in the config... We have this conf call that our IT Team sets up in advance, need to know how far in advance we can set it up. |
| 13:16.41 | hrhrhr | raddy: http://www.asteriskguru.com/board/zaptel-amp-x100pcom-fxo-vt3239.html |
| 13:16.45 | hrhrhr | seems like your issue |
| 13:17.28 | [TK]D-Fender | dapip: it normally isn't a setting. |
| 13:17.44 | [TK]D-Fender | dapip: Nows the time where you actually describe your call so we ha ve a clue where to look... |
| 13:17.54 | raddy | hrhrhr : even i thought the same. |
| 13:19.08 | dapip | Fender:We setup a conference call using the Callme features in Asterisk, and normally it will sit for an hour or so waiting for the meeting to actually start and others to join the conference call. Ocassionally it will disconnect before the meeting actually starts.... |
| 13:19.08 | Diffen2 | kaldemar are you there? |
| 13:19.57 | [TK]D-Fender | dapip: what will "sit for an hour"? What "callme features", and WHAT KIND OF CALLS? |
| 13:20.58 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:22.42 | Diffen2 | anyone that know where i can find information regarding checking the incoming caller id on a certain number and send it to different feature codes regarding the caller id? |
| 13:24.03 | *** join/#asterisk Lipsum (~sengebret@77.40.154.242) |
| 13:24.05 | dapip | Fender: A conference phone will sit for an hour connected to the Asterisk, sorry, MeetMe conference call. |
| 13:24.34 | *** join/#asterisk goddva (~glarsen@77.40.154.242) |
| 13:24.46 | [TK]D-Fender | dapip: What kind of phone? What version of *? what does * show? |
| 13:25.13 | [TK]D-Fender | Diffen2: "core show function CALLERID", "core show application gotoif" |
| 13:26.24 | tzafrir | raddy, not sure if something similar was fixed later on. Or if it is merely defective hardware |
| 13:26.32 | Diffen2 | d-fender thanks will check it out |
| 13:26.41 | DanFromUK | Hi, i've got a question about realtime. Can IAXPeers and SIPPeers come from the same table? |
| 13:27.40 | [TK]D-Fender | DanFromUK: No |
| 13:29.20 | *** join/#asterisk davix (davix@unaffiliated/bashtard) |
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| 13:39.22 | hrhrhr | raddy so did you try that fix? or a later dahdi...? |
| 13:43.08 | *** part/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 13:46.46 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 13:47.23 | raddy | hrhrhr : i am looking at the source, i find very minor difference between the path author's version and the implementation. i am planning to directly contact the patch author. |
| 13:50.55 | *** join/#asterisk adeel (~adeel@c-67-174-36-109.hsd1.ca.comcast.net) |
| 13:52.57 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 13:54.11 | adeel | is there any kind of SIP compatibility issue between 1.4.x and 1.8? |
| 13:54.28 | [TK]D-Fender | adeel: Shouldn't be |
| 13:55.14 | adeel | hmmm...this is odd then...i can't seem to get calls between my 2 boxes...well, outbound works, but i can't get inbound to work...keeps returning 403 forbidden, even though they register. |
| 13:57.21 | c0rnoTa | е |
| 14:00.55 | adeel | i wonder if it's because the inbound context is set to user and not friend.... |
| 14:02.17 | adeel | nope, that didn't do much |
| 14:04.34 | adeel | interesting...the problem lay within not setting insecure=invite,port; is there any documentation that would explain why that must be set for a trunk, even though it registers? |
| 14:05.52 | [TK]D-Fender | adeel: That isn't the problem, that is your choice of ways around it. This shows that you aren't authing right and are opting to not auth at all to get around it |
| 14:06.14 | DanFromUK | Ive got an issue with Asterisk. It keeps losing connection. SIP packets keep getting lost. "re-transmitting (no NAT)". In order to get it working, i've got to "stop now" and restart asterisk. then it stops again after a few minutes. |
| 14:06.30 | DanFromUK | Any ideas? its really causing me problems. |
| 14:06.32 | adeel | [TK]D-Fender, ok, so if i wanted to auth properly...any idea what i'm doing incorrectly? |
| 14:08.03 | *** join/#asterisk espiceland (~erin@nat/digium/x-bdlekefcmtubvebv) |
| 14:08.09 | *** join/#asterisk espiceland (~erin@nat/digium/x-ufbkwibasokpzpod) |
| 14:08.19 | adeel | DanFromUK, sounds like your firewall is killing the port after some inactivity |
| 14:09.24 | [TK]D-Fender | BOTH of you should be posting SIP DEBUG in a PASTEBIN for us to look at. |
| 14:09.26 | [TK]D-Fender | ~pb |
| 14:09.26 | 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. |
| 14:12.11 | *** join/#asterisk pokus (~focus@158.193.202.22) |
| 14:12.46 | *** join/#asterisk coppice (~chatzilla@m121-202-54-90.smartone-vodafone.com) |
| 14:18.00 | SeTTleR | hi all, i've got some trouble with dahdi and a tdm400p and asterisk 1.6.2.13. asterisk sees the dahdi channels just fine and i can use the Dahdi/1-4 devices in the dialplan without error, but it just does not work. i thought it was a problem with the cable but now i noticed, that the wcfxs module is not loaded |
| 14:18.41 | SeTTleR | whenever i do a modprobe wcfxs, i get nothing returned, even with -v. the module is simply not loaded. do i need that module or is that handled by the wctdm module? |
| 14:19.15 | [TK]D-Fender | SeTTleR: that isn't the right module for that card |
| 14:19.36 | SeTTleR | aha? but wctdm should be the correct one?! |
| 14:19.42 | [TK]D-Fender | yes |
| 14:20.18 | SeTTleR | i'm just running out of ideas, so i read something about wcfxs and wcfxo and thought i should try that... |
| 14:21.22 | *** join/#asterisk fofware (~Fabian@host186.190-225-12.telecom.net.ar) |
| 14:21.59 | SeTTleR | ok then everything is handled by the wctdm module.. the asterisk is set up fine, but it just does not work. i can't place calls on a Dahdi Channel. in fact, i can, but i hear nothing and asterisk does not say anything except that it places the call |
| 14:22.14 | [TK]D-Fender | SeTTleR: And we see nothing. change that. |
| 14:22.16 | [TK]D-Fender | ~pb |
| 14:22.16 | 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. |
| 14:25.24 | pokus | hello i want ask about "SIP registration keepalive problem" ...i have device which is connected throught modem to voip network. This device is sometimes unreachable (lost connection). When connection is back this device is still unreachable while: 1) command REGISTER is not send..so first solution is set low re-registration period for REGISTER command. 2) when is set option qualify=yes my device is "on-line" immediately when keep alive command is send...what |
| 14:26.53 | pokus | in second case re-register period is same... |
| 14:27.19 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:27.26 | SeTTleR | here is the chan_dahdi.conf, dahdi show channels and the output from the asterisk cli: http://pastebin.com/uXey9Dwm |
| 14:27.38 | Katty | stares blankly |
| 14:29.22 | [TK]D-Fender | SeTTleR: What do you have connected to this card? |
| 14:29.41 | [TK]D-Fender | Katty: Mew. |
| 14:30.43 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 14:31.10 | SeTTleR | i have channel four connected to a speedport w920v. a router with telephone system integrated that offers two analog channels |
| 14:31.21 | SeTTleR | if i connect an analog phone, everything works fine |
| 14:31.22 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 14:31.41 | SeTTleR | channel 4 is the fxo module using fxs signalling |
| 14:32.05 | SeTTleR | i have that one connected with the speedport |
| 14:32.17 | SeTTleR | and try to place outbound calls with asterisk |
| 14:32.34 | FlashDeluxe | hi! how can i decrease the volume of music on hold? I ve set up mode=files but i am not sure how to toggle the volume :( Can anybody tell me the string if it is possible to change the volume? |
| 14:32.53 | *** join/#asterisk p3nguin_ (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:32.56 | [TK]D-Fender | FlashDeluxe: You can't with *. adjust your files |
| 14:33.11 | fauxalliance | ^^feature request? |
| 14:33.35 | FlashDeluxe | [TK]D-Fender So i have to manipulate the sound files? |
| 14:33.54 | [TK]D-Fender | enables echo-cancellation on #asterisk |
| 14:33.56 | fauxalliance | FlashDeluxe, thats what HE SAID |
| 14:34.01 | fauxalliance | FlashDeluxe, thats what HE SAID |
| 14:34.31 | fauxalliance | [TK]D-Fender, must be BORKED |
| 14:37.13 | *** join/#asterisk rrb3942 (~rbullock@208.34.96.186) |
| 14:37.19 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
| 14:37.52 | FlashDeluxe | [TK]D-Fender Sorry, i didn`t express myself in a right way, i meant: is there no other possibility than manipulating the soundfiles? |
| 14:38.20 | fauxalliance | takes the ClueBAT(TM) out of its protective sheath and hands it off to [TK]D-Fender |
| 14:38.49 | [TK]D-Fender | reaches for his rusty-nail upgraded ClueBat (TM) |
| 14:39.12 | fauxalliance | [TK]D-Fender, i just leaded mine :( |
| 14:39.48 | [TK]D-Fender | FlashDeluxe: Did I use a conjunction in there? I don't think so.... |
| 14:40.24 | fauxalliance | FlashDeluxe, 'man sox' |
| 14:40.27 | *** part/#asterisk raddy (~raddy@117.192.231.16) |
| 14:40.47 | FlashDeluxe | man, you`re in a bad mood...sorry for asking |
| 14:41.12 | DanFromUK | Hi, i'm trying to get up IAX to get around the firewall issues i'm having. but i'm getting this when the remote asterisk server tries to dial: Host 94.193.x.x failed to authenticate as estates_201 |
| 14:41.30 | [TK]D-Fender | FlashDeluxe: You just asked me the same question 3 times... |
| 14:46.47 | FlashDeluxe | excuse me for wasting your time but i wanted to know if there _really_ is no other way, thats why asked. |
| 14:47.39 | *** join/#asterisk SuPrSluG (~SuPrSluG@host-64-179-106-158.ind.choiceone.net) |
| 14:48.28 | [TK]D-Fender | FlashDeluxe: Not with "mode=files" |
| 14:48.58 | [TK]D-Fender | FlashDeluxe: Spend the minute it's worth to normalize your files. |
| 14:50.04 | *** join/#asterisk meatbun (~wafers4@cpe-98-155-139-88.hawaii.res.rr.com) |
| 14:50.24 | meatbun | there a way to log long distance dialing? |
| 14:50.44 | [TK]D-Fender | meatbun: All calls are logged in CDR <- |
| 14:51.46 | [TK]D-Fender | FlashDeluxe: http://www.google.ca/#sclient=psy&hl=en&site=&source=hp&q=sox+normalize+volume&aq=f&aqi=&aql=&oq=&gs_rfai=&pbx=1&fp=935c2c1965a84845 |
| 14:52.04 | [TK]D-Fender | FlashDeluxe: Curiously the first link is to voip-info expressly for this purpose |
| 14:53.03 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:53.03 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:54.53 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:55.02 | FlashDeluxe | [TK]D-Fender thanky you^^ |
| 14:56.20 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-ixanbzmgynmxcizn) |
| 14:57.56 | *** join/#asterisk jhirley (~chatzilla@mail.mmdlaw.com) |
| 14:58.13 | adeel | is it possible to have asterisk bind to 2 ports at once? |
| 14:58.45 | [TK]D-Fender | adeel: No, one or all. thats it |
| 14:59.28 | adeel | [TK]D-Fender, is there a downside to having * bound to all ports? |
| 15:00.30 | [TK]D-Fender | adeel: Do you see an issue withthis? |
| 15:01.18 | adeel | its definitely sub-optimal; would also imply that nothing else could run |
| 15:01.46 | adeel | sorry, been up for too long; |
| 15:01.55 | *** join/#asterisk m_tadeu (~quassel@89.180.107.80) |
| 15:01.57 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:02.17 | [TK]D-Fender | ... |
| 15:02.26 | [TK]D-Fender | "nothing else could run"... huh? |
| 15:02.36 | adeel | no other service could bind to a port if it wanted to |
| 15:02.40 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 15:03.36 | [TK]D-Fender | adeel: like what? What other SIP apps would be fighting for control over your ports on your server? |
| 15:05.15 | adeel | [TK]D-Fender, no sip other sip apps, but any other service like ssh, mysql, etc |
| 15:05.42 | [TK]D-Fender | adeel: What do those have to do with * binding SIP on all ports or not? |
| 15:07.08 | adeel | [TK]D-Fender, afaik, you can't have multiple services binding to the same port...e.g. 2 webservers can't be running on port 80...so if * binds to all 65535 ports, then nothing else could |
| 15:07.34 | adeel | well, nothing udp based, that is |
| 15:07.36 | [TK]D-Fender | adeel: OMG... |
| 15:07.46 | Gugge | [TK]D-Fender: multiple ports, not same port on multiple addresses :) |
| 15:07.49 | [TK]D-Fender | adeel: Binds to SIP on all INTERFACES |
| 15:08.09 | Gugge | adeel: so no, you can bind to one port, and thats it |
| 15:08.16 | adeel | [TK]D-Fender, ohh, i was specifically talking about PORTS, not interfaces |
| 15:08.27 | [TK]D-Fender | [09:58]<adeel>is it possible to have asterisk bind to 2 ports at once? <- no not multiple PORTS on the same interface. 1 port. On either ONE interface, or ALL interfaces |
| 15:08.29 | Gugge | adeel: but you can forward other ports to that port with whatever firewall software your os has |
| 15:08.35 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:08.35 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:08.41 | adeel | Gugge, yeah, i know about that |
| 15:09.51 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 15:10.18 | *** join/#asterisk oldhack_ (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 15:11.45 | [TK]D-Fender | adeel: Why would you ened * running on multiple ports on the same interface even? |
| 15:15.17 | fullstop | Could you simulate listening on 2 ports with iptables? |
| 15:15.21 | fullstop | or socat? |
| 15:15.46 | [TK]D-Fender | fullstop: Of course |
| 15:15.57 | [TK]D-Fender | I still fail to see a reason to do so |
| 15:16.17 | *** join/#asterisk McBoingbo (~mcboingbo@mail.hrsg.ca) |
| 15:16.20 | fullstop | ...thinking... |
| 15:17.02 | fullstop | Maybe they want to have the same sip user registered from different phones. |
| 15:17.05 | McBoingbo | I keep getting "No compatible codecs, not accepting this offer!" when trying to do a simple dialplan test, I have the codecs listed in sip.conf, log doesnt seem to be more verbose, any ideas? |
| 15:17.17 | fullstop | A sip proxy is more appropriate in that case. |
| 15:20.44 | [TK]D-Fender | fullstop: Not possible regardless. This has nothing to do with the way * listens on ports |
| 15:21.03 | [TK]D-Fender | McBoingbo: Clearly your codecs don't match. You should be looking at SIP DEBUGF from * CLI |
| 15:22.32 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 15:25.00 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:28.25 | *** join/#asterisk zplinux (~zplinux@213.8.57.217) |
| 15:28.31 | zplinux | hi all |
| 15:28.44 | zplinux | to install freepbx I need to have asterisk running |
| 15:28.55 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:29.27 | zplinux | they suggest to use ./start_asterisk start and this infact issues |
| 15:29.56 | zplinux | export LD_LIBRARY_PATH=/usr/local/lib ;/usr/sbin/safe_asterisk -U asterisk -G asterisk |
| 15:30.20 | zplinux | but I can't run this from a script to install free pbx |
| 15:30.40 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 15:33.35 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:33.37 | [TK]D-Fender | zplinux: ... |
| 15:33.41 | [TK]D-Fender | ~freepbx |
| 15:33.41 | 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 |
| 15:38.26 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 15:39.10 | McBoingbo | using Playback(), error indicates that the sounds file does not exist, but it is there, what else can I do to check? |
| 15:40.12 | [TK]D-Fender | McBoingbo: I'm betting that you aren'te telling it right or it doesn't exist |
| 15:41.06 | McBoingbo | I literally take the error line and ls -lah it and it is there |
| 15:41.19 | *** join/#asterisk dmast (~justsayin@exchange.newpointe.org) |
| 15:41.35 | [TK]D-Fender | McBoingbo: I literally see nothing. |
| 15:42.09 | McBoingbo | lies |
| 15:42.16 | [TK]D-Fender | "but I did everything right!" typically = BS. |
| 15:42.27 | McBoingbo | yeah yeah I know, I sup[port users too |
| 15:42.36 | McBoingbo | but I did check the file does exist and is there |
| 15:42.37 | [TK]D-Fender | McBoingbo: Of course you can't see the problem... otherwise you'd have FIXED it by now and we wouldn't be eharing about it |
| 15:42.48 | [TK]D-Fender | McBoingbo: Your checks and eyes clearly aren't cutting it |
| 15:43.02 | McBoingbo | who shit in your cornflakes this morning? |
| 15:43.59 | [TK]D-Fender | McBoingbo: unless you were merely expecting to vent frustration, you like many others come in showing nothing and expecting answers for why things don't work like you expect. |
| 15:44.25 | McBoingbo | not venting, just dont think your sarcasm is needed |
| 15:44.46 | [TK]D-Fender | ~pb |
| 15:44.46 | 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. |
| 15:44.48 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 15:45.15 | [TK]D-Fender | McBoingbo: Show us the "problem" and we'll show you what's wrong. |
| 15:45.28 | *** mode/#asterisk [+q [TK]D-Fender!*@*] by russellb |
| 15:45.35 | russellb | tired of the attitude, kthxbye. |
| 15:45.38 | McBoingbo | you seriously want me to pastebin showing proof there are sound files in the directory....lol |
| 15:46.39 | WIMPy | McBoingbo: Did you specify the file WITHOUT extension? |
| 15:46.44 | titter | ^ |
| 15:46.56 | McBoingbo | with extension |
| 15:47.03 | titter | Thats the issue |
| 15:47.59 | McBoingbo | ohhh I thought without it goes through the normal codec preference, with would choose the codec that you chosen, checking it now |
| 15:48.14 | McBoingbo | yup that was it |
| 15:51.30 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 16:02.08 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 16:03.16 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:06.33 | *** join/#asterisk McBoingbo (~mcboingbo@mail.hrsg.ca) |
| 16:06.44 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 16:07.26 | *** join/#asterisk ph8 (ph8@unaffiliated/ph8) |
| 16:08.09 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 16:17.25 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 16:18.23 | *** join/#asterisk tehrabbitt (~root@pool-71-172-89-155.nwrknj.fios.verizon.net) |
| 16:18.32 | tehrabbitt | p3nguin_: you here? |
| 16:19.02 | p3nguin_ | tehrabbitt: yep. |
| 16:19.58 | tehrabbitt | p3nguin_: did you get the money i sent to you? |
| 16:20.07 | p3nguin_ | tehrabbitt: Yes. |
| 16:20.32 | tehrabbitt | p3nguin_: ok, would you be able to email me your address so i can still send you the phone? |
| 16:20.37 | tehrabbitt | rabbott@tenehawk.com |
| 16:21.21 | p3nguin_ | tehrabbitt: I suppose I can do that. |
| 16:21.38 | tehrabbitt | sorry again about not being able to find the handset / power brick :-\ at least you'll have the phone.. consider it an xmas gift lol |
| 16:22.41 | tehrabbitt | so otherwise how've you been? finally got my asterisk working well using SIP.... so far so good lol |
| 16:23.24 | p3nguin_ | Why wouldn't Asterisk work well using SIP? |
| 16:23.47 | *** join/#asterisk dmast (~justsayin@exchange.newpointe.org) |
| 16:25.19 | tehrabbitt | p3nguin_: i'm talking about the huge headache i went through trying to get SIP working at my old ISP so i was forced to use IAX2 instead of SIP... now since i'm back @ my parents on the FiOS line, with a different router, and now I can use SIP and it works much much better |
| 16:26.23 | outtolunc | {parents thoughts} oh no.. now he'll never leave! |
| 16:26.28 | outtolunc | er no |
| 16:26.35 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 16:26.44 | outtolunc | ww |
| 16:27.40 | *** join/#asterisk Lantizia (~lantizia@188-221-11-131.zone12.bethere.co.uk) |
| 16:27.49 | *** join/#asterisk dmast_ (~justsayin@exchange.newpointe.org) |
| 16:28.20 | Lantizia | Hey... if I wanted to overlay a recording on top of an existing call (so both parties can hear it) what section of asterisk should I be researching? |
| 16:28.21 | p3nguin_ | tehrabbitt: Email sent. |
| 16:28.37 | *** join/#asterisk Corydon76-home (one@c-69-137-80-31.hsd1.tn.comcast.net) |
| 16:28.37 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 16:35.48 | Lantizia | anyone have any ideas? |
| 16:35.55 | *** join/#asterisk fofware (~Fabian@host186.190-225-12.telecom.net.ar) |
| 16:39.05 | outtolunc | Lantizia: your best bet is to read channel.x app_chanspy.x and app_zapbarge.c for ideas how those do it |
| 16:39.33 | Lantizia | outtolunc, don't want zaptel/dahdi |
| 16:39.44 | outtolunc | ok, so leave the last one off |
| 16:39.52 | Lantizia | are you saying there is no built in way? |
| 16:40.20 | outtolunc | you could have to use chanspy's barge in (if you are running a newer asterisk) |
| 16:40.40 | outtolunc | or mod the code |
| 16:40.41 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 16:41.23 | Lantizia | outtolunc, so this chanspy barge can overlay a recording on top of an existing call? |
| 16:41.42 | outtolunc | the issue is you still see this as 'a call' |
| 16:42.07 | outtolunc | it is really 2 (3 counting spy) 'legs of a call' |
| 16:42.37 | Lantizia | nevermind I think I have a better way |
| 16:42.53 | outtolunc | np, gl |
| 16:43.20 | p3nguin_ | What is your better way? Millions want to know. |
| 16:43.22 | citywok | Lantizia: use a meetme conference |
| 16:43.59 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 16:44.08 | Lantizia | p3nguin_, im gonna program a button on the phone (that can be used during the call) to inform the system to enable call waiting - send another call to the phone - and reenabel call waiting |
| 16:44.26 | Lantizia | then the operator need only to push conference on the phone to have the recording play and hang itself up |
| 16:44.27 | citywok | Lantizia: what are you trying to do? |
| 16:44.49 | Lantizia | I'm not using meetme - I'm trying to get rid of dahdi long term |
| 16:45.19 | Lantizia | confbridge may be an option though but it'd mean rebuilding the phone system to be 1.6.2 or above |
| 16:46.27 | WIMPy | I have no idea, what you're trying to do, but that sounds like a good idea anyway :-) |
| 16:46.40 | citywok | yea... i asked WHAT you were trying to do |
| 16:46.40 | citywok | not HOW. lol |
| 16:46.54 | citywok | if you tell us what you are trying to accomplish we may have a better suggestion for you |
| 16:46.56 | Lantizia | citywok, I was not answering you - you'll note it says p3nguin_ at the start |
| 16:47.07 | Lantizia | I don't know why your all confused about WHAT I'm trying to do since I said it when I entered the channel |
| 16:47.15 | outtolunc | has ideas falling out his ears.. what were we talking about again? |
| 16:47.31 | citywok | now i'm definitely not going to offer any advice. good luck with whatever the f*** it is you are doing. |
| 16:47.43 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 16:48.00 | Lantizia | thanks troll |
| 16:48.08 | russellb | citywok: don't make me ban you |
| 16:48.12 | citywok | yup, that's how i roll |
| 16:48.13 | outtolunc | happy monday.. la la |
| 16:48.35 | citywok | hey rus, he started it. i was asking what he did so i could offer advice. |
| 16:48.43 | p3nguin_ | NO U! |
| 16:48.49 | Lantizia | I can only answer to many people at a time, god! |
| 16:48.53 | Lantizia | *so |
| 16:48.53 | russellb | group hug everyone |
| 16:49.08 | outtolunc | showers first please |
| 16:49.11 | fullstop | yay |
| 16:49.17 | outtolunc | individually |
| 16:49.23 | fullstop | I used dial today. |
| 16:49.29 | outtolunc | hehe |
| 16:49.36 | Lantizia | so were are all on the same page... my opening line was... <Lantizia> Hey... if I wanted to overlay a recording on top of an existing call (so both parties can hear it) what section of asterisk should I be researching? |
| 16:49.37 | citywok | fullstop: that was bad |
| 16:49.38 | p3nguin_ | Dial() ? |
| 16:49.44 | fullstop | ha ha! |
| 16:49.55 | citywok | lol, that was punny |
| 16:49.59 | fullstop | I do enjoy bad puns |
| 16:50.07 | citywok | me too :p |
| 16:51.22 | citywok | hmm. so both parties hear it? just having the other end hear it is reallllly easy. |
| 16:51.36 | WIMPy | Can anyone enlightnen me what IAX tries to tell me by contantly moaning about "Bad address cast to IPv4"? |
| 16:51.57 | russellb | WIMPy: it was a bug, has been fixed |
| 16:52.10 | russellb | upgrade to the latest from the 1.8 branch, it will be fixed in 1.8.1 |
| 16:52.34 | WIMPy | Ok, thanks. |
| 16:54.07 | WIMPy | Can that also cause audio drop outs on peers where ti doesn't appear? |
| 16:54.37 | russellb | nah, wouldn't be realted |
| 16:55.15 | WIMPy | Hmm. Then I've got two problems. |
| 16:55.24 | citywok | russellb: i did a crappy backport to 1.6.2 and have it running now. i couldn't figure out how to ast_clear_flag the new option, so ended up hard coding what i wanted the way i wanted it. lol |
| 16:55.41 | russellb | heh. |
| 16:56.36 | citywok | out of curiousity, can i just re-compile the old module, unload the app_meetme in *, install the module, and load the new one (assuming nobody is using meetme)? |
| 16:56.53 | russellb | in theory, yes |
| 16:56.55 | citywok | i have the old one compiled and ready to install in the event something goes wrong |
| 16:57.18 | citywok | ewwww, i don't like theory. lol |
| 17:00.48 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 17:00.52 | *** join/#asterisk tyman (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
| 17:04.16 | *** join/#asterisk skrusty (~skrusty@93-97-20-22.zone5.bethere.co.uk) |
| 17:04.32 | skrusty | does anyone here use freetds and odbc with asterisk cdr or realtime? |
| 17:05.21 | citywok | neither, but i do use freetds & odbc with perl |
| 17:05.43 | skrusty | ever seen this error before: HY000: [FreeTDS][SQL Server]Could not change transaction status |
| 17:06.02 | skrusty | causes asterisk to drop the connection to sql, and then kills asterisk |
| 17:06.54 | citywok | does * segfault? it shouldn't unload completely, sounds like you may have found a bug. |
| 17:07.06 | citywok | and can you do a test query using isql? |
| 17:07.06 | skrusty | yeah, it segfualts |
| 17:07.12 | skrusty | i have full logs |
| 17:07.23 | citywok | what version? |
| 17:07.38 | seanbright | cdr_tds ftw |
| 17:07.39 | skrusty | Asterisk 1.6.2.1 |
| 17:07.56 | citywok | can you upgrade to the newest? that's old |
| 17:08.10 | *** join/#asterisk jhirley (~chatzilla@mail.mmdlaw.com) |
| 17:08.11 | skrusty | yeah, i will do tonight |
| 17:08.47 | citywok | If it happens in the newer version compile with dont_optimize and post an issue on the bug tracker. http://www.asterisk.org/developers/bug-guidelines -- include the backtrace |
| 17:08.57 | *** join/#asterisk BMJ (~bmj@nat/digium/x-gbddchtwkljgekrx) |
| 17:08.57 | *** mode/#asterisk [+o BMJ] by ChanServ |
| 17:09.01 | skrusty | cheers |
| 17:09.37 | citywok | also http://www.voip-info.org/tiki-index.php?page=Asterisk%20debugging |
| 17:16.58 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 17:18.22 | russellb | citywok: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 17:18.43 | russellb | citywok: become friends with the real wiki :-p |
| 17:19.19 | citywok | oh, nice. w/ another article for debugging :P |
| 17:19.26 | citywok | backtracing* |
| 17:19.37 | russellb | indeed |
| 17:20.19 | citywok | yayyy, with the new meetme feature i can use an xml meetme interface on my phones, with autojoin. lol |
| 17:20.35 | russellb | and then you could write a wiki article on it! |
| 17:20.46 | citywok | ewwwwwwww, article?!? |
| 17:20.57 | citywok | i wtire rike a turd girder |
| 17:21.16 | russellb | you look like one, too! |
| 17:21.17 | russellb | OH SNAP |
| 17:21.23 | citywok | oooooh burn |
| 17:21.38 | citywok | btw, thanks for dinner, i'm not sure if i ever said that haha |
| 17:21.48 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:22.05 | russellb | heh, you're quite welcome. |
| 17:22.17 | russellb | you paid me back in code for a new feature |
| 17:22.28 | citywok | hah, attach that to the expense report :P |
| 17:22.42 | citywok | receipt? nope. just h() |
| 17:23.29 | russellb | you didn't go with Æ©() ? |
| 17:24.13 | citywok | sorry, i can't type that on my keyboard |
| 17:24.28 | citywok | though i really miss my upside down question mark. i thought that was alt 1 6 7 haha |
| 17:25.11 | outtolunc | thats 168 for ¿ |
| 17:25.22 | WIMPy | ¿ save that for later |
| 17:25.35 | citywok | ah, that's the trick. and it doesn't work in this irc client anyways, lol |
| 17:26.11 | citywok | hmmm russellb i'm looking at app_confbridge and it really doesn't look like it would be THAT difficult to add username announcements |
| 17:26.18 | russellb | DO IT |
| 17:26.42 | russellb | Trying to get ConfBridge() feature compatible with MeetMe() is really high on my 1.10 priority list |
| 17:26.45 | citywok | and how come meetme is 7,000 lines but confbridge is only 700? |
| 17:26.51 | russellb | yes |
| 17:27.10 | citywok | hah, then i could NOT do it and let YOU do it :P |
| 17:27.25 | russellb | part of that is features, and some of it is that a lot of stuff has been abstracted out into other areas |
| 17:27.39 | outtolunc | points to the sign above the door ~code till equals~ |
| 17:27.39 | russellb | the bridging API (main/bridging.c) and the bridging technology implementations bridges/*.c |
| 17:27.56 | citywok | lol outtolunc |
| 17:28.19 | citywok | ah, okay. i guess that explains why this code is so much cleaner |
| 17:28.46 | russellb | it's not perfect yet ... we've got a few crashes reported, but gosh darnit it'll get there |
| 17:29.43 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:31.31 | skrusty | another quick question: anyone know why i would (only sometimes) get SIP/2.0 401 Unauthorized? Seems to happen sometimes if i stop and start asterisk on phones behind NAT. |
| 17:31.56 | citywok | before the phone has re-registered? |
| 17:32.02 | skrusty | i rebooted the phone |
| 17:32.12 | skrusty | doesn't register and in sip edbug i see SIP/2.0 401 Unauthorized |
| 17:32.22 | citywok | did you see it say "xxxx has registered" in the console? |
| 17:32.27 | skrusty | but nothing in the console about invalid peer authentication (user/password) |
| 17:32.30 | skrusty | nop |
| 17:32.31 | skrusty | e |
| 17:32.41 | citywok | do you have verbosity turned on? |
| 17:32.47 | skrusty | 30 |
| 17:32.56 | citywok | then your peer probably hasn't registered. |
| 17:33.05 | *** join/#asterisk JonnyD_work (~Jon@12.222.63.34) |
| 17:33.06 | skrusty | even after a reboot |
| 17:33.07 | citywok | try it, when you get 401 unauth sip show peer xxx and see if it's registered |
| 17:33.09 | skrusty | of the phone |
| 17:33.31 | citywok | it should say it on the console when the phone registers, so if you don't see it i'm wondering if the phone ever registered |
| 17:33.35 | skrusty | i see a register event, then SIP/2.0 401 Unauthorized event |
| 17:33.48 | skrusty | it's a linksys 540G |
| 17:33.57 | skrusty | but i have seen it with 4xx series too |
| 17:34.22 | citywok | ah, okay so it is registered. |
| 17:34.32 | citywok | enable sip debugging and pastebin it, somebody can probably read it and see. |
| 17:34.42 | citywok | i'm okay-ish at reading sip debugs |
| 17:35.43 | SeTTleR | when i do "modprobe wctdm debug=1" where does asterisk place the debug output? does that depend on logger.conf entries or do i miss something? |
| 17:35.45 | skrusty | http://pastebin.com/fed2pmfU |
| 17:37.24 | citywok | why are the from and to IP addresses the same? |
| 17:38.12 | skrusty | i have no idea... |
| 17:38.25 | citywok | does it say unauth immediately? |
| 17:38.29 | citywok | like, within a few MS? |
| 17:38.39 | skrusty | yeah |
| 17:38.48 | skrusty | there's no time between messages |
| 17:38.50 | citywok | i think it's responding to itself, and then unauth responding itself |
| 17:38.59 | skrusty | why would it do that? :/ |
| 17:39.00 | citywok | i'm assuming your * server is the 169.222 |
| 17:39.25 | citywok | idk, post your sip.conf and sip show peer xdev |
| 17:39.48 | skrusty | im using realtime |
| 17:40.07 | skrusty | Reg. Contact : sip:10002_1@83.166.176.39:5060 |
| 17:40.11 | *** join/#asterisk dmast (~justsayin@exchange.newpointe.org) |
| 17:40.13 | skrusty | so it knows the IP |
| 17:40.41 | citywok | yea i saw that in the via, so i wasn't sure. something is wrong. |
| 17:41.21 | *** join/#asterisk guilhermebr (~Guilherme@189.63.48.180) |
| 17:43.00 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 17:43.28 | skrusty | and yet: Addr->IP : (Unspecified) Port 0 |
| 17:43.45 | skrusty | seeems odd there's no output form chan_sip.so in the console |
| 17:43.58 | *** join/#asterisk tyman (~tyler@173-14-203-33-fresno.hfc.comcastbusiness.net) |
| 17:45.38 | drudge` | anyone used xo ip flex? |
| 17:46.00 | *** join/#asterisk tyman (~tyler@173-14-203-33-fresno.hfc.comcastbusiness.net) |
| 17:47.03 | *** join/#asterisk robl^laptop (~robl@pdpc/supporter/active/robl) |
| 17:47.32 | citywok | xo won't talk to me b/c i don't do over 500k minutes a month, and they don't support * 1.6 |
| 17:48.53 | robl^laptop | citywok: time for another vendor |
| 17:49.10 | citywok | i don't use xo, lol. they wouldn't take my money :P |
| 17:49.16 | citywok | skrusty: here's my realtime def: http://pastebin.com/3CYpsMJC |
| 17:49.17 | thehar | lol |
| 17:49.36 | citywok | and they don't support 1.6, it's too new. not even 1.6.0. lol |
| 17:51.57 | citywok | skrusty: i found that on voip-info's wiki. there may be erroneous options there, for those i don't take the blame. |
| 17:52.34 | citywok | russellb will be here in a minute to smack me for not using the new asterisk wiki, but i did that a long time ago, long before the asterisk wiki existed. |
| 17:53.47 | *** join/#asterisk robl^laptop (~robl@pdpc/supporter/active/robl) |
| 17:55.50 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-075-154-214.pools.arcor-ip.net) |
| 17:56.45 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:57.02 | leifmadsen | russellb went to lunch :) |
| 17:57.17 | thehar | i'm getting so confused with all of these timezones |
| 17:57.57 | citywok | lol. leifmadsen do you work from home? you live in canada right? |
| 17:58.09 | leifmadsen | yes, I work from home and live just outside of Toronto |
| 17:58.20 | leifmadsen | thehar: timezones are overrated |
| 17:58.28 | thehar | well i'm working on PST time |
| 17:58.31 | thehar | actually in MST |
| 17:58.36 | thehar | and all of you are in all sorts of timezones |
| 17:58.46 | leifmadsen | I'm only in 1 timezone |
| 17:58.49 | citywok | thehar at least you don't live in hawaii or arizona. then it would be worse |
| 17:58.56 | thehar | oh god |
| 17:59.12 | *** part/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 17:59.13 | citywok | what time is it there? *I DONT KNOW OMG BLAM* |
| 17:59.29 | robl^laptop | Indiana is bad too.. only certain areas deal with daylight savings.. |
| 17:59.43 | citywok | really? i thought it was just hawaii & arizona lol. |
| 17:59.55 | WIMPy | citywok: ctcp time? |
| 18:00.26 | citywok | 10AM PST (currently GMT -8) |
| 18:01.41 | fullstop | Arizona and Indiana give us much grief. |
| 18:01.57 | robl^laptop | hrmm. Indiana changed in 2006. They do daylight savings everywhere now. but they have a handful of counties that bounce between Eastern and Central |
| 18:02.21 | fullstop | For some reason I thought that Indianapolis was different than the rest of the state. |
| 18:02.35 | robl^laptop | I hadn't live there in about 15 yrs |
| 18:02.42 | fullstop | I am sick of DST and ST. We should just move 30 minutes in one direction and keep it there. |
| 18:02.43 | skrusty | citywok: just looking at your paste now |
| 18:03.14 | Qwell | fullstop: Indiana and Arizona almost had it right. It's the rest of the continent that has it all screwed up. |
| 18:03.18 | robl^laptop | fullstop: it used to be.. certain major cities did daylight savings, but outside those metro areas, they did not.. that is what changed in 20076 |
| 18:03.59 | citywok | Qwell: yes, i wish we didn't have "daylight savings". can somebody explain to me how it "saves" light? i mean, the sun is up for the same amount of time... amirite? |
| 18:04.02 | fullstop | Qwell: I am basing all of my opinions while living in Pennsylvania and New Jersey. |
| 18:04.13 | Qwell | citywok: the farmers. they should just get up earlier. |
| 18:04.15 | citywok | yes yes i know farmers blah blah, saving electricity blah blah. think of all the programmers! |
| 18:04.36 | citywok | we had this conversation a couple weeks ago during the switch, us people that have to code for DST really hate it. lol. |
| 18:04.36 | Qwell | citywok: there was a study done, that says more money is wasted by dealing with the time changes, than is actually saved. |
| 18:04.42 | fullstop | Qwell: I do understand why people want DST in places like Perth, Australia.. |
| 18:04.51 | Qwell | fullstop: 30 minute timezone. DONE. |
| 18:04.55 | tehrabbitt | DST is nothing but a headache... |
| 18:04.56 | tehrabbitt | lol |
| 18:05.02 | citywok | hahaha, that's awesome. and i believe it. every time somebody hs to write a piece of code to fix DST, argh |
| 18:05.29 | fullstop | Qwell: we could never name that many timezones here.. :D Eastern East, Eastern West... |
| 18:05.38 | robl^laptop | do away with it.. and if its an issue, you just change your schedule.. work from 8 to 4 or 9 to 5 depending on the time of the year |
| 18:06.13 | fullstop | I remember reading about Perth, in particular, and how without some sort of DST it is daylight at 3 AM or something like that. |
| 18:06.59 | fullstop | In the dead of winter, I go to work when it is dark and come back from work and it is still dark. |
| 18:07.31 | robl^laptop | in parts of alaska, night time lasts for several months |
| 18:07.58 | Katty | HAI |
| 18:08.04 | thehar | Katty: !!! |
| 18:08.08 | Katty | i haz come to collect hugs. |
| 18:08.23 | Katty | hugs on thehar |
| 18:08.29 | citywok | oh hai thar |
| 18:08.33 | thehar | ohhaithar |
| 18:08.35 | thehar | hugs Katty |
| 18:08.45 | thehar | Katty: i am relocating next week and can give address for holiday goodness |
| 18:08.47 | Katty | hugs citywok |
| 18:08.53 | skrusty | citywok: what version of * is that for? |
| 18:08.53 | Katty | thehar: most excellent, sir |
| 18:08.59 | citywok | hugs Katty |
| 18:09.05 | fauxalliance | too hugs Katty |
| 18:09.07 | robl^laptop | charges Katty income tax on the hugs -- "I demand at least 1 hug out over every 10 your receive." |
| 18:09.08 | Katty | :> |
| 18:09.09 | citywok | skrusty: 1.6.1 or 1.6.2 |
| 18:09.15 | Katty | hugs on fauxalliance |
| 18:09.16 | skrusty | ok |
| 18:09.19 | thehar | lol robl^laptop |
| 18:09.22 | citywok | i had it on 1.6.1 and now run 1.6.2.11 |
| 18:09.25 | Katty | robl^laptop: for you sir, i'd give a minimum of 3. |
| 18:09.37 | Katty | hugs on robl^laptop for an extended ammount of time. |
| 18:09.49 | thehar | get a room |
| 18:10.03 | Katty | will you be there? |
| 18:10.07 | Katty | ^_^ |
| 18:10.14 | fullstop | too much hugging |
| 18:10.17 | thehar | ohmai |
| 18:10.19 | thehar | blushes |
| 18:10.29 | robl^laptop | woo hoo!! thanks, Katty |
| 18:10.47 | fauxalliance | prefers hugs to contention... but will take what he can get :D |
| 18:10.53 | Katty | has anyone seen tk? |
| 18:11.02 | citywok | Katty he's come and gone a few times today |
| 18:11.03 | fullstop | He was here earlier. |
| 18:11.05 | citywok | I haven't seen him post |
| 18:11.08 | fauxalliance | he was here earlier, dropped |
| 18:11.11 | fullstop | being as blunt as usual.. :D |
| 18:11.20 | citywok | really? he hasn't slowed down after -o? |
| 18:11.30 | fauxalliance | fullstop, well, the clues were too sharp |
| 18:12.00 | fauxalliance | citywok, thats just a silly hat non? |
| 18:12.40 | Katty | file: ping |
| 18:13.10 | WIMPy | Not -o, +q |
| 18:13.46 | *** join/#asterisk carterv (~2panther@63-7.logisoft.com) |
| 18:13.55 | paulc | Katty: file's too busy watching his slingbox HD ;-) |
| 18:14.03 | paulc | slightly envious, but not really, but kinda |
| 18:14.10 | citywok | fauxalliance: ? |
| 18:14.40 | fauxalliance | citywok, yes? |
| 18:14.50 | citywok | silly hat what? |
| 18:14.59 | fullstop | I have an antenna in my attic. I don't get many channels, but most of them are in HD. |
| 18:15.35 | fauxalliance | citywok, i may be lost in context... -o (lose ops, take off the h@t) |
| 18:15.53 | citywok | oh, gotcha haha. |
| 18:17.12 | Katty | paulc: mmmmmmk. |
| 18:17.19 | *** join/#asterisk seanjohn (~seanjohn@gateways.sheltoncomputers.com) |
| 18:17.55 | fauxalliance | Katty, try #FreePBX... |
| 18:18.26 | fauxalliance | Katty, i sent him your hugs... he said tks |
| 18:18.46 | fauxalliance | is a proxy |
| 18:19.20 | carterv | so anyone here into Asterisk :) |
| 18:19.50 | fauxalliance | carterv, yeah.. this is the idle chit chat that comes when all the PBX's are working. |
| 18:20.15 | *** join/#asterisk McBoingbo (~mcboingbo@mail.hrsg.ca) |
| 18:20.40 | *** join/#asterisk trialbyfire (~trialbyfi@199.30.197.215) |
| 18:20.57 | Katty | or not working. |
| 18:21.04 | Katty | that might be my fault tho |
| 18:21.18 | fauxalliance | Katty, you bring sunshine... hardly an interruption |
| 18:21.46 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 18:22.20 | carterv | ah I see, well as time moves along I'm sure I'll have questions. My company merged with a few others and the main company has an existing Asterisk box and my company has an Allworx setup that for some reason we don't want to dump. I've been tasked with finding out how to make internal extension calls from our box to the asterisk box. Oye. |
| 18:22.46 | Katty | fauxalliance: idk about sunshine, but i will definately bring the hugs! |
| 18:23.28 | carterv | So knowing nothing about Asterisk I've started to build a test Asterisk box on top of CentOS 5.5 and see if I can get it to work |
| 18:23.40 | carterv | or break everything :D |
| 18:23.45 | fauxalliance | Katty, i've noticed, after several _YEARS_ or irc logs... #asterisk is a pretty 'huggy' spot... hugs are just little rays of sunshine |
| 18:23.51 | Katty | carterv: sounds very risky! |
| 18:24.03 | carterv | don't I know it! |
| 18:24.12 | Katty | carterv: i remember the excitement all too well ^_^ |
| 18:24.36 | retentiveboy | Got a CENTOS5 machine using the asterisk16 packages from the YUM repos at packages.asterisk.org. Updated today and it busted the mysql command. ANybody else see this? |
| 18:25.03 | fullstop | sounds like something to talk to the centos people about... |
| 18:25.09 | carterv | ha, excitement? interesting definition you have |
| 18:25.44 | McBoingbo | Where can I get more information about the "Capabilities" line from Sip debug, ex. "Capabilities: us - 0x104 (ulaw|g729), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)" |
| 18:26.08 | fullstop | McBoingbo: The message is pretty clear.. |
| 18:26.10 | *** join/#asterisk qubix (qubix@unaffiliated/qubix) |
| 18:26.19 | McBoingbo | not to me it isnt, thats why I am asking |
| 18:26.31 | retentiveboy | Looks more like the asterisk16-addons-mysql-1.6.2.1-1_centos5 package at asterisk.org is awry |
| 18:26.32 | fullstop | McBoingbo: It lists what codecs you have, what codecs the peer has and what you have in common. |
| 18:26.51 | fullstop | In your case, you agree on ulaw. |
| 18:27.00 | McBoingbo | so first portion is the server itself, and if calling by softphone, the peer would be the softphone in my case? |
| 18:27.22 | fullstop | Yes. |
| 18:27.25 | McBoingbo | yeah what I didnt understand is what portion refers to the server, and which to the client side |
| 18:27.34 | fullstop | "us" is the server. |
| 18:27.39 | fullstop | in your situation |
| 18:27.54 | McBoingbo | so then my server is not doing gsm then? |
| 18:28.02 | fullstop | If you were connecting two servers over sip, that definition is less clear. |
| 18:28.11 | *** join/#asterisk Arsenick (~y@fedora/Arsenick) |
| 18:28.16 | fullstop | Yes, your server is not doing gsm for this sip user. |
| 18:29.14 | McBoingbo | the only codec I have allowed for that user is gsm.... |
| 18:29.32 | McBoingbo | gsm just "works" out of box does it not? |
| 18:29.37 | fullstop | It does. |
| 18:29.43 | fullstop | Pastebin the config |
| 18:29.49 | McBoingbo | k |
| 18:30.42 | McBoingbo | http://pastebin.ca/1992426 is my sip.conf |
| 18:31.00 | McBoingbo | user 1001 is the one I am calling |
| 18:32.16 | qubix | I love 1001 |
| 18:33.33 | sbrath | If I'm picking up a DAHDI call to quickly, will I get State= Rsrvd(1) instead of "Ring" ? |
| 18:33.38 | McBoingbo | " NOTICE[7656]: channel.c:4006 __ast_read: Dropping incompatible voice frame on SIP/1000-00000031 of format gsm since our native format has changed to 0x4 (ulaw)" |
| 18:34.01 | McBoingbo | hmm, I realized that my user 1000 needed to have gsm allow'd, did that and still no workie |
| 18:36.13 | fullstop | Did you restart asterisk or do sip reload after making those changes? |
| 18:36.13 | fauxalliance | reloads |
| 18:36.15 | McBoingbo | yuppers |
| 18:37.34 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 18:37.35 | citywok | skrusty: did you figure it out? |
| 18:37.35 | wcselby | o/ |
| 18:38.12 | skrusty | not yet... |
| 18:39.03 | McBoingbo | fullstop: I reduced the allow line to just be "allow=gsm" and now it is working, I thought you can have several allow lines to create the order of preference for codecs |
| 18:39.21 | retentiveboy | Can anybody suggest the correct place to report issues with the binary packages being published on packages.asterisk.org? |
| 18:39.32 | Qwell | retentiveboy: what's wrong with them? |
| 18:40.24 | retentiveboy | Qwell: The addon packages appear to have been built with difference compiler options compared to asterisk itself and the modules won't load. |
| 18:40.46 | retentiveboy | Qwell: at least that my read of the faults I'm getting. |
| 18:43.15 | fullstop | McBoingbo: you should be able to. |
| 18:45.01 | *** join/#asterisk dmast (~justsayin@exchange.newpointe.org) |
| 18:45.46 | Qwell | retentiveboy: issues.asterisk.org |
| 18:46.00 | retentiveboy | Qwell: k, thx |
| 18:46.16 | *** join/#asterisk Tim_Toady (~moi@77.49.252.191.dsl.dyn.forthnet.gr) |
| 18:53.15 | Katty | SO. |
| 18:53.30 | Katty | what's new and fun and all that jazz. |
| 18:53.43 | thehar | things |
| 18:53.44 | thehar | n stuff |
| 18:53.50 | qubix | Katty lol HAY! |
| 18:53.50 | Katty | nods |
| 18:53.52 | Katty | awesome. |
| 18:53.58 | Katty | qubix: ohaithar |
| 18:54.01 | wcselby | hey Katty o/ |
| 18:54.05 | Katty | :> |
| 18:54.09 | Katty | wcselby: i didn't see you :>>> |
| 18:54.12 | qubix | lol, the other room was getting way too... "serious" i had to leave before I started to cry |
| 18:54.12 | Katty | hugs wcselby |
| 18:54.25 | wcselby | i got my dCAP at astricon, if that counts |
| 18:54.28 | Katty | qubix: i'm just everyone donchaknow |
| 18:54.30 | wcselby | :) |
| 18:54.35 | Katty | wcselby: oooh fancy |
| 18:54.37 | qubix | I doknownow |
| 18:55.07 | qubix | But, I'm out the door.. I have to go to UPS and ship a package & go to Panara for some f-ing amazing soup. |
| 18:55.19 | Katty | enjoy ^_^ |
| 18:55.24 | qubix | Have a good one.. |
| 18:55.51 | wcselby | heh |
| 18:56.05 | wcselby | how anyone can say anything from a panara bread is "f-ing amazing" is beyond me |
| 18:56.17 | wcselby | even their free wifi sucks |
| 18:56.20 | Katty | lol |
| 18:57.02 | russellb | Atlanta Bread >>> Panera Bread |
| 18:57.21 | wcselby | but then again, I'm sure some of the things I like are things some people would consider "f-ing awful" |
| 18:57.45 | wcselby | when I was in DC, I ate at Chipotle three times I think, in the course of 6 days |
| 18:57.56 | *** part/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 18:58.07 | wcselby | i tried eating at the convention center restaurants, but they wanted way too much money for not very good food |
| 18:59.06 | Katty | i had a really hard time at the first cluecon |
| 18:59.07 | *** part/#asterisk carterv (~2panther@63-7.logisoft.com) |
| 18:59.19 | Katty | i was under the impression the hotel that was feeding us had vegan/vegetarian stuffs |
| 18:59.31 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 18:59.36 | Katty | apparently they don't realize that vegans and vegetarians need protein |
| 18:59.46 | Katty | and not just iceburg lettuce and bits of carrot |
| 19:00.05 | wcselby | lol |
| 19:00.43 | wcselby | heh, funny that - i almost went into this fancy looking burger joint down the street from the hotel, until I noticed it said it was a vegan burger joint, or some such |
| 19:00.55 | wcselby | that was in the small print on the door, not the big print on the title / sign |
| 19:01.59 | wcselby | yeah, it was this elevation burger place - http://www.elevationburger.com/EB.php |
| 19:02.47 | *** join/#asterisk jbroome (jbroome@unaffiliated/jbroome) |
| 19:03.17 | wcselby | although looking at their menu, I don't know how they can be a vegan place |
| 19:03.31 | wcselby | unless "beef patty" means something different to vegans than it does me? |
| 19:03.50 | Katty | hmm |
| 19:03.53 | Katty | it could mean they are vegan friendly |
| 19:03.57 | Katty | and serve original boca burgers |
| 19:04.07 | wcselby | i dunno, I guess maybe I read the small print incorrectly |
| 19:04.13 | Katty | hard to say |
| 19:04.23 | Katty | if you've not tried a veggie burger at some point you really should |
| 19:04.36 | Katty | the soy burgers aren't the greatest thing ever, but i sure am a fan of black bean burgers |
| 19:04.48 | *** join/#asterisk DanFromUK (~DanFromUK@91.109.104.77) |
| 19:04.51 | Katty | especially with a little liquid smoke |
| 19:05.31 | DanFromUK | Hi, im having problems with "No matching peer found" when type=user, using realtime. Does anyone have some free time? |
| 19:08.09 | wcselby | my wife likes veggie burgers |
| 19:08.14 | wcselby | the one or two I've tried I hated |
| 19:10.29 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 19:10.42 | fauxalliance | in Canada... President's Choice (Loblaws Brand) have Ancient Grains veggie burgers... < only good one's that dont contain meat |
| 19:11.52 | fauxalliance | Yves SUCK! |
| 19:12.31 | fauxalliance | http://www.oishisauce.com/ <yum |
| 19:12.57 | wcselby | DanFromUK - well, you defined a user, not a peer....what's the question? |
| 19:13.21 | fauxalliance | wcselby, this was disambiguated yesterday |
| 19:13.25 | wcselby | DanFromUK - you have users, you have peers, they're separate entities. if you have type=friend, you create one of each |
| 19:15.21 | wcselby | heh, okay |
| 19:15.47 | DanFromUK | I know, however, i have a polycom phone that has two lines. when i dial out line 2, it works fine. Line 1 doesnt work. |
| 19:15.57 | DanFromUK | Its because its matching via the IP and not the username |
| 19:16.15 | DanFromUK | Someone on here said that i need to change it to type=user to force matching via username. |
| 19:16.56 | DanFromUK | But then i get the " No matching peer found" message. |
| 19:18.01 | wcselby | DanFromUK - that seems odd. i've had multi-line registrations on polycom phones before, from the same IP address to the same server, and both lines work just fine |
| 19:18.28 | wcselby | you made a post on the list about this, right? i remember seeing it and thinking the response you got sounded odd |
| 19:19.20 | wcselby | how are your peers setup on the phone itself? I need sip.conf settings for each peer (or the realtime output) and then the settings from each phone, and possibly a sip debug from you trying to make a call from line 1 and line 2 |
| 19:20.18 | wcselby | use pastebin |
| 19:20.22 | DanFromUK | wcselby: here is the cli message when line 1 tries to dial. |
| 19:20.22 | DanFromUK | http://pastebin.com/F5B0cenv |
| 19:20.59 | DanFromUK | I say "line 1" but actually, only the last line to register can make calls. |
| 19:21.15 | wcselby | DanFromUK - please show me a sip debug of the same situation, and the other information I requested |
| 19:21.35 | DanFromUK | ok, give me 2 minutes to put it together. |
| 19:22.28 | *** join/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 19:23.37 | nny | so spent some time this weekend looking to call notification programs in windows. I have a client who wants a web popup on call with either CID or custom field from the dialplan. Everything I found was either a dead link, hasn't been touched in years, or did way more than what I needed. Any advice? looking at ADAT right now as the top contender. |
| 19:23.54 | nny | YACID and YAC both seem dead |
| 19:24.28 | wcselby | I use GoogleTalk and JabberSend as my screen popup |
| 19:24.32 | Katty | peeks in |
| 19:24.43 | nny | wcselby: yeah looked at XMPP |
| 19:25.15 | nny | wcselby: but need it to auto pop the web page up like http://asteriskserver/CID=5551212&CUSTOM=SOMETHING |
| 19:25.27 | DanFromUK | wcselby: seems to be tempermental. its working now. i'll check again shortly. |
| 19:25.29 | nny | yacid/yaacid |
| 19:25.54 | nny | the problem with yaacid is the lack of movement on it. Hate to commit them to a solution that has no support channel or updates. |
| 19:26.05 | fullstop | roll your own? |
| 19:26.10 | nny | even willing to pay for it if need be, just can't find anything that does what I need it to |
| 19:26.36 | nny | fullstop: my windows/.net experience is null. I am use to working with scripting languages only |
| 19:27.02 | fullstop | python + wxWidgets ? |
| 19:27.48 | nny | fullstop: noted. Gonna keep looking for something commercially or FOSS |
| 19:29.01 | wcselby | nny - doesn't FOP2 have a screen pop if you're logged into it? |
| 19:29.07 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 19:29.16 | nny | wcselby: hmm. I don't think so, but maybe he added it |
| 19:29.25 | nny | wcselby: will check that out, they use FOP2 currently |
| 19:29.26 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 19:29.49 | nny | wcselby: http://www.fop2.com/forum/viewtopic.php?f=4&t=313 |
| 19:29.58 | nny | interesting |
| 19:36.36 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 19:55.01 | *** join/#asterisk Corydon76-dig (mauve@c-69-137-80-31.hsd1.tn.comcast.net) |
| 19:55.01 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 19:55.18 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 19:57.55 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
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| 20:17.34 | rbht | hello everyone |
| 20:18.43 | skrusty | lo |
| 20:19.28 | rbht | hows it going skrusty |
| 20:21.26 | rbht | has anyone here setup GVoice with LOW_MEMORY enabled? mine seems to dump core on inbound call |
| 20:21.38 | skrusty | nope :/ |
| 20:21.38 | rbht | not sure if its a bug or some package error |
| 20:21.48 | rbht | without LOW_MEMORY - works fine |
| 20:22.47 | russellb | probably a bug in asterisk technically. |
| 20:22.56 | russellb | We set the thread stack size to be much smaller with LOW_MEMORY enabled |
| 20:23.10 | russellb | so code paths that are normally fine may run out of stack space with LOW_MEMORY on |
| 20:31.40 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 20:41.48 | Katty | dear lord. |
| 20:41.51 | Katty | i have had too much caffeine |
| 20:41.54 | Katty | jitters |
| 20:42.24 | robl^laptop | Katty: enable the jitterbuffer in katty.conf |
| 20:52.05 | fullstop | I picked up a bottle of Coca-Cola Refresco today. |
| 20:52.29 | fullstop | I rarely see them in the north east. Glass bottle + cane sugar. |
| 20:55.46 | *** join/#asterisk qubix (qubix@unaffiliated/qubix) |
| 20:57.43 | robl^laptop | like "Pepsi Throwback"? |
| 20:57.56 | jdoe | is there a 'best practice' for managing more than a handful of phones with persistent registrations? ... should I be hard-coding ip addresses in sip.conf or something, or is there a better way? |
| 21:01.29 | fullstop | Kind of. It's all in Spanish, though. I think most places call it Mexican Coke. |
| 21:03.16 | Corydon76-dig | jdoe: If you're getting too much registration traffic, extend the registration time |
| 21:03.55 | Corydon76-dig | jdoe: or do you mean outbound registrations? |
| 21:10.31 | Katty | robl^laptop: <3 |
| 21:24.05 | *** join/#asterisk tyman (~tyler@173-14-203-33-fresno.hfc.comcastbusiness.net) |
| 21:25.28 | carrar | moof |
| 21:29.53 | *** join/#asterisk [hC] (~voxter@74.198.148.43) |
| 21:30.08 | [hC] | polycom dudes: Am I supposed to be using this UCS firmware on newer phones? |
| 21:34.56 | luckman212 | I have 2 SIP trunks into an Asterisk 1.8 pbx. Inbound calls work fine & are routed correctly to my IVR. the "Problem" is that callers don't hear any ringing-- they hear a brief silence and then the IVR is talking to them. Somehow this confuses people and I am getting a lot of "your phones aren't working" comments from customers. Any idea why they wouldn't hear a ring or 2 before the IVR picks up? |
| 21:35.25 | luckman212 | I'd like to at least make it ring once or twice |
| 21:36.48 | *** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f1-69-94-238-22.totalprocess.net) |
| 21:37.54 | Letoric | Anybody able to point me to a good section of documentation on implementing features like *67 (or more specifically, my particular desire to enable certain personnel to disable call monitoring for 1 call) |
| 21:39.16 | robl^laptop | luckman212: trying putting in a wait(2) (wait 2 seconds) before the answer() |
| 21:40.26 | luckman212 | robl^laptop: i'll give that a shot, i'm using pbx-in-a-flash so im not sure if my manual edits to the dialplan will get preserved |
| 21:41.07 | robl^laptop | luckman212: ahh! that uses FreePBX for a GUI. Not too sure. Try asking in #freepbx for confirmation. |
| 21:41.14 | wcselby | luckman212 - you would have to add a ringing sound |
| 21:41.50 | *** part/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
| 21:41.58 | luckman212 | ok going to ask over in #freepbx, thx |
| 21:42.06 | wcselby | luckman212 - something like a "playback(ringing)", but I'm not sure if there's a file like that already |
| 21:44.52 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 21:46.01 | robl^laptop | wcselby: if you wanted to do it that way, you would use Playtones(ring), Wait(2), then StopPlaytones(). |
| 21:46.11 | wcselby | well there you go |
| 21:46.24 | wcselby | a Wait() isn't going to provide any sound, just make it take longer for sound to be played |
| 21:47.06 | robl^laptop | Playtones just starts playing that tone.. untill you explicity tell it to stop. so if you don't put wait() you wouldn't hear anything |
| 21:47.42 | wcselby | i meant, the Wait() before his answer isn't going to add any sound of ringing, which his customers are asking for |
| 21:47.48 | robl^laptop | without wait, the stop gets executed a fraction of a second after you started the tones |
| 21:48.14 | robl^laptop | wait before answer would let the phoine company provide the ringing |
| 21:48.41 | wcselby | why would they provide the ringing if they aren't already? |
| 21:49.57 | robl^laptop | if there is an inbound call, but not yet answered, the phone company typially provides a ringing tone until the call is answered |
| 21:50.09 | jdoe | Corydon76-dig: sorry, went to lunch. Basically I have two pbxes, the phones register to both, everything is good. One fails, the phones start using the other. If both fail, the phones won't re-register until the original registation expires. |
| 21:50.16 | wcselby | hmmm, that hasn't been my experience |
| 21:50.29 | robl^laptop | wcselby: also depends on the type of trunks |
| 21:50.35 | jdoe | Corydon76-dig: I guess what I'm wondering is what's the right thing to do... crank down the registration period? hard code ips? something I haven't thought of? |
| 21:50.35 | wcselby | the only time I've gotten a ringing tone was if it was an analog line coming into the system |
| 21:50.48 | wcselby | sip trunks and PRI just go straight into the system with no ringing |
| 21:50.49 | Corydon76-dig | jdoe: fix your servers? |
| 21:51.16 | robl^laptop | wcselby: if you use T1 or ISDN/BRI, prolly Progress() is a better option. |
| 21:51.25 | jdoe | Corydon76-dig: "fix"? |
| 21:51.33 | Corydon76-dig | If your two servers are down simultaneously often enough, there's something wrong with the servers you're using. |
| 21:51.58 | luckman212 | wcselby: its certainly possible that the IVR is just answering "too fast" for a ringtone to be played back by the service provider |
| 21:52.00 | jdoe | Corydon76-dig: they're likely to be restarted on an asterisk upgrade. |
| 21:52.14 | jdoe | Corydon76-dig: that's the only time I anticipate having both down at once, although realistically it'll happen eventually. |
| 21:52.15 | Corydon76-dig | jdoe: then stagger your upgrades |
| 21:52.46 | jdoe | so there's no right way to do it then? |
| 21:53.11 | Corydon76-dig | Realistically, the only time your redundant servers should ever be down is in case of a very long power outage in which your generators run out of diesel |
| 21:53.11 | wcselby | luckman212 - i dunno, when I add a "Wait()" onto the front of my IVR's, before an Answer(), I just get extra silence. This is on T1's. |
| 21:53.53 | Corydon76-dig | For any other case, see "Acts of God" |
| 21:54.24 | jdoe | that's reasonable I suppose, surviving one failure ought to be enough. |
| 21:54.53 | robl^laptop | luckman212: all of the options offered are worth a try for your sceneria. simplest is just to add a wait(). if not, you can try (Progress) or use Playtones() |
| 21:55.47 | luckman212 | robl^laptop: thanks, trying now |
| 21:55.58 | Corydon76-dig | Any other case, expect your servers to have smoke or water damage, and in that case, nobody will be using those phones at that time, anyway |
| 22:01.02 | robl^laptop | yeah, no one would be dialling 911 if the servers were on fire in the middle of a flood ;-) |
| 22:01.28 | *** join/#asterisk gnarf (~gnarf@c-98-213-203-220.hsd1.il.comcast.net) |
| 22:01.43 | gnarf | is there a way i can execute an AGI if the "called party" presses a button? |
| 22:02.16 | carrar | exten => 1,1,agi,poop |
| 22:02.55 | carrar | just like you would for a main menu type option setup |
| 22:03.29 | Kobaz | <PROTECTED> |
| 22:03.31 | carrar | or you mean a button on a phone? speed dial a agi exetnsion? |
| 22:03.35 | Kobaz | wow that's an awesome callerid |
| 22:03.47 | russellb | nice |
| 22:03.59 | carrar | mac address caller id? :) |
| 22:04.13 | Kobaz | callerid from a customer's grandstream fxo box |
| 22:04.25 | Kobaz | i need to get those things out of there |
| 22:05.36 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:06.17 | *** join/#asterisk [cannibalera] (~cannibale@201-11-104-114.fnsce703.dsl.brasiltelecom.net.br) |
| 22:06.23 | *** part/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:06.24 | luckman212 | robl^laptop: I added Wait(10) to the top of my IVR and I just get more silence- still no ringing :-( |
| 22:06.34 | luckman212 | same as wcselby |
| 22:06.38 | luckman212 | hrrm :/ |
| 22:06.59 | wcselby | luckman212 - what did you say you were using? elastix? |
| 22:07.10 | luckman212 | pbx-in-a-flash |
| 22:11.39 | wcselby | did you try one of the other options mentioned? I even think there's an asterisk command named "Ringing()" |
| 22:12.37 | luckman212 | yes there is Ringing() ... i'm checking on it now |
| 22:13.00 | wcselby | if that doesn't work, the Playtones(ringing) option that was mentioned earlier is probably the best bet |
| 22:13.11 | wcselby | Playtones(ring) |
| 22:15.17 | robl^laptop | Playtones(ring) then Wait(secs-to-ring) then StopPlaytones() |
| 22:15.49 | *** join/#asterisk [canniballllera] (~cannibale@201-11-104-114.fnsce703.dsl.brasiltelecom.net.br) |
| 22:16.06 | *** part/#asterisk [canniballllera] (~cannibale@201-11-104-114.fnsce703.dsl.brasiltelecom.net.br) |
| 22:16.31 | luckman212 | hey whaddya know |
| 22:16.54 | luckman212 | i added Ringing() before the Answer() |
| 22:16.59 | luckman212 | and I hear ringing now! |
| 22:17.42 | luckman212 | thanks robl^laptop, wcselby |
| 22:17.50 | wcselby | np :) |
| 22:18.15 | luckman212 | now just have to figure out how to keep pbx-in-a-flash from clobbering that change if I use the gui again |
| 22:18.16 | robl^laptop | wcselby: ohh! I forgot about Ringing() |
| 22:18.27 | luckman212 | http://www.voip-info.org/wiki/view/Asterisk+cmd+Ringing |
| 22:18.38 | p3nguin_ | It still only rings while waiting on something else. |
| 22:19.55 | *** join/#asterisk tamiel (~tamiel@ip-175.net-89-3-223.rev.numericable.fr) |
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| 23:09.26 | *** join/#asterisk marl_scot (~marl_scot@cpc1-cowc4-0-0-cust381.14-2.cable.virginmedia.com) |
| 23:11.10 | marl_scot | hi guys, i am trying to get chan_lcr working with my * box, i can now dial in without any problem, but when i try and dial out i get chanal unavailable after a 5 second pause, i get 'Incomming release from LCR, releasing ref. (cause=18)' in the * console, any one point me in the write direction, or let me know what other info you need? |
| 23:11.12 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 23:32.59 | marl_scot | i think i need to configure lcr to allow * to dial out, but i cant work out how :( |
| 23:36.03 | *** join/#asterisk kuku (~kuku@c-24-13-139-34.hsd1.il.comcast.net) |
| 23:36.28 | kuku | If people's voices are breaking up, what command can I run to see that on the CLI ? |
| 23:36.40 | kuku | (phone quality issue ) |
| 23:40.11 | jblack | It's one of two things. Either the system load is too high, or not enough bandwidth. |
| 23:41.53 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 23:44.40 | kuku | jblack: could it be not enough io on monitor ? |
| 23:45.08 | kuku | load is 0.05 |
| 23:49.39 | kuku | jblack: any way to see that there is not enough bandwith ? |
| 23:56.59 | citywok | kuku it can also be low bandwidth quality |
| 23:57.09 | citywok | high throughput doesn't mean good throughput |
| 23:57.28 | *** join/#asterisk akoma1s (quasselcor@unaffiliated/akoma1s) |
| 23:57.53 | citywok | if load is that low, i'd assume it's likely a network issue be it enough bandwidth, or quality of the connection. if this is a residential cable/dsl connection... the quality may not be that great during the ISP's primetime hours |