| 16:20.43 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 16:20.43 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.1 (2012/01/19), 1.8.8.2 (2012/01/19), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 16:24.52 | asteriskATmarmuD | bye guys, and thanks for your time! |
| 16:29.33 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 16:30.55 | FlashDeluxe | hi everyone! got a question: how can i allocate different voicemail intros to different numbers? |
| 16:31.12 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:31.12 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
| 16:31.20 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 16:31.27 | Qwell | FlashDeluxe: There's an option to skip the intro in app_voicemail, and then you'd just Playback() before calling Voicemail() |
| 16:32.04 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 16:32.25 | FlashDeluxe | qwell: ok thanks :) |
| 16:32.32 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 16:37.09 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
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| 16:43.53 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v000-101.mobile.uci.edu) |
| 16:46.43 | *** join/#asterisk plundra (1000@v0.article.se) |
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| 17:05.14 | c0rnoTa | WIMPy: coppice: after downgrading to 2.5.0.2 HDLC Abort message was disappeared :) |
| 17:05.24 | c0rnoTa | dahdi 2.5.0.2 I mean |
| 17:05.41 | WIMPy | Hmm. bad. |
| 17:05.52 | c0rnoTa | You think so? |
| 17:05.55 | c0rnoTa | Why? |
| 17:06.05 | *** join/#asterisk sp3 (~tom@li180-184.members.linode.com) |
| 17:06.15 | WIMPy | Looks liek there might be soemthing wrong with 2.6. |
| 17:07.20 | c0rnoTa | WIMPy: yes, you are right. I should to search something wired in jira before posting an issue. |
| 17:08.02 | c0rnoTa | May be it was already founded and fixed. And I haven't looked in JIRA before. It's my mistake. |
| 17:12.05 | pabelanger | c0rnoTa: which HDLC Abort message? |
| 17:12.24 | c0rnoTa | pabelanger: number 6, you know |
| 17:13.24 | c0rnoTa | pabelanger: PRI got this event about HDLC Abort (code number 6) on D-channel of my span to PSTN |
| 17:13.27 | pabelanger | c0rnoTa: odd, I just recently ran across that message the other day. Do you D-channel reset after it? |
| 17:13.46 | pabelanger | EG: go into red alarm |
| 17:14.09 | pabelanger | s/you/your |
| 17:14.33 | c0rnoTa | pabelanger: no, just messages. and worse fax statistic |
| 17:14.49 | c0rnoTa | s/statistic/stats |
| 17:15.00 | pabelanger | roger |
| 17:16.01 | c0rnoTa | pabelanger: I have wrote about it in chat recently. May be it was my messages about another server. |
| 17:16.17 | c0rnoTa | pabelanger: you should to know that another server has dahdi 2.6.0 |
| 17:16.31 | c0rnoTa | I'll downgrade it tomorrow |
| 17:19.29 | SeRi | p3nguin: you fax via voip.ms? |
| 17:20.38 | *** join/#asterisk jkroon (~jkroon@dsl-244-35-235.telkomadsl.co.za) |
| 17:20.49 | p3nguin | Yes I do. |
| 17:22.18 | SeRi | p3nguin: do you use a regular fax machine? |
| 17:22.25 | p3nguin | Yes. |
| 17:22.31 | p3nguin | attached to an ATA |
| 17:22.40 | SeRi | any special settings? |
| 17:22.41 | p3nguin | SPA 3102, to be exact. |
| 17:22.43 | p3nguin | No. |
| 17:22.55 | p3nguin | Just make sure it uses ulaw. |
| 17:23.01 | SeRi | ok cool. I am going to give it a spin |
| 17:23.07 | p3nguin | It is a little slow, but still works. |
| 17:23.08 | SeRi | what is the rate failure you experience? |
| 17:23.18 | p3nguin | I haven't had one fail yet. |
| 17:23.25 | SeRi | nice. |
| 17:24.09 | p3nguin | For fax reception, I use ReceiveFax. |
| 17:24.44 | SeRi | under asterisk? |
| 17:24.45 | p3nguin | I send with the fax machine, though. |
| 17:24.48 | p3nguin | Yes. |
| 17:24.56 | p3nguin | I use fax for asterisk. |
| 17:25.05 | p3nguin | res_fax and res_fax_digium |
| 17:25.16 | SeRi | I do the same thing for fax reciving but I need to figure out an option for fax sending |
| 17:25.24 | p3nguin | Fax machine! |
| 17:25.30 | SeRi | :P |
| 17:25.37 | SeRi | will try it now |
| 17:25.40 | p3nguin | Or you can use a scanner and scan your doc, then send with SendFax. |
| 17:25.56 | p3nguin | If you have a networked scanner, that would be simple enough. |
| 17:26.26 | SeRi | I dont :( I have an old HP all in one... and i mean old! |
| 17:27.02 | p3nguin | You can still scan it in, then send the image with asterisk. |
| 17:27.38 | p3nguin | It would be easy if you have samba configured on the asterisk system. |
| 17:28.11 | SeRi | I see. |
| 17:28.19 | SeRi | well Mhhhhh |
| 17:28.26 | p3nguin | But if you have a fax machine and an ATA, you're ready to fax as is. |
| 17:28.43 | SeRi | got it. I have both |
| 17:29.28 | SeRi | I guess ill send my self a fax to test |
| 17:29.51 | p3nguin | You can send it to me and I can send it back, if you want. |
| 17:30.02 | SeRi | perfect. Thanks! |
| 17:30.08 | SeRi | msg me |
| 17:30.10 | jkroon | hi guys, having some trouble with the Pickup application under ast 1.8.7.2 when trying to use the PICKUPMARK option |
| 17:31.07 | jkroon | http://pastebin.com/5UXP3ZP4 - it simply states "No target channel found for foobob", however, the pasted dialplan clearly shows that PICKUPMARK is being set on the channel, and core show channels shows that the channel does in fact enter the Ring state (allowing it to be picked up) |
| 17:31.09 | *** join/#asterisk libryder (~david@209.33.214.243) |
| 17:31.42 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 17:33.10 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:33.45 | p3nguin | I have two systems on UPS. One on a smart UPS connected by a cable and one on a dumb UPS with no cable. Both computers are connected on the LAN. The one on the smart UPS is using apcupsd and is configured to shutdown when the battery is almost empty. I want to use the one on the smart UPS to tell the other one via SNMP to shut down. Should I be looking at nutups or net-snmp to accomplish it most easily? |
| 17:33.54 | libryder | is there a way to tell what bitrate the voice channel on a call is set at? |
| 17:34.32 | paulc | libryder: you can do a "core show channel xxxxx" while the call is in progress.. or did you mean within the dialplan? |
| 17:34.35 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 17:35.05 | libryder | yeah that is perfect paulc thanks |
| 17:36.59 | jkroon | nm ... Set(_PICKUPMARK, not Set(PICKUPMARK ... |
| 17:37.36 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 17:37.37 | p3nguin | Ah, a channel variable inheritance issue. |
| 17:38.23 | libryder | paulc: it looks like core show channels is cutting off part of the channel name |
| 17:38.33 | *** join/#asterisk mahlon (mahlon@martini.nu) |
| 17:38.41 | p3nguin | core show channel <channel name here> |
| 17:38.46 | p3nguin | core show channels concise |
| 17:40.08 | p3nguin | incoming fax! |
| 17:40.20 | paulc | T.38 beeee BEEEEEEEEE brrrrbrbrrrbrbrbrbrrrrbrr |
| 17:40.25 | p3nguin | haha |
| 17:40.37 | paulc | love that sound :) |
| 17:40.41 | p3nguin | I can't hear it; it's within asterisk. |
| 17:40.43 | libryder | thanks p3nguin |
| 17:41.11 | paulc | times like that you need a trader-style turret that you can turn the volume up and down on.. that'd be cool :) |
| 17:41.21 | p3nguin | Fax failed. |
| 17:41.46 | *** join/#asterisk bluregard (~matt@c-24-15-36-70.hsd1.il.comcast.net) |
| 17:41.49 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 17:41.54 | bluregard | hello all |
| 17:43.38 | paulc | hello bluregard |
| 17:44.39 | bluregard | I'm having an issue using AMD() with manager originates. When the called party answers and the originate bridges the call with the extension specified which starts with AMD(), it takes about 2 seconds to actually call AMD(), so when a person answers and says hello, the AMD() app hasn't been called yet so every call is determined to be a machine. Any advice on this? |
| 17:45.18 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 17:46.05 | *** part/#asterisk libryder (~david@209.33.214.243) |
| 17:47.01 | bluregard | from what I can tell, its taking too long to register with asterisk that the originate has been answered. |
| 17:47.06 | paulc | bluregard: hmm.. that sounds odd.. if you replace AMD() with Playback(beep) - do you hear the beep immediately or 2 seconds after answering? |
| 17:47.20 | paulc | bluregard: also, what technology/channel are you using for the outbound call? |
| 17:47.25 | bluregard | paulc: let me try real quick. |
| 17:47.33 | bluregard | paulc: sip |
| 17:48.49 | bluregard | paulc, the beep comes right away, almost before I can get the phone up to my ear. |
| 17:48.52 | leifmadsen | note that the originate works by first waiting for the other end to answer, then connects to the dialplan/extension configured for the other side of the originate. It's possible the audio you're getting is early media and it takes a bit for the other side to "answer" the call. You could originate to the dialplan and answer the call immediately, then dial the other side. The immediate Answer() should trigger the other chan |
| 17:48.52 | leifmadsen | nel to start listening for AMD() right away. |
| 17:49.06 | leifmadsen | problem though -- it'll mess with your CDRs if you care |
| 17:54.21 | bluregard | yeah, I'm relying on CDR for a lot of stuff with this |
| 17:55.00 | bluregard | I seem to be having a problem with audio coming into asterisk again. It was just working perfectly but stopped working again. |
| 17:55.34 | paulc | hmm.. intermittent problems are a pain in the ass |
| 17:55.47 | bluregard | yes sir |
| 17:56.20 | bluregard | I have a feeling its nat causing problems again, but I'm not sure yet. |
| 17:56.40 | *** join/#asterisk twanny796 (~twanny@78.133.79.159) |
| 17:57.23 | *** join/#asterisk ccesario (~ccesario@189.29.54.144) |
| 18:02.27 | paulc | slap that router round and show it who's boss :-) |
| 18:02.58 | bluregard | yeah, part of this problem seems to be intermittent problems with audio coming into asterisk |
| 18:06.06 | bluregard | hmm, I wonder if its my rtp port forwarding on my router... |
| 18:12.44 | bluregard | BAH, I didn't realize asterisk doesn't use 10000:20000 for rtp by default. It was using rtp ports that weren't being forwarded by the firewall. |
| 18:13.27 | [TK]D-Fender | SMRT |
| 18:13.28 | [TK]D-Fender | :) |
| 18:14.29 | bluregard | :-p |
| 18:22.29 | p3nguin | The default rtp.conf does use 10000-20000, but without that being set, there is a different port range. |
| 18:23.06 | bluregard | the rtp.conf.sample does yes, but * by default does not. |
| 18:23.38 | p3nguin | That's what I said. |
| 18:24.10 | p3nguin | I never really understood that. |
| 18:24.38 | p3nguin | You'd think the default range configured in the sample file would correspond with the built-in default. |
| 18:25.32 | bluregard | yeah, that would make sense to me |
| 18:29.52 | drmessano | Are the defaults 5000 to 31000? Never thought to look |
| 18:30.24 | drmessano | No, bad google hit |
| 18:30.31 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-usnvlfdyuckalxnw) |
| 18:30.38 | leifmadsen | would be easier to check for sure in the code |
| 18:31.51 | wdoekes2 | 5K and 31K indeed |
| 18:31.59 | drmessano | Yep |
| 18:32.17 | drmessano | Not sure why I didn't just google for rtp.c from the start lol |
| 18:32.21 | leifmadsen | :) |
| 18:32.34 | leifmadsen | rtp.conf.sample should really be updated to reflect the true defaults |
| 18:33.00 | drmessano | So what happens when asterisk negotiates port 5060 for rtp? |
| 18:33.12 | drmessano | or 5038 |
| 18:33.33 | wdoekes2 | 5038 isn't used for udp, afaik, and 5060 is already bind, so not available |
| 18:33.36 | wdoekes2 | *bound |
| 18:34.24 | drmessano | So asterisk knows not to try to use 5038 and 5060, which are well within that 5k to 31k range, for RTP? |
| 18:34.46 | p3nguin | I'd say it would be able to use those ports if they are not already in use. |
| 18:34.59 | wdoekes2 | it will try, but fail and try a different port instead |
| 18:35.05 | drmessano | ah |
| 18:35.29 | *** join/#asterisk oliver1 (~oliver@manz-5f748128.pool.mediaWays.net) |
| 18:36.23 | rossand | Anyone know a trustworthy SIP or IAX2 provider with toll free DIDs in the UK, France, Finland, Sweden, and Germany? |
| 18:37.39 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 18:37.49 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 18:38.26 | autofsckk | hello everybody, any suggestions on a VPS for asterisk? i need a small one just for testing now, but i want to be able to make if bigger if i need too |
| 18:38.54 | p3nguin | linode is pretty cheap and works good for asterisk. |
| 18:39.22 | autofsckk | thanks p3nguin |
| 18:39.54 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 18:39.59 | autofsckk | p3nguin: have you make any changes to your p3nguin-aur-asterisk version? :D |
| 18:41.09 | autofsckk | well not aur really, i mean your version package of asterisk |
| 18:41.22 | p3nguin | Yes. |
| 18:42.05 | p3nguin | I should have 1.8.8.0 up there now. |
| 18:42.44 | autofsckk | can you share it with me please? |
| 18:43.55 | p3nguin | One moment. |
| 18:44.25 | autofsckk | thankx |
| 18:46.48 | p3nguin | See /notice |
| 18:48.16 | autofsckk | ok |
| 18:48.45 | autofsckk | thanks a lot |
| 18:48.58 | autofsckk | p3nguin: do you have an asterisk running on linode? |
| 18:49.07 | autofsckk | i mean, arch |
| 18:49.26 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 18:49.51 | p3nguin | I do have an asterisk on linode, but it's on CentOS rather than Arch Linux. |
| 18:51.03 | p3nguin | I run my own stuff on Arch. |
| 18:51.06 | WIMPy | rossand: No IAX, but sipgate should be a good start. |
| 18:51.39 | *** join/#asterisk AliRezaTaleghani (~taleghani@unaffiliated/AliRezaTaleghani) |
| 18:51.56 | *** join/#asterisk thomas (tm@tm.muc.de) |
| 18:51.57 | thomas | hiho |
| 18:52.03 | AliRezaTaleghani | hi all |
| 18:52.08 | thomas | is it posible to set an linebreak for "CALLERID" (name) ? |
| 18:52.31 | p3nguin | Sounds like a terrible idea. |
| 18:52.42 | AliRezaTaleghani | I have some problem with pthread time manager on asterisk 1.8 |
| 18:52.43 | p3nguin | I might even reject a call that does that. |
| 18:52.59 | rossand | thanks WIMPy, much appreciated. |
| 18:53.00 | AliRezaTaleghani | as i send about 30calls peer second with sipp to my server |
| 18:53.05 | thomas | p3nguin: me? |
| 18:53.36 | AliRezaTaleghani | and when the core channels goes over 120, it will crach |
| 18:53.37 | p3nguin | thomas: Right. That sounds like a terrible thing to do, and if I received a call that did that, I might even reject it immediately. |
| 18:53.49 | AliRezaTaleghani | with pthream_mutex_lock() |
| 18:53.52 | thomas | p3nguin: hm, then you have maybe a otheridea |
| 18:54.03 | thomas | my aastra-phone show only 14 digits |
| 18:54.09 | thomas | 1234567890abcd |
| 18:54.11 | thomas | but i need more |
| 18:54.15 | rossand | WIMPy: Nope. But thanks anyway. |
| 18:54.22 | thomas | i will show the companyname and maybe the name of person |
| 18:54.25 | p3nguin | You want the caller id name to show more characters? |
| 18:54.29 | thomas | like "Asterisk AG - Thomas Mebes" |
| 18:54.37 | thomas | p3nguin: jeppa |
| 18:55.08 | WIMPy | rossand: nope what? |
| 18:55.08 | *** join/#asterisk rjvvliet (~rjvvliet@178-85-122-187.dynamic.upc.nl) |
| 18:55.32 | rossand | WIMPy: sipgate does not do international DIDs. |
| 18:56.08 | WIMPy | rossand: The do everal countries. |
| 18:56.16 | thomas | p3nguin: jep, i want |
| 18:56.53 | p3nguin | thomas: I can't really think of how to override a phone's display limitations. |
| 19:01.29 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:09.58 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 19:12.37 | leifmadsen | p3nguin: thomas: only way I can think of is when you call the device, that you use something like a microbrowser that updates the screen and have the microbrowser actually show what is going on |
| 19:12.55 | leifmadsen | but that is hacky for sure -- there won't be a way to override the maximum values a phone supports/accepts |
| 19:24.27 | p3nguin | > 13:22:16 |
| 19:24.52 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 19:31.05 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 19:32.14 | sawgood | If an Asterisk box has a context in sip.conf .. and the host=ip address ... does this peer need to use a username/password as well (or is this optional) |
| 19:32.36 | sawgood | One Asterisk box is registering to another Asterisk box as a 'friend' |
| 19:32.48 | p3nguin | You lost me at "context in sip.conf." |
| 19:32.54 | sawgood | I know ... I'm sorry |
| 19:33.07 | p3nguin | If you're asking if a peer is required to have a password, the answer is no. |
| 19:33.14 | sawgood | ok ... Asterisk box A = has an account on Asterisk box B (inside of sip.conf) |
| 19:33.43 | sawgood | What if A sends a username/password but that information is NOT in sip.conf on Asterisk B? |
| 19:35.13 | sawgood | I guess does the usage of host=ip address (cut off any/all need to use any username/password even if the exist)? |
| 19:36.15 | p3nguin | host=ip-address only turns the peer entry into a peer entry that does not accept registrations. For a peer that registers to you, you must use host=dynamic. |
| 19:36.42 | sawgood | cool ... so username/passwords are only used IF host=dynamic then? |
| 19:36.44 | p3nguin | It has nothing to do with passwords. |
| 19:37.11 | p3nguin | Passwords are used to authenticate, host is used to match. |
| 19:37.20 | sawgood | nice! |
| 19:37.38 | sawgood | So, how does one require both host=IP address and authentication in sip.conf? |
| 19:37.51 | [TK]D-Fender | ... |
| 19:37.58 | [TK]D-Fender | have a secret. |
| 19:37.59 | p3nguin | If the host line has an IP address and you get a call from a different IP address, that peer entry does not match. End. |
| 19:38.24 | sawgood | perfectly understood |
| 19:38.25 | p3nguin | If you also want to require password authentication, use secret=somesecretpassword. |
| 19:38.36 | sawgood | what about username? |
| 19:38.48 | p3nguin | It is what you make it. |
| 19:39.05 | sawgood | so, can one have host=ip and just a secret and not a username? |
| 19:39.21 | p3nguin | The peer name is the peer's name. |
| 19:39.30 | sawgood | oh ... very cool ... |
| 19:39.34 | p3nguin | [peer-name] |
| 19:39.38 | sawgood | thank you for helping, p3nguin |
| 19:40.02 | sawgood | sometimes, I see the usage of username inside of a lot of sip.conf files, so I thought it was required for authentication |
| 19:40.39 | sawgood | So, for security ... a host=ip address and a secret=something would be best? |
| 19:40.46 | p3nguin | http://pastebin.com/Ag7tknm2 |
| 19:41.31 | sawgood | ty! |
| 19:43.13 | sawgood | another thought ... what is client B keeps sending a 'registration' statment to server A, but server A does not have host=dyamic |
| 19:43.43 | sawgood | These darn 'error' statments are bothering me ... I do not have control over the other side of the connection (sending me the registration statement) |
| 19:43.49 | p3nguin | You'll have to fix client B, because server A will complain about the misconfiguration. |
| 19:44.00 | bluregard | does AMD() read amd.conf every time its called, or is amd.conf read at load time? |
| 19:44.06 | sawgood | and it does complain! |
| 19:44.28 | p3nguin | If it is sending you registrations and you cannot control it, you will HAVE to change your entry to host=dynamic to accommodate its registration attempts. You have no other choice. |
| 19:44.58 | sawgood | thank you for the help, p3nguin that is what I did for now until I hear back from the admin of the other box |
| 19:44.58 | p3nguin | And then, I recommend you use an ACL in addition to the dynamic host configuration. |
| 19:45.27 | p3nguin | It's not really a problem to accept registrations. Configure your side accordingly and move on. |
| 19:45.27 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 19:45.28 | sawgood | ACL in a firewall process or with Asterisk? |
| 19:45.39 | p3nguin | I meant in asterisk, but you can do both. |
| 19:45.53 | sawgood | I'm using pfSence right now |
| 19:46.07 | p3nguin | You create the ACL with a deny and permit pair in the peer entry. |
| 19:46.18 | sawgood | ty |
| 19:46.44 | p3nguin | deny=0.0.0.0/0.0.0.0 permit=206.158.99.18/255.255.255.224 for example. |
| 19:47.01 | sawgood | nobody gettin by that rule! |
| 19:47.24 | p3nguin | If the peer's address never changes, use subnet mask 255.255.255.255 instead. |
| 19:48.21 | p3nguin | If it can change, determine the netblock it uses and set the subnet mask accordingly. |
| 19:49.27 | p3nguin | The ACL in asterisk keeps certain people out, but a firewall should be used to keep out most all others. |
| 19:52.41 | sawgood | cool |
| 19:54.05 | sawgood | If client B is set and working on Server A ... (and since host=ip address) is set (leaving the connection as unmonitored) ... outside of a qualify=yes statement, is there any way to check from server A if client B is still 'there' and working'? |
| 19:54.37 | p3nguin | The host= line has nothing to do with it being unmonitored. |
| 19:54.45 | p3nguin | That's all on the qualify. |
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| 20:38.16 | fireman_biff | hi, how do you change the amount of time asterisk waits for you to press a digit before it starts to dial? |
| 20:38.58 | WIMPy | fireman_biff: Are you sure it's Asterisk and not your phone? |
| 20:39.53 | p3nguin | If you're talking about an IP phone, that is controlled by the phone. |
| 20:40.19 | fireman_biff | k, thanks guys |
| 20:45.39 | fireman_biff | WIMPy & p3nguin: is this the case for analog phones also? |
| 20:46.06 | fireman_biff | I'm seeing a command TIMEOUT(digit) that seems like what I want, but i was hoping for a setting |
| 20:48.13 | WIMPy | fireman_biff: Connected how? If via an ATA then that will decide when to dial. |
| 20:48.51 | fireman_biff | WIMPy: the pbx connects directly to regular analog lines |
| 20:49.03 | WIMPy | Wrong end. |
| 20:49.15 | fireman_biff | analog on both ends |
| 20:49.20 | WIMPy | It's beween the phone and Asterisk. |
| 20:50.42 | WIMPy | From Asterisk to the PSTN there is no timeout, but with analog a delay. |
| 20:50.43 | fireman_biff | there are FXS cards in the PBX if that's what you mean |
| 20:50.54 | WIMPy | ok |
| 20:51.13 | WIMPy | Yes, then it depends on your chan_dahdi.cfg and the dialplan. |
| 20:52.12 | fireman_biff | is there a setting for chan_dahdi.cfg that specifically handles the timeout? |
| 20:53.10 | WIMPy | I'm not sure about that one. But there is immediate and alwaysimmediate that decide if it's chan_dahdi that collects the digits and imposes the timeout or the dialplan. |
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| 20:54.17 | fireman_biff | alright, I'll see what I can find out about those settings |
| 20:54.19 | fireman_biff | thanks |
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| 21:22.40 | *** join/#asterisk CGMChris (~chatzilla@74.143.228.142) |
| 21:24.17 | CGMChris | I'm trying to configure an SPA504g in a 1-employee branch office so that it will connect to the main office PBX. When it dials an internal extension, it's specifying that the audio stream is on 10.0.0.X port Y...anyone familiar with this and how to get it to use its external IP in SIP packets? |
| 21:24.49 | Kobaz | is there a way to allow sip calls to localhost |
| 21:25.00 | Kobaz | [2012-01-25 16:23:54] -- <Local/1@_SipBasicTest-be12;1> Got SIP response 482 "Loop Detected" back from 192.9.200.189:5060 |
| 21:25.12 | Kobaz | when i dial a sip peer with host = localhost |
| 21:25.23 | Kobaz | i can loop back with iax2, but not sip |
| 21:25.42 | Lipsum | We're experiencing some deadlocks on 1.4.42: http://pastebin.com/aGpwFxCT and http://pastebin.com/M1F1uxw2 - we've looked around but we cannot find anything useful. Any ideas on what to look at in order to solve these? |
| 21:25.44 | p3nguin | cgmchris: Configure asterisk correctly for NAT and never configure an end point to try to do its own NAT traversal. |
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| 21:26.59 | CGMChris | p3nguin: The phone is at one office on one internet connection. The PBX is at another office on another internet connection. No VPN. Doesn't the remote SPA504g need to identify itself by its proper external IP address for audio to work properly? |
| 21:27.23 | p3nguin | cgmchris: Configure asterisk correctly for NAT and never configure an end point to try to do its own NAT traversal. <----------------------- |
| 21:29.16 | CGMChris | p3nguin: Not helpful. Any more insight? I don't see how NAT has anything to do with configuring a phone to connect to what is essentially a hosted PBX. |
| 21:29.28 | p3nguin | I've told you how to fix it. |
| 21:29.49 | p3nguin | It should take you no more than two minutes to complete. |
| 21:30.43 | CGMChris | p3nguin: One of us doesn't get it, lets say it's me since you're the pro at this.... can you throw me a bone? some type of config setting I need in sip.conf or what? |
| 21:31.08 | CGMChris | p3nguin: Asterisk and the phone arent on the same internet connection... |
| 21:31.22 | p3nguin | In sip.conf, in the general section, set nat=yes to enable asterisk to work with natted devices. |
| 21:31.54 | p3nguin | And make sure the peer entry for that phone does not say nat=no, or it will negate the nat=yes setting in the previous step. |
| 21:32.15 | p3nguin | Save the change to the file. Run "sip reload" on the asterisk CLI. |
| 21:32.35 | p3nguin | Then try another call. |
| 21:33.03 | CGMChris | I've got nat=route, I'll try nat=yes....changed this when I was debugging T.38 w/ broadvox a year or so ago. Let me try again. |
| 21:33.39 | p3nguin | Also look for the nat=no in the phone's peer entry and remove it if it exists. |
| 21:39.42 | navaismo | its possible to connect a microphone and speakers to the machine running asterisk and register to another asterisk as normal phone?? |
| 21:40.13 | p3nguin | Yes. |
| 21:41.01 | becca_r | it would be easier to use xlite or a softphone to connect to the other Asterisk box though. |
| 21:41.34 | navaismo | sure, |
| 21:42.19 | navaismo | but the isea is use an a small appliance like ALIX boards and use it like a phone |
| 21:42.45 | navaismo | s/isea/idea/ |
| 21:43.40 | navaismo | so im looking if its possible install asterisk under uClinux and redirect the media to the audio hardware |
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| 21:45.35 | navaismo | do I'm telling nonsenses ? |
| 21:46.36 | becca_r | I haven't done it personally, but I would say you could. |
| 21:46.52 | WIMPy | There are several console channels that will do that. |
| 21:48.26 | navaismo | hmmm ok, i will search a lot, thanks anyway |
| 21:49.38 | navaismo | i think its "easier"install asterisk on uClinux tha continue doing this: http://www.youtube.com/watch?v=9hQsDSy8sPY&list=UUS_hMfj5LyW81I-AQ8B-dOQ&index=1&feature=plcp |
| 21:50.00 | navaismo | trying to insert the IAX2 library to the PICmicro |
| 21:51.11 | WIMPy | Asterisk is a pretty big thing. |
| 21:52.14 | navaismo | yes |
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| 22:04.36 | p3nguin | I don't understand why sometimes my asterisk is started by root rather than by asterisk. |
| 22:05.49 | WIMPy | How often do you start it? |
| 22:06.43 | p3nguin | Depending on what I am doing, it could be one time each 30 days, or 30 times in one day. |
| 22:07.09 | p3nguin | Today, it is several times because I am trying to determine why it is sometimes started as root and other times as asterisk. |
| 22:07.37 | WIMPy | But if YOU start it, YOU should know how. |
| 22:08.01 | p3nguin | I don't know what your statement is supposed to mean. |
| 22:08.53 | WIMPy | If it can be one way or another there must be some difference in how it is started, |
| 22:09.55 | p3nguin | There are two ways my asterisk can be started: I can run /etc/rc.d/asterisk start on the command line (as root), or a cron job can run the same script if asterisk it not already running. |
| 22:11.40 | p3nguin | Okay, I think I have solved it. |
| 22:12.31 | p3nguin | That was bothersome for a moment. |
| 22:13.26 | p3nguin | Apparently there is no default runuser and rungroup, despite documentation indicating the default is asterisk:asterisk. |
| 22:14.04 | p3nguin | So when I ran the script (as root) to start asterisk, root ran asterisk. When the cron job started asterisk (asterisk's crontab), asterisk owned the process. |
| 22:14.50 | p3nguin | I just needed to set the runuser and rungroup, and now starting asterisk with the script (as root), asterisk owns the process like it should. |
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| 22:41.33 | Lipsum | We're experiencing some deadlocks on 1.4.42: http://pastebin.com/aGpwFxCT and http://pastebin.com/M1F1uxw2 - we've looked around but we cannot find anything useful. Any ideas on what to look at in order to solve these? |
| 22:41.34 | Lipsum | It looks like two of my threads are waiting for line 5021 (http://pastebin.com/XVNvGREf), but I cannot see which thread is locking 5021? Am I missing something? |
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| 23:12.37 | paulc | Did JabberSend() in Asterisk 1.6.x become something else in Asterisk 10? |
| 23:13.49 | p3nguin | Does "core show applications like jabber" show anything? |
| 23:14.07 | p3nguin | How about "core show applications like xmpp"? |
| 23:14.25 | paulc | Nope.. 0 Aplications matching.. for both jabber and xmpp |
| 23:14.44 | p3nguin | Did you install and load the relevant modules? |
| 23:16.02 | paulc | Hmm.. I *thought* so.. but possibly not.. going from the instructions for our 1.6.x build, in make menuselect I enable func_curl in dialplan functions section, and res_jabber in the resource modules section. |
| 23:16.12 | paulc | Upon reflection, perhaps that second bit didn't happen.. |
| 23:16.34 | paulc | ah no, there it is.. [*] res_jabber |
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| 23:32.14 | malconxx | please send traffic to asteriskvoip.zapto.org username=003, passwd=123456, codec=g729 |
| 23:32.38 | navaismo | joke right |
| 23:34.38 | paulc | Problem solved.. iksemel library couldn't be found, so res_jabber.so wasn't loading - life continues... |
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| 23:58.45 | *** mode/#asterisk [+o ChanServ] by pratchett.freenode.net |