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| 00:31.54 | tladuke | is there a way to not have to define every device in sip.conf? i want to base device names off mac addresses or IP address or hostnames... |
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| 00:33.12 | WIMPy | 1. no, 2. and 3. only if you don't need authentication. |
| 00:34.18 | tladuke | i don't. or i can use the same password for everything. it's all on an isolated LAN |
| 00:36.11 | tladuke | doesn't this file get big if you have like a million customers? |
| 00:38.39 | WIMPy | If you don't need auth, you don't need to define any peers. You can just dial the hostname or IP. |
| 00:39.26 | WIMPy | But make sure noone untrusted has access to that LAN then. |
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| 00:52.37 | tladuke | thanks |
| 01:12.54 | tladuke | now if i can just figure out no auth |
| 01:13.08 | *** join/#asterisk fireman_biff (~biff@65.48.222.203) |
| 01:15.35 | WIMPy | enable guests |
| 01:15.48 | fireman_biff | Hi, I'm using "Goto(from-internal-mycontext,5551234,1)" to connect inbound callers to an external number, and thats working fine. How can I redirect the caller to voicemail if the external number doesn't answer after 20 seconds? (asterisk 1.8) |
| 01:16.37 | WIMPy | You need to do that after you Dial(). |
| 01:20.10 | fireman_biff | k, thanks |
| 01:21.09 | fireman_biff | is the from-internal context in standard asterisk or is that added by freepbx? |
| 01:21.40 | WIMPy | ~freepbx |
| 01:21.40 | 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 |
| 01:22.03 | fireman_biff | does that mean its not standard asterisk? |
| 01:23.37 | WIMPy | There is no standard Asterisk Dialplan. |
| 01:24.13 | fireman_biff | cool, thanks |
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| 01:51.50 | jeffspeff | can you use variables in a context name? |
| 02:06.12 | WIMPy | Where? |
| 02:06.36 | WIMPy | In the definition probably not. In a Goto or the like, yes. |
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| 03:12.28 | rx4change | has anyone seen an issue with realtime voicemail when the database connection times out? |
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| 03:20.07 | ledoktre | hey guys quick question (elementry Im sure, but dont want to miss this). When a call comes in, the person calling is the caller, i am the callee. if I park the call, and then later pick it up, I assume I still am the callee, right? I mean, I did dial the parked extension.... Just want to clarify. |
| 03:27.53 | jpsharp | rx4change: Another guy in here earlier today was complaining about that. Voicemails were getting eaten if there was a database problem. |
| 03:29.26 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 03:29.39 | fling | [Oct 25 10:20:53] NOTICE[5944]: chan_sip.c:20959 handle_response_peerpoke: Peer 'xxx' is now Lagged. (2124ms / 2000ms) |
| 03:29.59 | fling | how to increase the timeout to 4000ms ? ^ I have a lot of this in the console |
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| 03:32.35 | jpsharp | Set qualify=4000 in the sip.conf entry for the device. |
| 03:33.10 | jpsharp | but also, ask yourself, why is there a 4 second lag to your phones? Are you running a secret moonbase? |
| 03:33.19 | *** part/#asterisk fireman_biff (~biff@65.48.222.203) |
| 03:33.58 | fling | jpsharp: am I running a moon base? yes. |
| 03:34.07 | fling | jpsharp: thanks :D |
| 03:36.30 | fling | jpsharp: now I will not see the lag in my log when astronauts are downloading torrents :p |
| 03:46.30 | jpsharp | Three simple letter: QOS :) |
| 03:50.51 | *** join/#asterisk fling (~fling@fsf/member/fling) |
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| 04:08.12 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 04:08.15 | cj | hey folks |
| 04:08.19 | cj | anything new and exciting? |
| 04:08.45 | cj | SerajewelKS: how've you been, sir? |
| 04:09.42 | cj | carrar: hey, my friend Rick might want to get set up with you guys |
| 04:19.46 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 04:21.04 | *** join/#asterisk devCloud (~devand@d108-172-107-16.bchsia.telus.net) |
| 04:32.30 | cj | should I have them go through me, or do you want to go direct with them? they're not multihomed, but they are directly connected to Island Network |
| 04:33.02 | cj | http://www.opalco.com/programs/island-network/ |
| 04:33.58 | cj | they provide me a redundant wireless link for when pogozone.com/mtbakercable.com hiccups |
| 04:34.23 | cj | I haven't gotten my HA scripts written yet, so it's currently a manual cut-over sort of thing. |
| 04:34.46 | cj | okay, back to trying to get this DSLAM set up... |
| 04:35.41 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 04:38.45 | jpsharp | I recommend a sledgehammer. |
| 04:38.53 | ChannelZ | I wanna be |
| 04:39.14 | ChannelZ | You sledgehammer |
| 04:39.23 | ChannelZ | s/You/Your/ |
| 04:39.26 | ChannelZ | whoopsie |
| 04:39.38 | jpsharp | infobot: Yay, you're back! |
| 04:40.08 | jpsharp | Dial(SIP/sledgehammer) |
| 04:40.59 | ChannelZ | got his 'So' 25th Anniversary box set today |
| 04:56.49 | *** join/#asterisk rroche (~rroche@201.140.135.226) |
| 04:58.06 | rroche | Hi I need to know the billableSeconds on a bridged outbound call, is there a way to get those? since CDR isn't being called and Unlink doesn't show any data, Asterisk -V 1.8.8.0 |
| 04:58.36 | rroche | I meant fired, instead of called |
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| 06:00.08 | cj | I've been fired instead of called |
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| 06:08.28 | iprouteth0 | anyone manage to setup BLF on the digium phones yet? |
| 06:08.39 | iprouteth0 | I've got it provisioning through DPMA so far |
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| 06:29.36 | fling | jpsharp: Peer 'moonbase' is now Lagged. (5233ms / 4000ms) hehe |
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| 07:14.50 | *** join/#asterisk din3sh (~din3sh@41.212.202.23) |
| 07:14.58 | din3sh | hey all |
| 07:15.59 | din3sh | Has anyone been able to configure CallCompletionRequest in * 1.8? |
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| 07:44.20 | bombev | hi any idea how to configure when extension not answer 10 seconds the call to be redirected to mobile phone? |
| 07:45.10 | ChannelZ | just use two sequential Dial()s, the first with a timeout |
| 07:45.31 | ChannelZ | Dial(SIP/Whateva,10) |
| 07:45.40 | din3sh | Has anyone been able to configure CallCompletionRequest in * 1.8? |
| 07:45.41 | ChannelZ | Dial(SIP/Mobile) |
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| 08:44.32 | carrar | cj |
| 08:44.35 | carrar | sounds good |
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| 09:22.08 | krotos | hi all |
| 09:24.03 | n0nac | hey |
| 09:30.57 | krotos | i'm using SMS cmd in asterisk for receive/send sms from and to sip phone that are capable to send sms. I've already set the sms service center |
| 09:31.15 | krotos | and created the relative exten in my dialplan. But, where i cand found the body of sms? |
| 09:34.54 | Rumbles | hmmmm, can anyone tell me more about this error when I'm trying to load the res_digium_phone module: |
| 09:34.56 | Rumbles | [2012-10-25 10:28:28] WARNING[29615]: loader.c:398 load_dynamic_module: Error loading module 'res_digium_phone.so': /usr/lib/asterisk/modules/res_digium_phone.so: undefined symbol: ast_msg_alloc |
| 09:35.34 | Rumbles | I've followed the guide on the digium website and found someone with a similar issue, but not the same, not sure what that error means though :/ |
| 09:36.12 | Rumbles | does it mean asterisk is missing support for something the digium phones require? |
| 09:36.44 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-ksicuzmoakhjnyfi) |
| 09:42.45 | kaldemar | Rumbles: which guide? |
| 09:43.57 | Rumbles | 1 sec |
| 09:44.05 | Rumbles | http://downloads.digium.com/pub/telephony/res_digium_phone/README |
| 09:45.32 | Rumbles | and from the mailing lists: |
| 09:45.32 | Rumbles | http://lists.digium.com/pipermail/asterisk-users/2012-June/272898.html |
| 09:45.53 | kaldemar | did you download the right version of the module? |
| 09:45.55 | Rumbles | the first issue was that avahi wasn't up to date, so I updated it and hit that issue |
| 09:47.05 | Rumbles | I believe so |
| 09:47.20 | Rumbles | I downloaded res_digium_phone-1.8.11_1.3.0-x86_32.tar.gz |
| 09:47.29 | kaldemar | you have 1.8.X version of asterisk, right? |
| 09:47.52 | Rumbles | uname -m shows i686 and using Asterisk 1.8.14.0 |
| 09:48.46 | Rumbles | from the guide that looks like the right version, does that seem right for i686 ? |
| 09:49.04 | din3sh | Has anyone been able to configure CallCompletionRequest in * 1.8? |
| 09:51.23 | kaldemar | Rumbles: my guess is that your version of asterisk is not 1.8.x-digiumphone. |
| 09:52.47 | kaldemar | Rumbles: regular 1.8 releases don't seem to support the digium phone module yet. |
| 09:53.18 | kaldemar | din3sh: maybe you should ask a precise question to get any answers. |
| 09:53.41 | Rumbles | so we have to install a complete different version of asterisk to make DPMA work? |
| 09:54.24 | n0nac | i have some problems calls drop aftear 5 mins in or out calls , i have checked the RTP timeouts and sip config. and seems to be all ok, log shows thats was hangup, some one any idea |
| 09:54.28 | kaldemar | Rumbles: seems so. |
| 09:54.38 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 09:55.08 | Rumbles | :-( |
| 09:55.10 | Rumbles | thanks though |
| 09:55.18 | n0nac | and i doesnt use NAT |
| 09:57.14 | n0nac | it starts when i upgraded from 1.6 to 1.8 |
| 10:00.15 | kaldemar | Rumbles: it's the "certified" version you need if you continue using 1.8.x. |
| 10:01.35 | din3sh | kaldemar:https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+%28CCSS%29 |
| 10:02.10 | kaldemar | din3sh: why are you giving _me_ that link? |
| 10:02.11 | din3sh | CallCompletionRequest is described as a means of callback I suppose, asking if anyone got this working |
| 10:02.23 | kaldemar | ~polls |
| 10:02.23 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 10:02.43 | din3sh | is there another way to implement callback on busy/no answer? |
| 10:03.10 | kaldemar | what's wrong with that approach? |
| 10:03.31 | *** join/#asterisk rbowles (~bb@75-147-63-201-NewEngland.hfc.comcastbusiness.net) |
| 10:04.31 | din3sh | I tried, not working, maybe I am doing it wrong, thats why I asked if anyone here got it working |
| 10:05.08 | kaldemar | ~ask |
| 10:05.08 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 10:15.05 | hurdman | hello folks |
| 10:17.28 | hurdman | [TK]D-Fender: i have found my problem |
| 10:17.49 | hurdman | arf not here ^^" |
| 10:20.05 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
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| 10:43.26 | jmls1 | I haven't got a grasp of groups. I need something simple, and for the life of me can't figure it out :) |
| 10:44.06 | jmls1 | I have 2 inbound queues. I want to know the number of calls in each queue |
| 10:44.34 | jmls1 | group(queue1)=inbound |
| 10:44.41 | jmls1 | group(queue2)=inbound |
| 10:45.08 | jmls1 | now, when the call is connected to an agent, I want to remove it from the inbound group |
| 10:45.11 | jmls1 | how do I do that > |
| 10:45.21 | kaldemar | jmls1: you don't need groups for that. "core show function QUEUE_WAITING_COUNT" |
| 10:47.13 | jmls1 | kaldemar: ah. the wrinkle is that I'm not actually usiung the Queue function :( |
| 10:47.40 | kaldemar | if you insist on using the group functions, set the group to something else in a subroutine that is executed with the gosub parameter for app Queue. |
| 10:48.04 | jmls1 | See above: we're not using the queue function |
| 10:48.06 | kaldemar | are you implementing queue-like functionality in dialplan yourself? |
| 10:48.24 | jmls1 | not in the dialplan - we're using jms to request an agent from our application |
| 10:48.32 | kaldemar | and then? |
| 10:48.46 | jmls1 | get a message back that says "yes, connect to agent on sip/1234" |
| 10:49.11 | jmls1 | which we then dial .. |
| 10:49.40 | kaldemar | Dial has U() and M(). |
| 10:50.06 | jmls1 | yeah. I know. :) What I don't know is how to clear the channel from the group ! |
| 10:50.48 | kaldemar | MASTER_CHANNEL will help you with setting variables for the calling channel. |
| 10:51.37 | kaldemar | you just set the GROUP() value to something different from "inbound" or what ever you use. |
| 10:52.39 | *** join/#asterisk sustav (~vpp@12.230.214.130) |
| 10:53.25 | jmls1 | yeah - I know what my problem is. I'm trying to keep a count of inbound calls. inbound queue calls as separate group_counts |
| 10:53.41 | jmls1 | so, I'm doing |
| 10:53.43 | jmls1 | group(queue1)=queuewaiting |
| 10:53.51 | jmls1 | group(queue1)=queuecalls |
| 10:53.57 | jmls1 | group(queue1)=inboundcalls |
| 10:54.03 | jmls1 | group(queue1)=totalcalls |
| 10:54.38 | jmls1 | so I can say group_count(totalcalls) or group_count(inboundcalls) |
| 10:54.39 | jmls1 | etc |
| 10:55.06 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 10:55.07 | jmls1 | and I was wanting to remove the queuewaiting group from the channel |
| 10:55.20 | jmls1 | sorry, remove channel from queuewaiting group |
| 10:55.26 | kaldemar | jmls1: "Each channel can only be member of exactly one group per <category>." |
| 10:56.00 | kaldemar | if you do all those sets for a channel, the only group the channel is really in is totalcalls as it is the last set. |
| 10:56.09 | jmls1 | bugger. yes. that's why I did it the other way round orginally |
| 10:56.24 | jmls1 | sorry. I was looking at the dialplan test from today. I originally had |
| 10:56.38 | jmls1 | group(waiting)=queue1 |
| 10:56.39 | *** join/#asterisk danfromuk (~IceChat77@2.27.12.9) |
| 10:56.47 | jmls1 | group(inbound)=queue1 |
| 10:56.47 | jmls1 | etc |
| 10:58.22 | jmls1 | so, if I set group(waiting)= would that remove the channel from the waiting category ? |
| 10:58.40 | kaldemar | so, when you have GROUP(queuewaiting)=queue1 set and wat to remove the channel from that group, do Set(GROUP(queuewaiting)=null) or something. |
| 10:58.48 | kaldemar | empty value might also do. test it. |
| 11:02.38 | *** join/#asterisk din3sh (~din3sh@41.212.203.179) |
| 11:04.20 | *** join/#asterisk VSpike (~johncc@37.48.226.6) |
| 11:06.31 | VSpike | If you have two SIP phones on a LAN using a single asterisk server, and one phone calls another, will the voice streams go direct between the phones or via the server? |
| 11:07.48 | jmls1 | see, this is where I'm getting confused |
| 11:08.13 | jmls1 | I have two calls: 1 on queue1 and 1 on queue2 |
| 11:08.13 | kaldemar | VSpike: depends on your configuration. |
| 11:08.33 | jmls1 | group_count(@waiting) only gives me a count of 1 |
| 11:08.51 | jmls1 | group_count(queue1@waiting) gives me a count of 1 |
| 11:09.07 | jmls1 | group_count(queue2@waiting) gives me a count of 1 |
| 11:09.18 | kaldemar | jmls1: "uses the channel's current group if not specifed" |
| 11:09.28 | kaldemar | jmls1: core show function GROUP_COUNT" |
| 11:09.55 | jmls1 | right, so how can I get a total count of the @waiting category ? |
| 11:10.22 | jmls1 | urgh |
| 11:10.29 | jmls1 | group_count_match ... |
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| 11:11.33 | jmls1 | GROUP_MATCH_COUNT(*@waiting) ? |
| 11:12.24 | jmls1 | Regex compile failed on: *# |
| 11:12.28 | drzed | re |
| 11:12.29 | jmls1 | ^Regex compile failed on: * |
| 11:12.45 | jmls1 | what's the regex for "everything" |
| 11:12.48 | kaldemar | try .* |
| 11:12.54 | VSpike | kaldemar: right, fair enough. I have a more specific question. I'm looking at Switchvox appliances, and we have two offices, both with ISDN 30. They have suggested a single 305 appliance at one office and a gateway appliance (1G102F) at the other... |
| 11:13.07 | jmls1 | looking better |
| 11:13.43 | VSpike | I'm wondering if a phone in the office without the 305 makes a call, and that call goes via the ISDN in that same office, will the voice stream go over the inter-office link twice, or direct from phone to gateway? |
| 11:13.53 | jmls1 | yay! |
| 11:13.56 | jmls1 | working |
| 11:14.00 | jmls1 | kaldemar: many many thanks |
| 11:14.02 | VSpike | AFAICT all the appliances here are asterisk boxes at heart |
| 11:16.30 | kaldemar | VSpike: probably twice. that really depends on the network setup. |
| 11:17.46 | VSpike | Yes, that's probably not ideal. I think we'd better paying extra to have two small appliances, one in each location. |
| 11:19.28 | kaldemar | jmls1: np |
| 11:19.50 | VSpike | That's basically what we have now, except they are old servers running AsteriskNOW |
| 11:20.02 | VSpike | Then we have an IAX trunk between the two sites |
| 11:27.07 | *** join/#asterisk Lantizia (~lantizia@cpc6-stok15-2-0-cust8.1-4.cable.virginmedia.com) |
| 11:28.03 | Lantizia | Any such thing as a T.38 proxy? e.g. you're receiving a T.38 fax from your ITSP to Asterisk and you want to send it to both a peer registered to t38modem (for fax to email) and also to a T.38 aware ATA to a real fax |
| 11:30.08 | *** join/#asterisk sertaconay (~sertacona@unaffiliated/sertaconay) |
| 11:32.09 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:36.12 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
| 11:52.12 | *** join/#asterisk hwt (hwt@curb.thorkildssen.com) |
| 11:52.43 | hwt | is there any way to set transport to TCP from an AGI app, and then just Dial/user@host:port with transport=tcp? |
| 11:53.17 | hwt | or do i need to have all SIP peers i need to talk to also defined in sip.conf? |
| 11:53.39 | hwt | my plan is to do dynamic routing of calls based on data I get from a REST service. |
| 11:56.03 | *** join/#asterisk din3sh (~din3sh@41.212.200.193) |
| 11:56.48 | *** join/#asterisk w9sh (~chatzilla@64.238.96.125) |
| 12:00.39 | kaldemar | hwt: try Dial(SIP/username::::tcp@host) |
| 12:01.54 | hwt | kaldemar: wow, really? i can try it, but what's the logic? |
| 12:02.24 | kaldemar | SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port] |
| 12:02.56 | kaldemar | that's from the sample config. |
| 12:02.57 | hwt | kaldemar: aha. this was not documented on http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 12:03.47 | kaldemar | it is not specific to Dial and voip-info is... not the best place to consider as documentation. |
| 12:04.06 | hwt | kaldemar: yeah, i know. should have just checked the sample config. |
| 12:04.30 | hwt | kaldemar: just curioust, is it also possible to set other sip peer stuff via the dialplan? for example nat=yes|no and dtmfmode? |
| 12:04.49 | hwt | kaldemar: actually, canreinvite=yes|no is most interesting, i guess. (proxy media or not) |
| 12:05.31 | jmls1 | wouldn't it be nice if WARNING[21616][C-00010b23]: app_dial.c:2433 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) actually told you which device it was trying to dial so that you could find and fix it ? |
| 12:05.52 | *** join/#asterisk aurs (~aurs@110.84-49-69.nextgentel.com) |
| 12:06.29 | kaldemar | hwt: afaik, no. |
| 12:07.05 | hwt | ah, shame. this is a bit of a weakness when using Ast as a B2BUA. especially compared to e.g. Sippy. |
| 12:11.05 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 12:11.08 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:11.15 | *** join/#asterisk aboven (~aboven@62.4.144.227) |
| 12:11.56 | jkroon | hi all, is there any way to capture the "Name" portion of REGISTER request? |
| 12:12.00 | aboven | No audio in early media, what can I do (asterisk 1.8) ? |
| 12:12.17 | hwt | jkroon: sure, just grab the SIP header? |
| 12:12.41 | hwt | jkroon: ${SIP_HEADER(from)} |
| 12:13.11 | hwt | aboven: nat=yes ? |
| 12:13.21 | aboven | yes |
| 12:13.23 | jkroon | It seems that a REGISTER request contains the actual endpoint being registered to in the To: field, I'd like to capture whatever is the name, eg: To: "SomeName" <sip:123@server> <-- I want "SomeName" stored somewhere for later use... |
| 12:14.05 | [TK]D-Fender | jkroon, No. |
| 12:15.11 | hwt | jkroon: should be the From AND To. but just grab it and parse out what you need. |
| 12:17.01 | *** join/#asterisk blee (~blee@72.188.117.219) |
| 12:18.25 | bombev | I have problem with me follow me configuration, I set up my number 03341233345# into follow me list my cell phone rings and when i try pick ip the call, it hangs up |
| 12:18.58 | bombev | here is the asterisk code pastebin.com/TGn8ra3B |
| 12:22.17 | *** join/#asterisk chris-NB (~chris@home.fuerstaller.com) |
| 12:23.10 | jkroon | hwt, REGISTER never enters the dialplan. |
| 12:23.21 | jkroon | [TK]D-Fender, thanks. |
| 12:23.35 | [TK]D-Fender | jkroon, Correct... |
| 12:24.48 | jkroon | [TK]D-Fender, just trying to explain to hwt why his idea won't work. |
| 12:25.02 | kaldemar | jmls1: that would be a minor addition to app_dial.c |
| 12:25.06 | [TK]D-Fender | jkroon, Sad that you have to.... |
| 12:25.37 | hurdman | [TK]D-Fender: perhaps found my bug between sandy bridge and digium card ... something with BUS speed and PCI-E version, after down to PCI-E v1 ( auto was v3 =) it's not perfect but better. |
| 12:25.41 | hwt | jkroon: yeah, sorry, i wasn't thinking clearly |
| 12:26.01 | jkroon | hwt, we all have our moments :p |
| 12:26.32 | jkroon | [TK]D-Fender, i have a client that want me to display the names configured on his phones on a page listing all registrations ... |
| 12:26.41 | jkroon | going to all the phones to fetch the info is just stupid. |
| 12:26.49 | *** join/#asterisk barkbus (~barkbus@2.176.43.195.dsl.static.ip.kpnqwest.it) |
| 12:26.49 | *** join/#asterisk n0nac (~canon999_@server.bent.si) |
| 12:27.05 | hurdman | jkroon: sip ? |
| 12:27.49 | hwt | jkroon: it's not pretty, but maybe you can do something with asterisk -rx "sip show users" and compare that to info from SIPPEER()? |
| 12:28.29 | jkroon | hurdman, yes. hwt, unfortunately not, and it would be sip show peers actually. |
| 12:28.37 | [TK]D-Fender | jkroon, I take the the phones are not centrally administered? |
| 12:28.51 | jkroon | but the name (as per [TK]D-Fender) isn't stored for the peer anyway. |
| 12:28.54 | hwt | jkroon: um, yes, that's what i meant. |
| 12:28.58 | jkroon | [TK]D-Fender, distributed over the country. |
| 12:29.12 | hurdman | jkroon: oui AMI ? |
| 12:29.16 | hurdman | with* |
| 12:29.18 | jkroon | and no, unfortunately not, nor can I enforce that. |
| 12:29.19 | [TK]D-Fender | jkroon, Yeah, with any luck multiple makes/models as well.. this is kinda dead... |
| 12:29.58 | jkroon | [TK]D-Fender, jip, mix with God-only knows what. I think the crappiest one I've seen on here recently was an IPT100H phone. Whatever that is. |
| 12:30.08 | jkroon | even had user agent strings that just said "phone" |
| 12:30.26 | [TK]D-Fender | jkroon, This is dead without hacking up chan_sip or shoving a proxy or something to capture it |
| 12:30.47 | jkroon | would such a patch potentially be useful to others in your opinion? |
| 12:31.48 | [TK]D-Fender | jkroon, We don't haev it yet and you're the first to mention it. |
| 12:32.01 | jkroon | ie: probably a no :p |
| 12:32.08 | [TK]D-Fender | jkroon, Peopler tend to actually be in control of their environment so I suppose this isn't an issue for many |
| 12:32.38 | jkroon | rofl, my demarkation to the client is that SIP channel, so I control the environment on my side, but not the other side. |
| 12:33.04 | jkroon | now i've got a reseller supporting some clients, and he'd like to be able to have that information. |
| 12:33.24 | *** part/#asterisk jacekowski (jacekowski@jacekowski.org) |
| 12:33.31 | jkroon | would be nice, but not the end of the world if I can't give that to him. |
| 12:36.20 | *** join/#asterisk n0nac (~canon999_@server.bent.si) |
| 12:36.35 | *** join/#asterisk sustav (vpp@conference/digium/x-mgwewpqknumvgohe) |
| 12:46.14 | n0nac | hwt: session-expires=refuse didnt helped was droped aftear duration=313 |
| 12:46.34 | hwt | n0nac: i would set up an ngrep trace, or use sip debug in asterisk |
| 12:46.44 | hwt | n0nac: see if there is a BYE involved or if it's just the media "dying" |
| 12:46.49 | n0nac | yes it is |
| 12:46.57 | n0nac | BYE |
| 12:47.05 | n0nac | so in log looks all normal |
| 12:47.11 | n0nac | hangup |
| 12:48.27 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 12:48.56 | hwt | n0nac: and from the other side? |
| 12:49.27 | hwt | n0nac: if so, i would say definitely session timers. |
| 12:51.20 | WIMPy | Session-timers seem to be evil. |
| 12:52.20 | WIMPy | jkroon: Did you spend time on that BRI issue or did you give up? |
| 12:52.55 | hwt | WIMPy: they are. |
| 12:53.59 | WIMPy | I have had random disconnects as well. |
| 12:53.59 | jkroon | WIMPy, client cancelled. |
| 12:54.08 | jkroon | got the machine in the office if you'd like to take a look. |
| 12:55.11 | WIMPy | jkroon: I'd surely like to find out what's going on there, but I think it's the external environment that's needed there. |
| 12:56.08 | WIMPy | Maybe I should just try to set up a test exnvironment with non syncronised lines and see if there is an issue when using dahdi then. |
| 12:56.20 | hwt | is it possible to set whether or not to relay media from the Ast diaplan? |
| 12:56.46 | [TK]D-Fender | that does not make sense. |
| 12:56.52 | jkroon | probably going to do my "monkey" test on it a little laster ... basically loop ports 1 to port 3, and 2 to 4, then start one one side and keep looping the call until there are no more channels, then I can choose, immediately hang up or keep the call on echo. |
| 12:57.02 | WIMPy | I have just found out that Astribanks seem to require sunced lines even for BRIs. |
| 12:57.35 | WIMPy | s/sunc/sync/ |
| 12:58.11 | WIMPy | I think the vendors need to clearly advise on that issue. |
| 12:58.25 | jkroon | specifically on the B410P card? |
| 12:58.32 | jkroon | using the DAHDI drivers? |
| 12:59.20 | WIMPy | No. The card should handle it. But I didn;t get a clear answer to my question if it could depend on the driver. So a test might be a good idea. |
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| 13:04.11 | n0nac | hwt: i have now set all timers to refuse session-timers=refuse , session-expires=refuse now drops at 6 mins :D |
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| 13:13.55 | jkroon | WIMPy, last you got? |
| 13:17.32 | jkroon | i saw your statement re also getting disconnects |
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| 13:35.49 | *** join/#asterisk troutowi (~troutowi@141.216.135.29) |
| 13:37.28 | troutowi | quick question, will there be any patch updates to 11 rc2 once the final release is out? |
| 13:37.46 | troutowi | meeaning will there be an automated update for it |
| 13:41.50 | [TK]D-Fender | Patch should only take you from the release before... |
| 13:42.05 | troutowi | ok |
| 13:42.21 | troutowi | and [TK]D-Fender, so ive had more issues lol ^_^ |
| 13:42.46 | troutowi | just compiling fresh from source with 18 or 10, and trying to start asterisk gives me the core dumps |
| 13:43.43 | troutowi | 1.8* |
| 13:45.38 | troutowi | trying again with the dont_optimize flag set |
| 13:48.10 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 13:49.36 | troutowi | [TK]D-Fender, so compiling with the dont_optimize flag allows /etc/init.d/asterisk run and start without core dumps |
| 13:49.48 | troutowi | but then the issue from yesterday arises |
| 13:50.02 | troutowi | but without that flag, nothing will start |
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| 13:54.35 | jkroon | troutowi, sounds like a buggy compiler |
| 13:54.53 | troutowi | im using whats given in ubuntu 12.04 |
| 13:54.56 | troutowi | suggestions? |
| 13:55.24 | troutowi | im running in KVM too fyi |
| 13:55.33 | jkroon | gcc 4.7.2? we ran the ACM ICPC contest on Sat using that and didn't see anything obvious. |
| 13:55.55 | jkroon | asterisk inside a VM? is that wise? |
| 13:56.06 | troutowi | 4.6.3 |
| 13:56.51 | troutowi | what are your concerns with running in a VM? |
| 14:03.08 | cusco | hi... |
| 14:03.10 | cusco | looking at http://paste.debian.net/203523/ why is it that I can hear the other party and talk to them, althoug my softphone still shows RINGING ?, they don't seem to be sending the 200 OK |
| 14:04.04 | jkroon | troutowi, timing usually, but i'll admit that recently it's been much better than a few years back. |
| 14:04.10 | jkroon | scheduling unpredictability. |
| 14:05.14 | troutowi | ah ok, well we have been using asterisk for a few years on a VM without any issues. its just come to upgrading now, as we are on i think v1.6 atm, with an outdated distro |
| 14:05.27 | troutowi | but i just install gcc 4.7.2, and will report back |
| 14:05.31 | troutowi | installed* |
| 14:14.51 | troutowi | jkroon, it was in fact gcc :D |
| 14:15.08 | troutowi | 12.04's provided gcc 4.6.3 has issues apparently |
| 14:15.35 | troutowi | grabbed a ppa and installed 4.7.2, compiled and all works |
| 14:15.42 | jkroon | cool. |
| 14:15.53 | troutowi | thanks for your help |
| 14:17.21 | *** join/#asterisk krotos (~d3v1l@b2b-client.as48500.net) |
| 14:17.22 | krotos | hi all |
| 14:17.52 | *** join/#asterisk mjordan (mjordan@conference/digium/x-nrdnlhvhazxjfgml) |
| 14:17.52 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:18.00 | krotos | i have an ivr for incoming call that choose to send a call on three different queue |
| 14:18.18 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 14:18.39 | krotos | if i set'up an extension for pickup a call to an internal phone using Pickup cmd, not work. It' only work if the call is between two internal phone |
| 14:18.57 | krotos | there is a way to pickup a call , without specifying the context? |
| 14:20.53 | cusco | krotos: you ar asking how to do B when you want to do A, what is it that you wish to acomplish? |
| 14:22.24 | jkroon | troutowi, it's a pleasure. |
| 14:51.28 | *** join/#asterisk egonzalez_ergio (~egonzalez@186.137.1.3) |
| 14:51.36 | egonzalez_ergio | Hi to all. |
| 14:51.36 | egonzalez_ergio | I have a problem with asterisk+iamodem+hylafax. The system work great, I can send to a number and receive faxes without problem. But now I need to call a number, request signal to an human operator and then send the fax. How I can configure that? |
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| 14:51.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:54.04 | [TK]D-Fender | describe the human part in better detail as well as what is initiating the call-out |
| 15:04.35 | egonzalez_ergio | <PROTECTED> |
| 15:05.23 | [TK]D-Fender | egonzalez_ergio, Who is "you" calling a number? Be VERY specific about what is starting this process and where user invertntion exists on both ends. |
| 15:07.18 | egonzalez_ergio | [TK]D-Fender, ok: Me, human, dial a number with my ip phone. A human respond in the other side. |
| 15:07.42 | egonzalez_ergio | the other side respond using his standard fax machine |
| 15:07.49 | egonzalez_ergio | like a normal phone |
| 15:07.55 | egonzalez_ergio | when I request signal |
| 15:08.09 | egonzalez_ergio | this person push the standard button in his fax machine |
| 15:08.39 | egonzalez_ergio | is an old fashion fax machine |
| 15:08.42 | egonzalez_ergio | :-) |
| 15:08.52 | [TK]D-Fender | egonzalez_ergio, yso you verbally ask the other person to start their receive processand then you want to finish by ending the voice call and sending the fax to the active channel? |
| 15:09.05 | egonzalez_ergio | exactly |
| 15:09.34 | [TK]D-Fender | egonzalez_ergio, thats waht SendFax() is for. |
| 15:09.52 | [TK]D-Fender | egonzalez_ergio, So go place your call then TRANSFER to an extension that calls SendFax for you. |
| 15:10.25 | egonzalez_ergio | there are some example of configuration? |
| 15:11.07 | *** join/#asterisk Praise- (~Fat@unaffiliated/praise) |
| 15:11.08 | jkroon | egonzalez_ergio, no. |
| 15:11.24 | jkroon | why doesn't the person just hit the fax button when your fax machine hurts his/her ear? |
| 15:12.12 | egonzalez_ergio | jkroon, because here, in Argentina, the fax machine is used usualy in this way. |
| 15:14.17 | egonzalez_ergio | is in an internal used normaly like a phone. Or in rotative lines |
| 15:14.40 | egonzalez_ergio | in the 99% of cases the uses is like described |
| 15:14.46 | jkroon | eek. that makes things hard. [TK]D-Fender's suggestion is a good one, however, hylafax will try to dial the number using IAX over your ast box to transmit the fax, so I'm guessing what you should do then is ReceiveFAX() it to a file, then request a human to call the recipient of the fax and transfer that into a waiting SendFax(). |
| 15:14.56 | jkroon | fugly but it may just actually work. |
| 15:15.24 | [TK]D-Fender | jkroon, He's the sender |
| 15:15.39 | jkroon | perhaps after ReceiveFAX send an email or generate an automated call of sorts, so call origination here may help a lot. |
| 15:15.48 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:15.58 | [TK]D-Fender | jkroon, Ah, as a relay to Hylafax |
| 15:16.03 | jkroon | [TK]D-Fender, i know. but his "normal" scenario is email? => hylafax =IAX=> asterisk => PSTN |
| 15:16.06 | jkroon | [TK]D-Fender, yes. |
| 15:16.20 | [TK]D-Fender | jkroon, I suppose you could build a queueing system of sorts for that like "pick one to relay and even originate it out. |
| 15:16.25 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:17.32 | jkroon | i would ReceiveFAX() on ast, so basically "steal" the fax, then after ReceiveFAX finished, orignate a call to the operator, once operator answers, ask them in a pre-recorded message to hold and ask for fax, then send the channel to the outbound, once other person answered, hang up, which then drops that end of the channel into the appropriate SendFAX |
| 15:17.33 | egonzalez_ergio | nop, there are not a que, the fax must be sended inmediatly. I need to replate the nomal use in a machine-to-machine old normal fax |
| 15:17.36 | [TK]D-Fender | egonzalez_ergio, Since you're the sender, what are you looking to use to do the actual send? |
| 15:18.25 | jkroon | ok, re-reading that it's confusing ... |
| 15:18.28 | [TK]D-Fender | jkroon, egonzalez_ergio : Alternative idea : PARK the call to an alternate parking lot and then have whatever you'd use to dial the fax out dial INTO the parking lot |
| 15:18.30 | [TK]D-Fender | ^ |
| 15:18.51 | jkroon | hmm, [TK]D-Fender didn't even think of that one. Yea, that's a pretty good one. |
| 15:19.21 | egonzalez_ergio | [TK]D-Fender, like in a conference? |
| 15:19.28 | [TK]D-Fender | No. like PARKING |
| 15:19.30 | jkroon | but (even if egonzalez_ergio says otherwise) hylafax does internal queueing. |
| 15:19.40 | jkroon | egonzalez_ergio, core show application Park |
| 15:20.07 | [TK]D-Fender | jkroon, Depends as to what concurrency looks like, when he queus it up in there, etc... |
| 15:20.19 | [TK]D-Fender | jkroon, HF does complicate this of course. |
| 15:20.49 | jkroon | which is why I'd probably rather go with the robot voice asking the operator to ask the other end for a fax before hanging up |
| 15:21.22 | [TK]D-Fender | jkroon, Nifty idea, by HF would probably give up long before that point... |
| 15:21.45 | jkroon | [TK]D-Fender, that's why you get HF out of the way by store-forwarding the fax with ReceiveFAX/SendFAS |
| 15:21.49 | jkroon | *SendFAX |
| 15:22.47 | jkroon | what will be the hardest part is to ensure the fax doesn't go lost, ie, keep on re-originating until such time as fax delivery was successful, or maximum number of tries got exceeded. |
| 15:22.52 | [TK]D-Fender | jkroon, Yeah I suppose that'd do... then again I'd like to think that HF could also just store as a file somewhere and call a triggered process and not go through any of the actual fax routines as it's all still internal. |
| 15:23.02 | jkroon | and a successful call originate in this case != successful transmission. |
| 15:23.16 | [TK]D-Fender | jkroon, Basically as a client-server interface for their clients is all... but not for any of the "dirty work" |
| 15:23.30 | jkroon | [TK]D-Fender, sure, it needs to generate a faxable tiff anyway. |
| 15:24.10 | [TK]D-Fender | jkroon, which is where that would be taken as a unique file with dest # and off it goes to Originate, |
| 15:24.45 | jkroon | nice |
| 15:25.15 | jkroon | well, i have to make like a tree and leave. thanks for the mental exercise. chat again. have a good day. |
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| 15:45.37 | egonzalez_ergio | ok |
| 15:45.42 | egonzalez_ergio | thanks to all! |
| 15:45.59 | egonzalez_ergio | I'll try the park option, seems to be more simple |
| 15:47.02 | *** join/#asterisk BorjaGVO (d51beb92@gateway/web/freenode/ip.213.27.235.146) |
| 15:47.11 | egonzalez_ergio | bye! |
| 15:47.17 | *** part/#asterisk egonzalez_ergio (~egonzalez@186.137.1.3) |
| 15:52.53 | *** join/#asterisk wonderworld (~wonderwor@dsdf-4db55384.pool.mediaWays.net) |
| 15:53.44 | wonderworld | wow, windows 8 launch presentation is boring |
| 15:54.25 | aberrios | needs more cowbell |
| 15:55.07 | wonderworld | would need Balmer on speed. |
| 15:56.16 | cj | wonderworld: you didn't have to say it twice |
| 15:58.59 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 16:00.09 | wonderworld | yes, there he is :) |
| 16:00.10 | wotanskrieger | hy folks |
| 16:00.29 | wotanskrieger | please, anyone can help me? Just check it out: http://pastebin.com/rJmBDrkJ |
| 16:01.40 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 16:02.18 | chuckf | wotanskrieger: did you check the bios time? |
| 16:03.45 | wotanskrieger | chuckf: no, but I think there's not a delay on BIOS time. |
| 16:03.45 | *** join/#asterisk mjordan (mjordan@conference/digium/x-irekorwxbsuaqdkw) |
| 16:03.45 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:05.03 | chuckf | That's all I got then. |
| 16:05.58 | wotanskrieger | chuckf: do you know if dmidecode command also show BIOS time? |
| 16:08.50 | wonderworld | wotanskrieger: maybe this is the same problem? http://wl500g.info/showthread.php?14885-Asterisk-Incorrect-time-in-CDR |
| 16:09.33 | wotanskrieger | hi wonderworld, thanks. I saw this link last night :) |
| 16:09.35 | chuckf | wotanskrieger: I've never used it |
| 16:09.47 | wotanskrieger | wonderworld: I'll try to use it again |
| 16:09.58 | wotanskrieger | chuckf: http://lgallardo.com/2009/10/06/configurar-el-reloj-del-bios-en-debian/ |
| 16:10.18 | wotanskrieger | chuckf: I think it solves and yes, my BIOS time was delayed |
| 16:10.29 | sekil | anyone experienced issues with 1.8.x and sip session timers...I'm getting calls cut after 90+ secs |
| 16:12.12 | chuckf | wotanskrieger: glad that seems to have solved it for you |
| 16:12.15 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:13.08 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 16:14.49 | wotanskrieger | chuckf: I just set new time but I need to test it soon (5 PM). By the way, I hope you can help me again: Did Asterisk use only NTP service to synchronize you time? There is any internal component (.conf, agi rules script or someone else) which manager fusetime ?! |
| 16:16.14 | *** join/#asterisk danfromuk (~IceChat77@2.27.12.9) |
| 16:19.46 | danfromuk | Hi, how do I go about diagnosing this: res_rtp_asterisk.c:2356 ast_rtp_read: Unknown RTP codec 126 received from |
| 16:19.48 | danfromuk | ? |
| 16:20.09 | danfromuk | The user is complaining about poor quality even though another PC on the same internet connection is working fine. |
| 16:20.52 | sekil | danfromuk: those are custom codecs numbers ... > 96 I think |
| 16:21.25 | danfromuk | custom codec? |
| 16:23.00 | danfromuk | They are using 3cxphone. They can't get xlite to connect. |
| 16:24.23 | sekil | danfromuk: RTP Dynamic Payload Numbers |
| 16:24.38 | sekil | they should have a line with name in the sdp |
| 16:25.38 | *** join/#asterisk Galen (~Galen@rrcs-24-43-20-118.west.biz.rr.com) |
| 16:28.02 | danfromuk | I've not captured the sip packets yet. I just noticed the RTP notice in the CLI and realised that it may be the source of the user's sound issues. |
| 16:29.56 | *** join/#asterisk gr0mit (~tim@2a01:9c41:ffff:30:4161:ef1f:c51c:47b5) |
| 16:34.26 | gr0mit | ok, anyone got a worksround to this? I've got a customer who answeres a call, parks it, picks it up at a different extension, and then tries to transfer the cal using her snoml. the transfer fails. |
| 16:35.28 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 16:37.05 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:38.00 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 16:47.40 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 16:48.25 | chuckf | wotanskrieger: Asterisk is a hobby for me. I don't know enough to answer your question about what it uses for a timing source |
| 16:49.10 | chuckf | wotanskrieger: though judging by the problem you have, it uses more than NTP for its time source somewhere |
| 16:53.06 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 17:01.43 | *** join/#asterisk pabelanger (pabelanger@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:01.43 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:06.40 | *** join/#asterisk k610 (~Instantbi@109.130.39.164) |
| 17:10.41 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 17:10.41 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 17:15.32 | krotos | cusco: i need to have an extension to dial by a phone that allow me to pickup a call to another phone. I.e. An incoming call from pstn go to queue1. Queue 1 has phone P1,P2,P3 in ringall strategy |
| 17:16.08 | krotos | but P1,P2,P3 ring without answer. Then a P6 Phone pickup the call directed to P1(or p2,or p3) |
| 17:16.43 | krotos | is this possible? or i'have to pickup the call directed to the queue? |
| 17:33.27 | *** join/#asterisk TriJetScud (~TriJetScu@2001:470:e97f:1000::1) |
| 17:33.57 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 17:40.06 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:40.34 | [TK]D-Fender | krotos, Set a PICKUPGROUP |
| 17:40.37 | *** join/#asterisk bluOxigen (~noreply@unaffiliated/bluOxigen) |
| 17:42.44 | SerajewelKS | cj: not bad, you? you're the cj from #mono, yes? |
| 17:57.30 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
| 17:59.05 | krotos | [TK]D-Fender: i've already setted the pickupgroup. After that i've to pickup the enteire group? |
| 18:03.18 | [TK]D-Fender | ... |
| 18:03.21 | [TK]D-Fender | You issue a pickup... |
| 18:03.34 | krotos | ? |
| 18:04.04 | *** join/#asterisk ankso (~nimda@46.198.136.133) |
| 18:05.32 | krotos | ok, solved.. |
| 18:05.51 | krotos | thankyou for your support, i've found the error..it was my error |
| 18:05.55 | *** part/#asterisk tladuke (~travis@pool-96-247-87-153.lsanca.fios.verizon.net) |
| 18:05.59 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 18:10.31 | *** join/#asterisk adeel (~adeel@216.183.80.220) |
| 18:11.21 | ankso | Dear all, i have a problem, i just made a yum update and i get continously this error http://inetpro.org/pastebin/ea22f20a12cb49e7b78c52ac6f871455 when i run the command amportal start |
| 18:11.29 | ankso | any idea? |
| 18:13.32 | ankso | also i take this error when i try amportal -rvv... |
| 18:15.33 | navaismo | try in the elastix channel |
| 18:15.41 | ankso | ok |
| 18:17.56 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v017-125.mobile.uci.edu) |
| 18:40.37 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 18:56.45 | *** join/#asterisk natschil (~nathanael@stgt-5f70869e.pool.mediaWays.net) |
| 19:01.06 | natschil | Hello. I'd like to replace skype with something open source. I've heard asterisk can be used to setup a voip server, however, I know very little about voip in general. Therefore, I would like to ask the following before starting with learning about asterisk etc... Is it possible to run a standalone asterisk server to which users can connect to on some arbitary port and call other users that are also connected to the server? |
| 19:01.47 | WIMPy | yes |
| 19:02.11 | WIMPy | If you want to learn about Asterisk, try the |
| 19:02.13 | WIMPy | ~book |
| 19:02.14 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:11.16 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 19:11.17 | natschil | WIMPy: thanks. |
| 19:15.00 | *** join/#asterisk sherl0k (~cbalcum@199.167.121.226) |
| 19:15.45 | *** join/#asterisk silverado|2 (~kvirc@108.178.221.178) |
| 19:16.31 | sherl0k | Hi all, anyone have experience with the getting DTMF tones to register correctly? I'm having trouble with my conference line and outside callers pressing keys on their phones. |
| 19:17.16 | sherl0k | running FreePBX 2.10.1.2 / Asterisk 1.8.16.0 |
| 19:17.43 | *** part/#asterisk ankso (~nimda@46.198.136.133) |
| 19:20.03 | silverado|2 | Hi everyone. I really hope someone can help me here. I am running Fedora 18 and asterisk 1.8.16.0. When I install asterisk it does not create the system asterisk user or the asterisk.ctl file. I have tried several methods but to no avail. |
| 19:22.25 | chuckf | silverado|2: fedora 18 is still beta I think, you might want to file a bug report with them |
| 19:22.48 | troutowi | [TK]D-Fender, have u ever had issues with sip trunks registering? |
| 19:23.40 | [TK]D-Fender | troutowi, Maybe we could move a few steps forward and you could chow us the probaly you are actually having.... |
| 19:23.55 | troutowi | lol |
| 19:23.57 | troutowi | fair enough |
| 19:24.29 | [TK]D-Fender | show* |
| 19:26.47 | *** join/#asterisk keith__ (~keith@udp278022uds.hawaiiantel.net) |
| 19:27.22 | troutowi | http://pastebin.com/4KXkrAFs |
| 19:28.03 | *** join/#asterisk Tim_Toady (~fuzzy@178.128.200.147.dsl.dyn.forthnet.gr) |
| 19:31.00 | [TK]D-Fender | troutowi, This means enabling SIP DEBUG and actually LOOKING at the attempts... |
| 19:31.04 | [TK]D-Fender | "sip set debug on" |
| 19:31.23 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 19:31.29 | troutowi | ok |
| 19:32.36 | silverado|2 | I always get Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) even when root |
| 19:33.27 | [TK]D-Fender | silverado|2, And is Asterisk even running? |
| 19:33.34 | [TK]D-Fender | Is the PID file where it's expected to be? |
| 19:36.16 | silverado|2 | [TK]D-Fender, Yes, I have made sure several times |
| 19:37.03 | [TK]D-Fender | silverado|2, Maybe you could show us... |
| 19:37.08 | [TK]D-Fender | ~pb |
| 19:37.08 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:37.11 | [TK]D-Fender | ^^^ |
| 19:40.55 | troutowi | http://pastebin.com/GiHcssfM |
| 19:42.29 | chuckf | silverado|2: PMing me will not help solve your issue. That you ahd it working at one time and then reformmatted, then trying two distros tells me you're doing something wrong |
| 19:43.19 | *** join/#asterisk danfromuk (~IceChat77@2.27.12.9) |
| 19:44.31 | danfromuk | Hi, I'm struggling to locate the source in the dialplan thats dialling SIP/h which doesnt exist. Any ideas how I can debug it? |
| 19:45.04 | danfromuk | It occurs after hangup but it dials SIP/h and SIP/OTHER_ASTERISK_SERVER/h |
| 19:45.18 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:49.51 | [TK]D-Fender | troutowi, If you're gong to mangle the evidence we aren't going to get very far |
| 19:50.07 | troutowi | hm? |
| 19:50.18 | troutowi | what else do you want? |
| 19:50.58 | [TK]D-Fender | danfromuk, "source in dialplan"? huh? |
| 19:52.52 | danfromuk | I'm trying to figure out why SIP/h and SIP/OTHER_ASTERISK_SERVER/h is being dialled when a call is hung up or not answered |
| 19:53.05 | danfromuk | There are no options in the dial cmd |
| 19:53.07 | [TK]D-Fender | danfromuk, I'd start by looking at the call....... |
| 19:54.31 | troutowi | [TK]D-Fender, what is it you would like, my id and ip |
| 19:54.32 | troutowi | ? |
| 19:54.47 | [TK]D-Fender | I'd like to see that actual full SIP debug. |
| 19:54.57 | troutowi | that was the full debug... |
| 19:55.08 | [TK]D-Fender | UNFILTERED |
| 19:55.59 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 20:00.20 | drmessano | IP and usernames are important to spot inconsistencies. What you think is obvious and feel safe masking may be the very proof that you have a something misconfigured |
| 20:00.54 | drmessano | Yes, there are many cases of incorrectly placed IPs, IP addy's being fat fingered, usernames being fat fingered, etc |
| 20:01.42 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 20:01.55 | troutowi | drmessano, the ips and username are correct as i am literally copy and pasting from our old * server |
| 20:03.30 | troutowi | the same issue is happening on the old * server too... had to shut the old one down for a few hours, and upon reboot, trunk couldnt register. |
| 20:03.37 | troutowi | but im working with [TK]D-Fender on it |
| 20:03.40 | drmessano | We don't care or trust you. If you're asking for outside help, then obviously what you have isn't working, and you don't know what isn't correct. |
| 20:03.54 | troutowi | i understand |
| 20:04.33 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 20:04.35 | drmessano | It's common sense.. If you can't spot an error, and are asking for someone else to spot it, how can you logically trust anything in your configuration? |
| 20:05.13 | troutowi | for the configuration of other places yes ill agree, but in this case, the trunk info is correct |
| 20:05.25 | drmessano | So you say |
| 20:06.59 | drmessano | [TK]D-Fender, as much as he is very excitable, is more patient that I am about masked info. He'll scream and demand it until some gets the idea and stops masking it. I just simply refuse to work with someone who wants to play games |
| 20:07.52 | troutowi | cool drmessano. but as i said, he is taking a look |
| 20:08.40 | *** join/#asterisk cmendes0101 (chris@conference/digium/x-rzdzonlxznycozwg) |
| 20:08.59 | drmessano | Problem solved then, I guess. You got someone to look and didn't have to learn a lesson |
| 20:09.18 | [TK]D-Fender | drmessano, He provided it quickly in private... |
| 20:09.23 | [TK]D-Fender | Didn't have to ask twice. |
| 20:09.31 | drmessano | Ok |
| 20:09.52 | drmessano | Not sure why he continued to argue the point then. Nevermind |
| 20:10.02 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 20:10.20 | troutowi | drmessano, ur the one who kept at it, i told u right away he was looking at it |
| 20:10.25 | troutowi | but no biggie |
| 20:11.42 | drmessano | You didnt exactly elaborate and state you had provided the info |
| 20:11.44 | drmessano | But whatever |
| 20:11.58 | [TK]D-Fender | Yeah, we're all good here..... |
| 20:14.14 | drmessano | Anyone using Asterisk 11 and XMPP/Motif? |
| 20:14.47 | drmessano | Having an authentication issue |
| 20:15.58 | drmessano | "Waiting to Request TLS" |
| 20:19.30 | cmendes0101 | Anyone have the url for asterisk 11? I didn't catch it on the slides at astricon |
| 20:19.41 | kaldemar | danfromuk: it's exten => .,1,Dial(SIP/${EXTEN}) most likely. that's why you should not use "." as a pattern alone. |
| 20:20.22 | kaldemar | cmendes0101: url? for what? source package? |
| 20:21.10 | cmendes0101 | I believe it was for the tar version of the source |
| 20:21.26 | kaldemar | cmendes0101: you'll find them all here, always: http://downloads.asterisk.org/pub/telephony/asterisk/ |
| 20:21.37 | cmendes0101 | I could be wrong. They said they just put it online before the end chats |
| 20:21.51 | kaldemar | it's there. |
| 20:22.13 | *** part/#asterisk navaismo (~Airo@189.191.94.52) |
| 20:23.03 | cmendes0101 | kaldemar, Thanks |
| 20:28.47 | [TK]D-Fender | Checkout time, BBL |
| 20:32.57 | sherl0k | Hi all, anyone have experience with the getting DTMF tones to register correctly? I'm having trouble with my conference line and outside callers pressing keys on their phones. |
| 20:32.58 | sherl0k | running FreePBX 2.10.1.2 / Asterisk 1.8.16.0 |
| 20:37.14 | sherl0k | i've tried changing the dtmfmode, setting relaxedtmf, changing the order of codecs |
| 20:37.19 | sherl0k | getting nowhere :/ |
| 20:38.04 | WIMPy | How do they reach you? |
| 20:41.48 | *** join/#asterisk elshaa (~elshaa@o.es6.aedgency.net) |
| 20:41.50 | elshaa | hello |
| 20:43.39 | sherl0k | They can dial into our office just fine, reaches the extension that points to an IVR |
| 20:43.59 | sherl0k | but when they attempt to press a key that the IVR will redirect their call to, it isn't recognized |
| 20:44.09 | sherl0k | I've turned on DTMF debugging in the consoler |
| 20:44.28 | sherl0k | Some calls I'll see that the DTMF tone isn't long enough and the phone system will extend it automatically |
| 20:44.50 | sherl0k | Other cases though it just register at all |
| 20:48.58 | elshaa | I am running asterisk on a physical machine with a Sangoma card (E1). Would it be possible to virtualize it ? I have a doubt about how the virtual machine would use the physical Sangoma card... |
| 20:50.24 | *** join/#asterisk egonzalez_ergio (~egonzalez@190.189.44.43) |
| 20:50.37 | WIMPy | elshaa: Ask Sangoma if teir dirvers can be used that way? |
| 20:50.49 | *** part/#asterisk egonzalez_ergio (~egonzalez@190.189.44.43) |
| 20:51.46 | *** join/#asterisk BorjaGVO (532ad2d0@gateway/web/freenode/ip.83.42.210.208) |
| 20:51.52 | jpsharp | Depends on if your virtual host can do direct PCI I/O. |
| 20:54.10 | BorjaGVO | Hi everyone. How can I play part of a sound file for just 20 seconds (without editing the file)? The only sound-related cmd that I found that has a way to start/stop it is Playtone/StopPlaytone but as far as I read it doesn't play regular sound files. Any ideas of how to do this? |
| 20:54.35 | WIMPy | not |
| 20:54.47 | WIMPy | Edit the file. |
| 20:55.23 | WIMPy | Ok, it's not entirely true. You could do it externally via AGI or AMI. |
| 20:56.33 | BorjaGVO | Ok, but not with Asterisk dialplan... |
| 20:56.37 | BorjaGVO | ok |
| 20:56.42 | BorjaGVO | thanks |
| 20:57.07 | BorjaGVO | I think this could be something to be done...it's kind of useful... |
| 20:57.54 | WIMPy | I can't thnik of a use case. What do you want it for? |
| 20:59.39 | BorjaGVO | Yeah, may be not that obvious...What I want to do is set a minimum time of 20 seconds for the queue. I mean, if a caller geets into the queue, the music must get played 20 seconds at its minimum, even if there are no agents available |
| 21:00.10 | BorjaGVO | There is not such option, right? |
| 21:00.55 | BorjaGVO | The way I was thinking of doing it is to play the song for 20 seconds and THEN putting the caller into the queue... |
| 21:00.59 | WIMPy | is not in to queues. |
| 21:01.35 | WIMPy | Does the queue resume MOH where it was left? |
| 21:02.40 | BorjaGVO | it doesn't matter |
| 21:03.33 | WIMPy | If it does you could just StartMusicOnHold() and then Wait(20). |
| 21:06.03 | BorjaGVO | but it wont stop until file ends, right? |
| 21:06.37 | WIMPy | There's StopMusicOnHold() as well. |
| 21:06.59 | WIMPy | Maybe you don't even need that. I've never tried to use it. |
| 21:07.18 | WIMPy | But it seems worth a try for what you want. |
| 21:10.32 | BorjaGVO | yes! of course it does..I didn't have notice about it...I'll give it a try. I think is just what I want! |
| 21:13.40 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 21:13.41 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 21:16.54 | *** part/#asterisk sekil (~Ognjen@78.24.104.82) |
| 21:20.19 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 21:22.53 | *** join/#asterisk goddva (glarsen@conference/digium/x-qlpknojokicglcoi) |
| 21:23.13 | goddva | how does the autoconf/*.m4 files work? are they generated or are they "static"? |
| 21:29.13 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 21:39.58 | *** join/#asterisk sertaconay (~sertacona@unaffiliated/sertaconay) |
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| 22:06.38 | elshaa | WIMPy: Ok, thanks, and sorry fordelay |
| 22:07.07 | elshaa | jpsharp: The direct PCI I/O doesn't depend on the hypervisor only ? KVM in my case? |
| 22:20.35 | *** join/#asterisk natschil (~nathanael@stgt-5f70869e.pool.mediaWays.net) |
| 22:20.46 | natschil | Hello. How do I change the SIP port that asterisk listens on? |
| 22:20.54 | natschil | I can't seem to find how to do that on the internet |
| 22:23.03 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 22:39.55 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
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| 22:56.21 | jeffspeff | ok, trying to setup a sip trunk between 2 boxes. I'm setting the static host address on the sip accounts so they don't have to register. also setting them to qualify. But i can't get them to qualify. not seeing anything on either side. how do i set this up right? |
| 23:07.15 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:12.12 | jeffspeff | [TK]D-Fender, so glad you joined |
| 23:12.19 | jeffspeff | ok, trying to setup a sip trunk between 2 boxes. I'm setting the static host address on the sip accounts so they don't have to register. also setting them to qualify. But i can't get them to qualify. not seeing anything on either side. how do i set this up right? |
| 23:14.14 | [TK]D-Fender | If you can't see anything then your networking is screwed up or are pointing a box in the wrong direction entirely |
| 23:15.57 | jeffspeff | my sip provider has there setup this way, and i made these settings very similar to how i set them up |
| 23:16.32 | jeffspeff | just a sip account, set name, port, host, allow, disallow, context, qualify |
| 23:16.37 | [TK]D-Fender | What does your "provider" have to do with YOUR two boxes talking between each other? |
| 23:16.54 | jeffspeff | nothing, but i'm trying to mimic same functionality |
| 23:17.37 | jeffspeff | we don't register with sip to their system, i'm trying to figure out how this is done |
| 23:18.02 | [TK]D-Fender | make a peer on each side. place calls. done |
| 23:18.12 | [TK]D-Fender | show us actual ATTEMPTS |
| 23:19.35 | jeffspeff | ok, made peers on both sides, when the call comes through from A to B it doesn't go to the configured context within B for the account, it goes to the default trash context specified in [general] |
| 23:19.57 | jeffspeff | when i do a sip show peers on either box, the peers show as unreachable |
| 23:20.38 | [TK]D-Fender | waits for story-time to finish |
| 23:20.50 | jeffspeff | lol |
| 23:21.05 | jeffspeff | the call goes through, but not properly |
| 23:21.54 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
| 23:23.03 | jeffspeff | [Oct 25 13:27:00] NOTICE[10837]: chan_sip.c:23577 handle_request_invite: Call from 'TEST' (12.49.160.154:5060) to extension '8702155555' rejected because extension not found in context 'UNKNOWN-INBOUND'. |
| 23:23.44 | [TK]D-Fender | jeffspeff: Pastebin. EVERYTHING. And I'f not seeing those calls with SIP DEBUG enabled it isn't "everything" |
| 23:27.14 | *** join/#asterisk elico (~Thunderbi@bzq-109-64-221-76.red.bezeqint.net) |
| 23:28.45 | jeffspeff | [TK]D-Fender, http://pastebin.com/58qLunHw |
| 23:28.54 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 23:31.12 | [TK]D-Fender | Found peer 'TEST' for '2705649026' from 12.49.160.154:5060 |
| 23:31.21 | [TK]D-Fender | Looking for 8702155555 in UNKNOWN-INBOUND (domain 192.168.0.10) |
| 23:31.26 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 23:31.32 | [TK]D-Fender | Looks cut & dry |
| 23:31.53 | [TK]D-Fender | identifies as a peer and then there is nothing to match what they dialed. |
| 23:32.24 | *** join/#asterisk philfry (~aj@h222.191.101.208.static.ip.windstream.net) |
| 23:32.39 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 23:32.53 | philfry | i'm running asterisk now and anytime i try to dial out using a SIP or IAX2 trunk i get unavail |
| 23:33.11 | jeffspeff | The peer 'TEST' has a context of 'TEST-INBOUND' which is where it's call should go to. 'UNKNOWN-INBOUND' is set in [general]... plus the peers aren't qualify'ing |
| 23:33.35 | [TK]D-Fender | jeffspeff: Doesn't look like. And *I* don't see it..... |
| 23:34.47 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe93:fa45) |
| 23:35.09 | jeffspeff | [TK]D-Fender, how do i get them to qualify properly without doing a sip register? |
| 23:35.14 | jeffspeff | or can you even do that? |
| 23:36.05 | jeffspeff | philfry, you might try #asterisknow channel |
| 23:37.03 | [TK]D-Fender | jeffspeff: Why are we jumping track? |
| 23:37.19 | [TK]D-Fender | Registration has NOTHING to do with "qualify" working |
| 23:37.30 | jeffspeff | oh |
| 23:38.24 | jeffspeff | [TK]D-Fender, much to my embarassment i had forgotten to do a sip reload and apply the changes that added the context to the TEST peer. so that works now |
| 23:40.05 | WIMPy | Is there a way to supply the soundfile containing the user name wehn joining a confbridge? |
| 23:41.10 | philfry | this really isn't asterisk now specific |
| 23:42.36 | WIMPy | philfry: Your question wasn;t specific in any way. |
| 23:42.47 | [TK]D-Fender | ~pb |
| 23:42.47 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 23:42.53 | [TK]D-Fender | philfry: ^^^^^^^ |
| 23:42.58 | philfry | sorry |
| 23:43.00 | [TK]D-Fender | philfry: SHOW us the failed call. |
| 23:43.05 | philfry | sure thing |
| 23:43.26 | [TK]D-Fender | philfry: with SIP DEBUG enabled if it's going out that way, IAX2 DEBUG enabled if the other |
| 23:44.54 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 23:57.20 | philfry | i took out the username in the iax registration |
| 23:57.22 | philfry | http://pastebin.com/6q3uvuET |