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| 00:26.29 | Katty | looks in |
| 00:26.33 | Katty | hands newtonr a pringle. |
| 00:27.12 | newtonr | nom nom nom |
| 00:27.22 | newtonr | The problem is, now I can't stop |
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| 00:28.06 | Katty | yes. i think there is a requirements that you eat about 3 pringles at a time. |
| 00:28.19 | Katty | and that's just one bite. |
| 00:32.03 | newtonr | That is the law of chips |
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| 00:35.02 | newtonr | Katty, have a good night! |
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| 02:53.09 | linocisco | can asterisk support large scale enterprise deployment more than 11,000 concurent calls with reference to http://goo.gl/h1RXKI |
| 02:53.11 | linocisco | ? |
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| 03:04.24 | volga629 | everything depend on requirements |
| 03:04.46 | volga629 | or what role is will play in voip network |
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| 03:36.14 | phix | linocisco: sure, why not |
| 03:38.19 | linocisco | phix, why was it said in comments? |
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| 03:41.10 | phix | linocisco: well it is detailed on the digium website |
| 03:41.29 | phix | that is has been and is being used in large enterprise origanisations |
| 03:41.31 | linocisco | phix, link or URL please! |
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| 03:46.17 | phix | http://www.asterisk.org/ |
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| 06:50.14 | alex999 | Hello |
| 06:51.08 | linocisco | hi |
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| 07:46.14 | Eric-K | Anyone ever experienced that Asterisk won't send any more T.38 data after the receiving fax sends the CFR? I'm talking Asterisk 11 right now. |
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| 09:24.30 | alex999 | hello ... who has Asterisk 13.2 installed |
| 09:24.51 | alex999 | i need to do a test |
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| 09:43.12 | ChannelZ | I does |
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| 09:44.26 | alex999 | hello channelz |
| 09:44.50 | ChannelZ | ahoy |
| 09:44.55 | alex999 | I wonder whether you have any problems with MWI |
| 09:46.35 | madax | hi guys i'm using asterisk 11, making calls with ami originate with async=true by doing some curl. I don't receive in response an OriginateResponse, anyone know if there is something in particular to do ? |
| 09:46.39 | ChannelZ | not that I know of |
| 09:47.39 | alex999 | for my opinion asterisk does not notify any "message-summary" ... I did a debug and do not see any trace of those ip packs |
| 09:48.02 | ChannelZ | chan_sip or pjsip? |
| 09:48.05 | alex999 | by your side it works correctly |
| 09:48.32 | alex999 | pjsip |
| 09:49.06 | ChannelZ | do you have a mailboxes=xxx@xxx for the endpoint entries in question? |
| 09:49.28 | alex999 | let me have a look |
| 09:51.51 | alex999 | all extensions are configured as mailboxes=xxx@device Note Im using freepbx |
| 09:52.53 | ChannelZ | you'd have to ask #freepbx then |
| 09:53.55 | ChannelZ | Given a mailbox 200 in the context [default] in voicemail.conf, mailboxes=200@default for the 'endpoint' entry in pjsip.conf works |
| 09:54.20 | alex999 | i am going to try.... |
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| 09:57.26 | ChannelZ | good luck - I'm off to bed |
| 09:58.17 | alex999 | channelz.... you are magic .... now it works |
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| 10:45.45 | mjt | hello. We've 2 offices, with asterisk running on both ends. Right now asterisks are interconnected ising IAX, and there are a few VoIP phones on each end. I noticed that when I call a number in the same office, the call goes directly between the two phones, -- is it possible to do the same when I call another office? |
| 10:46.33 | WIMPy | Yes, but you need to stay on one protocoll all the way. |
| 10:46.50 | mjt | hm. So I should replace IAX with SIP? |
| 10:47.21 | WIMPy | But usually you would disable the possibility to avoid issues with that. |
| 10:47.55 | WIMPy | You could, but if there's any NAT involved, you probably don't want to. |
| 10:48.22 | mjt | I ask because right now, when I make a call from 3rd location, I connect to "my" office which forwards the call to another office, so the traffic doubles compared with when I call the other office directly |
| 10:48.34 | mjt | and at both sides, we pay for traffic |
| 10:49.19 | WIMPy | And what's that 3rd location? |
| 10:49.39 | mjt | we've internal network (vpn) where all internal addressing is directly reachable |
| 10:49.52 | mjt | 3rd location is, say, my home or an internet cafe |
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| 10:51.21 | WIMPy | You probably don't want to try to redirect media from 3rd parties. |
| 10:51.40 | mjt | it all stays inside the vpn anyway |
| 10:52.43 | WIMPy | Ok, that should work then. But you still could have the issue that the endpoint will try to communicate with the public IP they diescovered for themselves. |
| 10:52.53 | WIMPy | So it can still become interesting. |
| 10:53.04 | mjt | yeah, I see what you mean |
| 10:53.26 | mjt | like, basically, is the network routing set up correctly? |
| 10:54.12 | mjt | some voip phones don't even let to choose which sip account to use for a given outgoing call... |
| 10:54.33 | WIMPy | Hmm. Sounds bad. |
| 10:55.44 | mjt | (i can avoid placing the call to a "wrong" office by setting up 2 accounts for each phone, one at one office and another at second office, but this becomes ugly to say at beast) |
| 10:56.50 | WIMPy | So that would only be for people on the road? |
| 10:57.03 | mjt | maybe I'm doing it all wrong and we should have just one asterisk, |
| 10:57.31 | mjt | but I'm not sure I want the internal phone system to stop working if the external network goes down |
| 10:57.40 | WIMPy | That would help. |
| 10:57.50 | WIMPy | I can see that. |
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| 11:00.34 | mjt | what's a downside of replacing IAX with SIP ? |
| 11:00.57 | mjt | (btw, I'm comple newbie to asterisk and to voip in general ;) |
| 11:02.04 | WIMPy | IAX uses less bandwidth on multiple concurrent calls. Also I find it a lot more trustworthy. |
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| 12:01.46 | edgy | Hi, I am new to asterisk and it's my first time here so welcome me in this seems to be nice world! ;) |
| 12:02.33 | edgy | I just want to make sure I understand the basic hardware stuff before delving deeper |
| 12:04.16 | edgy | If I install asterisk on a server and let's say I want 60 extensions. my understanding is I need to buy 60 ip phones, plug them all in a switch, plug the switch in asterisk, configure it somehow and viola, wrong? |
| 12:04.50 | WIMPy | Yes. |
| 12:05.33 | WIMPy | But "extension" means somehting else in the Asterisk world, so better call them phones. |
| 12:05.52 | edgy | WIMPy: thanks, so there is no need to buy cards and plug them on the computer? |
| 12:06.24 | WIMPy | You only need tose to interface to normal phones or to phone lines. |
| 12:07.46 | edgy | WIMPy: aha! so if I have 6 phone lines (rj11), then I need a card for that, right? |
| 12:09.14 | WIMPy | If you want to use them, yes. |
| 12:09.31 | WIMPy | You could change that to IP as well, off course. |
| 12:10.04 | WIMPy | Interfacing with analog lines isn't nice. |
| 12:10.50 | edgy | WIMPy: what do you mean by change that to IP? |
| 12:11.33 | WIMPy | Use an ITSL instead of lines. |
| 12:11.40 | WIMPy | ~itsp |
| 12:11.41 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 12:11.48 | WIMPy | Use an ITSP instead of lines. |
| 12:15.04 | edgy | WIMPy: if I go the ITSP route, I would take new numbers from them and not be able to use my old numbers, right? |
| 12:15.52 | WIMPy | You should be able to port your numbers to another company in most parts of the world. |
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| 12:16.40 | edgy | WIMPy: can you point me to a documentation on how to configure my lines with itsp? |
| 12:16.51 | edgy | I am not familiar with that |
| 12:17.28 | WIMPy | More or less the same as an IP phone. |
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| 12:22.50 | edgy | WIMPy: so I don't need the POTS lines, I would just conact an itsp, pay him money, tell him my numbers, connect my switch to the internet via a router and that's all? |
| 12:23.40 | WIMPy | Pretty much, yes. |
| 12:24.19 | WIMPy | Depends on how porting numbers to a new carrier is handled in your area. |
| 12:24.52 | WIMPy | And you need a reliable internet connection with enough bandwidth, off course. |
| 12:25.23 | edgy | the routing of numbers is done by the itsp or my carrier? |
| 12:26.16 | WIMPy | That process will involve both you old and your new carrier and possibly some authority. |
| 12:26.36 | edgy | WIMPy: and in case the internet connection is not good, then I need to buy something like http://www.digium.com/en/products/telephony-cards/analog/8-port |
| 12:26.55 | WIMPy | Yes. |
| 12:27.19 | WIMPy | Or if you can change to ISDN, that would be a lot less pain than POTS. |
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| 12:49.43 | mjt | an alternative to "cards" is to use voip (fxs or fxo) switches |
| 12:50.38 | mjt | 2-port (fxs+fxs, fxo+fxo or fxs+fxo) switches are quite common and cheap |
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| 13:12.09 | edgy | mjt: can you point me to a good voip switch that supports 6 lines? |
| 13:12.59 | edgy | and what if I want to go wireless? |
| 13:13.31 | WIMPy | Wireless what? |
| 13:15.50 | edgy | wireless handsets. I don't want to dig holes in my office walls. can I just make everything wireless inside the office |
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| 13:17.09 | WIMPy | How many handsets? |
| 13:19.02 | WIMPy | How many simultaneous calls? |
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| 13:35.22 | mjt | another commonly used solution for that is to use dect phones |
| 13:35.58 | mjt | gigaset a510 ip supports up to 6 handsets, 6 sip accounts and 2 simultaneous calls |
| 13:36.24 | mjt | and it is cheaper than most 2-port voip gateways :) |
| 13:39.38 | WIMPy | Definitely the way to go. If it's big enough. |
| 13:41.49 | edgy | 60 handsets |
| 13:42.03 | edgy | around 10 sim calls |
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| 13:43.36 | WIMPy | You need one of the big solutions then, like Snom. |
| 13:46.55 | edgy | by 10 sim calls I mean for all the phones not in one phone |
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| 13:47.29 | edgy | each phone only one call at a time |
| 13:47.30 | [TK]D-Fender | I've heard many claims of the Snom M9 series as being flakey |
| 13:47.42 | WIMPy | Yes, that would only be one call anyay, no matter how many you have on hold. |
| 13:48.25 | WIMPy | I think the M9 is EOL anyway. But it would not fit the purpose anyway. |
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| 13:49.18 | WIMPy | You'd need multiple M700. |
| 13:51.01 | WIMPy | While one base can handle 10 calls, it will only allow 30 handsets. |
| 13:53.04 | WIMPy | "only"... |
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| 13:56.11 | edgy | ok thanks a lot I will read about those devices, install asterisk, get the feel of it and then come here to annoy you with questions, thanks again ;) |
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| 14:02.25 | mutilator | anyone here into configuring snom handsets? have a few questions... can you make it reset the network settings each reboot so auto-provisioning downloads them each boot? any way to reset the auto provision URL so even after a hardware reset it's your URL? and.. maybe a few more questions... |
| 14:03.35 | WIMPy | You obviousely need to have networking working for provisioning. |
| 14:04.34 | mutilator | i want it to reset to defaults.. e.g. no VLAN, DHCP w/ all options |
| 14:04.42 | WIMPy | And you can get custom firmwares with your URLs on request. or you can register the phone and your URL with them. |
| 14:06.23 | WIMPy | You can provision whatever you want. But I'm not sure I understand the idea. |
| 14:06.33 | mutilator | because i have a few issues, if i'm at the office, i plug in a phone it will provision to the tagged voice vlan at the office |
| 14:06.49 | mutilator | i now bring that phone off site and it holds onto that vlan setting and will never dhcp becuase it's trying to tag |
| 14:07.22 | mutilator | the other issue is when a phone is defaulted it'll get dhcp from dns, but when it vpns in i need dns from the vpn |
| 14:07.41 | mutilator | the openvpn implementation on the phone isnt properly pushing dns settings configured via openvpn |
| 14:07.49 | mutilator | so i have to set the dns via the auto config statically |
| 14:08.02 | mutilator | which once you download the config for the first time it can never reach those servers becuase they're behind the vpn |
| 14:08.24 | mutilator | and it doesnt get on the vpn before it tries to download the auto-provision |
| 14:08.32 | WIMPy | TWhy don't you just always use the vpn? |
| 14:08.42 | mutilator | ? |
| 14:08.54 | mutilator | it tries to grab the provision before it connects to the vpn |
| 14:08.58 | mutilator | not after |
| 14:09.18 | WIMPy | I guess that makes sense. |
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| 14:09.39 | mutilator | it's ideal for sure |
| 14:10.03 | mutilator | the problem is that either the network settings need to default before it grabs that provision url or openvpn dns push options need fixed |
| 14:10.03 | mutilator | heh |
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| 14:10.21 | mutilator | defaulting settings would be best because it'd also remove the VLAN setting from any previous provisions |
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| 14:41.49 | uluatu | Is there anybody here facing problems with interruption on RTP during calls for logged Agents in a queue on version 1.8.32? During the call the attendant can't hear the other party for 1 or 2 seconds. Im facing the same problems with different servers in different places(networks). |
| 14:42.16 | uluatu | It is not a netwrok related problem. |
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| 18:06.48 | Synthase_ | uluatu: tcpdump and analyze RTP in Wireshark. I have seen this as an issue from a very large provider's media IP's |
| 18:07.14 | Synthase_ | And I didn't read the entire description...disregard. |
| 18:09.11 | Synthase_ | Have you looked for which call leg is missing the expected RTP? |
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| 19:18.02 | uluatu | Synthase_: Yes I did a tshark dump. And the RTP is missing for the Rx traffic to the agent. But the recording is fine. Bothe legs are ok. The RTP is coming to Asterisk, but for some reason it is not deliverd to the Agent. |
| 19:18.55 | uluatu | I can hear the Agent telling the customer that he is not listen to him, but I also can hear the customer talking during this time. So the record is fine. |
| 19:24.11 | Synthase_ | And you've done the same for Asterisk-->Agent's SIP endpoint? Asterisk is bridging two call channels, ruling out one, does not make the other faultless. |
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| 19:29.20 | uluatu | Im sorry. I didn't understand what you want to say. |
| 19:30.00 | uluatu | Another point is that when this problem occur, every Agent get the same problem at the same time. So, this affects everybody that is on call. |
| 19:30.55 | uluatu | I tried to find bugs related to that, but I didn't get success. This problem is also happening with certified version. |
| 19:45.57 | Synthase_ | uluatu: A call that traverses from an ITSP trunk to an Asterisk server, to a SIP endpoint is in fact two call channels bridged together. Seeing no problems on the ITSP to Asterisk leg does not mean the Asterisk to endpoint leg is not having intermittent issues. Rule out the phone LAN as well. |
| 19:49.50 | Synthase_ | Keep in mind I am making some assumptions as you have not defined basic network/call topology. |
| 19:50.17 | uluatu | Ok, But the same problem is happening on 5 diferrent servers in 5 different locations with 5 different ITSP. |
| 19:51.05 | uluatu | the phones are on LAN. |
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| 20:09.16 | rrittgarn | Going through reports of dropped calls on a PRI and I'm finding: [2015-03-04 14:46:03] DEBUG[41747][C-000025f3] dsp.c: Requesting Hangup because the busy tone was detected on channel DAHDI/i1/3409986341-1b4c |
| 20:09.32 | rrittgarn | do I need to contact the provider of the PRI or digium as it is a digium card? |
| 20:10.07 | rrittgarn | i think this may be causing hangups mid conversation - I'm recording the calls with MixMon, logging with CDR / CEL, full debugging enabled, as well as pri debug... LOTS of data... |
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| 20:50.58 | SuperNull | :| go to asterisk log to see why it died: ERROR[30309] astobj2.c: refcount -1 on object 0x7f474c371598 |
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