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| 10:58.07 | asari3 | I'm reading about Monitor on the wiki (http://www.voip-info.org/wiki/view/Monitor+setup+sample). Are there any way to store sound file without decoding (e.g. in a pass-thru manner)? In my setup, all caller/receiver should use speex codec, so I wish I can store sound file as-is in speex format. |
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| 17:10.26 | ruied | hello. I want to do something like: "originate sip/23 application Playback tt-monkeys" but want to Set(_ALERT_INFO_=ZZ) and SipAddHeaded(YY) so I can do it with Page(). what is the best way to do it? |
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| 17:14.46 | ruied | I want to Page a recorded sound to a phone from a cron job. Can set with "asterisk -rx originate and Playback()" but I want to be like a paging. The remote phone mus play the sound without a person answering it... |
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| 17:20.15 | scv | anybody ever seen voicemail delivered via email being distorted yet okay when retrieved via phone? |
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| 18:16.18 | igcewieling | scv: what format are the files? |
| 18:17.57 | igcewieling | ruied: try local channels |
| 18:22.59 | Simon-- | hrm, is there any trick to convincing the config parser to accept a literal newline as the content of a configuration setting? I see some have certain un-escaping, but I just mean the base parsing |
| 18:23.29 | Simon-- | otherwise I'll propose a patch to include "Auto-Submitted: auto-generated" in app_voicemail emails ;) |
| 18:26.52 | igcewieling | you've tried \n ? |
| 18:31.00 | igcewieling | take a look at voicemail.conf.sample the e-mail notification section shows \n as new lines. don't know if it will help your specific issue |
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| 18:57.43 | Simon-- | nope.. I was hoping it was dumb enough that I could inject a header by fooling it |
| 18:57.50 | Simon-- | but somebody actually cared when coding it..damn :) |
| 18:57.58 | Simon-- | it tries to mime-encode it if there are control characters ,etc |
| 18:58.18 | Simon-- | and it only really parses \n out of emailbody and emailsubject |
| 18:58.35 | Simon-- | and it's not doing substitutions anywhere useful that I can tell |
| 19:01.04 | igcewieling | drat |
| 19:01.35 | igcewieling | You can use a custom MTA, would not be too tough to create a perl or php script to handle it. |
| 19:01.55 | Simon-- | well, I'd argue that the lack of that header is a bug and probably I should just submit a patch for that |
| 19:02.36 | Simon-- | found, e.g., http://issues.freepbx.org/browse/FREEPBX-11783 |
| 19:02.39 | igcewieling | Simon--: might be a good idea, but 1) expect to have to fight for it and 2) it might not make it into Asterisk until Asterisk 15 released. |
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| 19:03.29 | igcewieling | BTW, I find it is rarely helpful to fight with Asterisk, just work around the issue and don't worry about it. |
| 19:04.30 | igcewieling | I'm so sick of Asterisk's SIP auth that I allow guest calls and handle auth in an AGI script now. |
| 19:04.48 | igcewieling | <PROTECTED> |
| 19:05.49 | Simon-- | hm, is it really that bad? we submitted some stuff years ago for, e.g., saving queue state |
| 19:05.59 | Simon-- | that was far more invasive |
| 19:06.43 | igcewieling | Simon--: it is terrible for our uses, I don't know about others. IIRC the issue is that host= is used for ACL not permit/deny when using type=friend |
| 19:06.53 | Simon-- | hehe yeah |
| 19:07.07 | igcewieling | oh, you mean for patches. Digium has gotten more corporate in the last 5 years or so. |
| 19:07.54 | igcewieling | I'm pretty sure I wanted to accept calls from any hosts matching permit/deny and make outgoing calls to the host= host. |
| 19:09.22 | Simon-- | is it still default insecure=yes? ;) |
| 19:09.40 | igcewieling | insecure was deprecated in Asterisk 1.4 or 1.8 |
| 19:10.06 | Simon-- | shows you how often I look at it |
| 19:10.11 | igcewieling | sorry, insecure=yes was deprecated. there are other values allowed. 'host' and 'port' I think. |
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| 20:22.04 | lwlvl | I do not always get a dialtone, when I call somebody...I use nat="force_rport,comedia" any ideas? |
| 20:23.28 | igcewieling | Dialtone is provided by the SIP device, not the PBX. You can have dialtone on a SIP phone without a connection to a PBX. Not always getting it is... odd. |
| 20:24.00 | lwlvl | igcewieling, I mean the tone, that is coming when a device is ringing.... |
| 20:24.37 | robmal | So... ringtone? |
| 20:24.48 | igcewieling | Is the asterisk server behind NAT? |
| 20:25.01 | lwlvl | igcewieling, my calls are going like this: hook off....tone...dial number....silence.....when the receipient takes the call, I can hear his voice....but not the tone, indicating that the remote phone is ringing.... |
| 20:25.06 | lwlvl | igcewieling, yes. |
| 20:26.49 | igcewieling | If using chan_sip, take a look at the progressinband and prematuremedia options. If you are using pjsip, then someone else will have to help you. |
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| 20:28.08 | lwlvl | igcewieling, okay....I'll have a look at it |
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| 20:41.23 | igcewieling | lwlvl: btw, the correct term is "ringback" |
| 20:45.36 | lwlvl | igcewieling, progressinband=yes did the job...but know i guess asterisk is sending the ringback no matter what the remote-side is doing :) |
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| 22:51.43 | robink | I have an install of Asterisk 14.0.0 which complains tshuly at startup: asterisk: symbol lookup error: /usr/lib64/asterisk/modules/res_pjsip.so: undefined symbol: pjsip_tls_transport_start2 |
| 22:51.46 | robink | *thusly |
| 22:52.02 | robink | Is there an obvious cause for this problem? |
| 22:53.07 | robink | I'm pretty sure I compiled pjproject with IPv6 support. |
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| 22:58.34 | *** join/#asterisk ctcx (~user@unaffiliated/ctcx) |
| 23:00.00 | ctcx | Hello. Sorry if it's a stupid question, but I have already done a previous search. Is there a way to send/receive text messages with asterisk? By using the SIP (or SIMPLE?) protocols? |
| 23:00.29 | ctcx | I read it seems possible, but it's always talked about command-line stuff... |
| 23:05.28 | ctcx | But not by using the phone device itself, but by using perhaps a "client" for pc that could connect to asterisk's SIP |
| 23:05.40 | robink | ctcx: I think Asterisk will gate SIP messages (the textual kind) between SIP clients, but gating them onto an SMS network is another matter. |
| 23:06.15 | robink | ctcx: The other issue is a number of SIP clients don't support the messaging/IM protocol. |
| 23:06.17 | ctcx | No, no, I'm not talking about SMS. |
| 23:06.42 | robink | ctcx: Right, and you'd need to find someone who'd gate text messages in *any* over-IP form to SMS. |
| 23:06.57 | ctcx | Yes, that I already read indeed. |
| 23:07.07 | ctcx | So no, nothing about SMS. |
| 23:07.16 | robink | ctcx: Twilio does that, but they have their own proprietary API for doing so, so wiring something like that up to Asterisk would be a challenge. |
| 23:07.24 | robink | ctcx: I guess I'm not sure what you're asking then. |
| 23:07.41 | ctcx | But I thought it was possible when I was mentioned SIP protocol's extension SIMPLE. |
| 23:07.52 | robink | ctcx: There is no "hidden magic" within Asterisk that can be leveraged for gating messages to the SMS network via a standard PSTN trunk. |
| 23:08.14 | ctcx | Made me think "oohh, so the protocol asterisk runs over also supports text aside just voip" |
| 23:08.19 | robink | ctcx: It's not an issue of protocol, it's an issue of whether or not you can get someone trunking to PSTN to gate SMS messages. |
| 23:08.41 | robink | ctcx: Asterisk could natively speak SS7, you still would have to find someone to talk to on the other end. |
| 23:09.46 | ctcx | robink: er... I thought we had made clear I'm not into SMS stuff, but I'm making a difference between "chat text" and "SMS", being the earlier what I look for and the later only possible through carrier services. |
| 23:10.26 | ctcx | robink: mm... may I try to make myself clearer before becoem annoying for everyoen here? |
| 23:10.30 | ctcx | *everyone |
| 23:10.30 | robink | ctcx: ...and like I said, Asterisk will happily gate SIP messages between clients. |
| 23:10.45 | robink | ctcx: The issue there is client support, which in a way is another form of the problem with gating SMSes. |
| 23:11.03 | robink | ctcx: i.e. Asterisk will happily send information along, you need someone to talk to. |
| 23:11.13 | ctcx | robink: and that's what I was about to ask: *which client for pc to use?* |
| 23:11.29 | robink | ctcx: Well if you're running Linux, Empathy/Telepathy would be a good bet. |
| 23:11.44 | robink | ctcx: ...or anything libpurple-based is probably OK too. |
| 23:12.10 | robink | ctcx: I actually haven't looked into SIP clients for Windows in a long time, and the only ones I can think of offhand support VoIP and not much else. |
| 23:12.20 | robink | ctcx: Pidgin for Windows *might* have SIP support compiled in; I'd have to check. |
| 23:12.41 | ctcx | <robink> ctcx: i.e. Asterisk will happily send information along, you need someone to talk to. You mean I need precisely the right client application capable of receiving/sending the messages, right? |
| 23:12.48 | robink | ctcx: Yep. |
| 23:12.59 | robink | ctcx: On both ends. |
| 23:13.17 | robink | ctcx: i.e. the recipient needs to be running software capble of parsing/displaying SIP messages. |
| 23:13.58 | ctcx | robink: I think I found one of my understandign issues: you say "on both ends". And that's right, but for me is more like "highly obvious". I'd just give the 2 speakers the same client! |
| 23:14.11 | robink | ctcx: Provided you're able to do that. |
| 23:14.23 | ctcx | Why not? If all are windows workstation? |
| 23:14.29 | ctcx | workstations |
| 23:14.38 | robink | ctcx: I didn't want to make any assumptions about how much control you had over which SIP clients people ended up using. |
| 23:15.03 | robink | ctcx: It's not a matter of compatibility or ease of deployment, it's more a matter of "Is this person going to let me dictate what kind of software they run?" |
| 23:15.19 | ctcx | Aaahhh. |
| 23:15.29 | ctcx | I thought it was kind fo obvious, but it wasn't. Sorry. |
| 23:16.13 | ctcx | Well, if I'm part-time working for the TI boss of an office, I think it could be possible... |
| 23:16.42 | robink | ctcx: but I mean if you have total control over the entire communications platform, you can do whatever you want. Route calls through Jingle and just have any remaining clients connecting via SIP have their messages routed over XMPP. |
| 23:17.11 | robink | ctcx: I guess. I tend to assume a little more in the way of platform agnosticism even if I have that degree of control. |
| 23:17.27 | ctcx | robink: I'd like to make myself clearer anyway, if I may. Starting with the fact that I'd help to install the whole asterisk server from scratch. |
| 23:17.44 | ctcx | Server that would serve the entire office. |
| 23:17.57 | robink | ctcx: That won't make a huge difference. Just about any stock deployment of Asterisk should be able to pass SIP messages around. |
| 23:18.13 | robink | ctcx: Again, since you also control the server, there's nothing stopping you from tying the whole mess into an XMPP gateway. |
| 23:18.19 | robink | ctcx: Or whatever. |
| 23:18.36 | ctcx | robink: aahh. Yes, thanks. Sorry. |
| 23:20.39 | *** part/#asterisk kharwell (kharwell@nat/digium/x-wevcanosuhhyipzi) |
| 23:21.13 | robink | ctcx: So if we're talking pie-in-the-sky total control over both the client and server ends, direct client<->client messaging with Asterisk should be no problem. I'm just not coming up with a SIP client (besides Pidgin, possibly) for Windows that supports SIP (textual) messages. |
| 23:21.49 | robink | ctcx: It's once we get to the point of trying to interface with conventional & proprietary messaging/signalling/calling platforms that things start to get more interesting. |
| 23:22.13 | robink | ctcx: because you have waaaaay less control once you're trying to graft any VoIP setup onto the telephone network. |
| 23:22.36 | ctcx | robink: still attempting to make myself clearer, boss tells me "we need an asterisk server for having ip phones. Ok. But we'll also need to have a way to text chat. Using asterisk itself if possible". |
| 23:22.42 | ctcx | robink: I already thought about openfire and clients like spark and pidgin. All 3 have sip support through plugins. But this way chat would be running over openfire server, not asterisk itself. If anything asterisk would be like a satellite extension of openfire, not the other way around which is what we seek. |
| 23:23.04 | robink | ctcx: Er, you'd have Pidgin et al connect to the Asterisk server instead of the OpenFire server. |
| 23:23.44 | robink | ctcx: ...and if you wanted, you could fully integrate Asterisk and OpenFire, so that people could connect to, say, OpenFire, and message/call someone connected to Asterisk. |
| 23:24.12 | robink | ctcx: So that anybody who refuses to migrate to Asterisk or switch to a client that supports SIP will still be able to converse with people who've migrated to Asterisk. |
| 23:24.59 | ctcx | robink: sir, you have knowledge about openfire stuff as well? |
| 23:25.00 | robink | ctcx: Giving one of the two services primacy is as much an issue of policy as it is a technical issue. |
| 23:25.03 | ctcx | robink: sir, you have knowledge about openfire stuff as well? |
| 23:25.05 | ctcx | (OUCH) |
| 23:25.16 | robink | ctcx: Very little. I've used it, but not deployed/adminned it. |
| 23:25.38 | robink | ctcx: I tend to prefer jabberd2, provided I'm not needing to do things like XML-RPC. |
| 23:26.28 | robink | ctcx: I *do* know that Asterisk will happily gate to Jingle/XMPP/WebRTC, so you literally can have a web-browser-based XMPP client connect either to something like OpenFire or directly to Asterisk and be treated like a first-class citizen. |
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| 23:27.05 | robink | ctcx: Then the only requirement for a client platform is a web browser. |
| 23:27.18 | robink | ctcx: No softphones, no VoIP-capable IM clients, just a web browser. |
| 23:28.47 | robink | ctcx: and provided you have sufficient trust in the JS MediaSource/WebRTC/WebCrypto APIs, you have direct and complete "control" over what software clients are running at all/any time(s). |
| 23:29.08 | robink | ctcx: Need to "update" the client platform? Just push the new code to the web server. |
| 23:30.13 | ctcx | robink: the thing is, if all and each of the office users are given an ip phone device, asterisk-openfire full integration would be a bit pointless, wouldn't it? That's why I'd seek more like the first thing you mentioned: use pidgin to purely connect to asterisk, hence using the very same logins as the asterisk users/ip phones. |
| 23:30.41 | robink | ctcx: Sure, if you have no need of an XMPP server. |
| 23:31.58 | ctcx | If not, then what you mentioned would apply: |
| 23:31.58 | ctcx | <robink> ctcx: So that anybody who refuses to migrate to Asterisk or switch to a client that supports SIP will still be able to converse with people who've migrated to Asterisk. |
| 23:32.17 | ctcx | Though in that case I think i'd use Spark for client in ALL workstations. |
| 23:32.20 | robink | ctcx: Right. |
| 23:32.23 | ctcx | Spark can message/call |
| 23:32.30 | ctcx | And even conferences |
| 23:32.42 | robink | ctcx: Sure, if you already have a SIP client in mind that people don't mind using, go for it. |
| 23:32.44 | ctcx | (Though no idea how well/mediocre) |
| 23:32.59 | robink | ctcx: I mean it sounds like you already have a decent idea of what you want to do and how you want to do it. |
| 23:33.18 | robink | ctcx: All I can tell you is that Asterisk will happily pass around client<->client messages, and can integrate with other chat/VoIP platforms if you need it to. |
| 23:33.48 | robink | ctcx: Integrating with older-school telephony without a lot of fuss is a different matter. |
| 23:34.15 | ctcx | robink: I hope so, thanks. But then my doubts would be: how on earth can I connect pidgin client purely to asterisk, given I already have some users and extensions? And, would old asterisk 11 support all of this? |
| 23:34.32 | ctcx | I have never used pidgin |
| 23:34.45 | robink | ctcx: Asterisk 11 would support a connection from Pidgin just fine, provided the build of Pidgin you're using supports SIP. |
| 23:35.05 | ctcx | robink: I'd get latest stable pidgin of course! |
| 23:35.20 | robink | ctcx: libpurple (the library underpinning Pidgin) has had SIP support for a while, it just was not enabled by default at the outset and I don't know if current Windows builds ship with it enabled (I'm guessing they do, but I don't know for sure). |
| 23:35.22 | ctcx | robink: by chance, have you used pidgin? |
| 23:35.26 | robink | ctcx: Sure. |
| 23:35.31 | ctcx | OOoooooohhhhhhhhhhhh |
| 23:35.36 | robink | ctcx: I still use libpurple, just not via Pidgin. |
| 23:36.06 | robink | ctcx: It works as a SIP client. Right now I'm using SofiaSIP via Telepathy to connect to my SIP server on my Linux desktop, but IIRC Pidgin does just fine. |
| 23:36.42 | robink | ctcx: ...and libpurple is the only way I can reliably get on AIM for the one-or-two people I know who still use it. |
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| 23:38.09 | ctcx | robink: do you happen to accept pm's? |
| 23:38.24 | robink | ctcx: Sure, I read PMs. I usually respond to them, too :-) |
| 23:38.54 | ctcx | I just sent you one... |
| 23:39.26 | robink | I saw it (and responded) |
| 23:39.27 | ctcx | robink: ah, thanks. |
| 23:39.32 | ctcx | sorry |
| 23:39.35 | robink | np |
| 23:40.54 | ctcx | robink: for windows workstations, and for my current needs in general, would you advise indeed for a libpurple client? |
| 23:45.22 | ctcx | robink: ??? |
| 23:45.39 | robink | ctcx: It'd be worth a try. |
| 23:46.10 | robink | ctcx: I'd see how it feels and evaluate how likely it is to meet your use case before you standardize on it. |
| 23:46.23 | robink | ctcx: If we're talking a department-wide deployment, I'd strongly recommend taking it for a spin first. |
| 23:46.30 | robink | ctcx: but worth looking into? Absolutely. |
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