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| 04:30.58 | Penguin | Anyone ever use a Valcom device for overhead paging? I connected the thing to a Cisco ATA, but it goes off hook and blasts a dial tone through the overhead speakers, then goes to congestion tone blasting, then finally goes silent but stays off hook. |
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| 05:01.52 | phix | Penguin: fun |
| 05:01.57 | phix | I use snome <3 |
| 05:01.59 | phix | snom* |
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| 05:43.16 | Penguin | phix: You have a Snom overhead paging device? |
| 05:45.50 | phix | I have set one up before, I don't own one though |
| 05:46.09 | Penguin | I can't determine if this Valcom is supposed to be connected to an FXS or FXO port. |
| 05:48.29 | Penguin | The manual talks about "PABX analog station port," "page port," and "system central office port." |
| 05:48.31 | phix | an FXS port that has FXO signalling on it |
| 05:49.13 | Penguin | I don't know what they mean when they use these terms. It's almost like they don't know what kind of port it requires. |
| 05:52.09 | Penguin | And the device has a battery feed switch for loop start. |
| 05:52.11 | Penguin | So confusing! |
| 05:55.44 | phix | hmmmm, coffee ime |
| 05:55.45 | phix | time* |
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| 06:08.39 | Penguin | Check this out... |
| 06:08.41 | Penguin | Valcom V-2001A |
| 06:08.42 | Penguin | connects to an FXO port, so more suitable for interface with PBX or other phone system |
| 06:08.58 | Penguin | I found this on voip-info.org. |
| 06:09.09 | phix | nice |
| 06:09.29 | Penguin | If it connects to an FXO port, that's not something a regular phone ATA provides. |
| 06:09.38 | Penguin | They sent the wrong type of ATA to the site. |
| 06:10.19 | Penguin | They sent a Cisco ATA with two ports to connect phones to it. Those would be FXS ports on the ATA. |
| 06:10.30 | Penguin | That would be why it isn't working. |
| 06:11.13 | phix | hmmm strange |
| 06:11.16 | phix | BBL work |
| 06:15.32 | Penguin | At least I figured out part of the problem. I'll have to let the client know so they can get the customer to buy a proper FXO gateway device. |
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| 06:29.33 | Penguin | They have a Cisco SPA112, which is the equivalent of the Linksys SPA2102. Someone said on some forum that they are using the SPA2102 with the Valcom. That means the Valcom must be able to be changed from needing an FXO port to being able to work with an FXS port, but I guess the ATA is going to have to provide auto-answer. |
| 06:43.56 | fury_ | what format does the 'originate' command line command use for phone numbers? in other words, obviously a US dial is "15555555555". but what about international dials? what's the format expected there, and where is the relevant documentation if any? |
| 06:44.20 | Penguin | The originate command doesn't have a number format. |
| 06:44.55 | Penguin | Are you running this call through your own dialplan? |
| 06:46.47 | fury_ | sure! that all works. if you can envision it, an example would be "originate SIP/15555555555@<provider> extension config@starting_extension" from the asterisk command line |
| 06:47.03 | fury_ | but what if I wanted to dial an international phone? no '+'? |
| 06:47.11 | fury_ | does it depend on my SIP provider? |
| 06:47.56 | Penguin | The number format in that position of the command is determined by the peer on the far end. And your syntax is potentially wrong. |
| 06:48.15 | Penguin | If you have <provider> defined in your sip.conf, you would use SIP/provider/number |
| 06:48.16 | fury_ | so it depends on my SIP provider |
| 06:48.20 | Penguin | not number@provider |
| 06:48.45 | fury_ | I don't think I'm doing this right and this is really just in the exploratory phase |
| 06:48.54 | Penguin | number@undefined.domain.com |
| 06:48.59 | Penguin | SIP URI format |
| 06:49.35 | Penguin | And please stop using the phrase "SIP provider." It's an ITSP. |
| 06:50.24 | fury_ | fair enough, apologies - I'm not really familiar with terminology |
| 06:50.29 | Penguin | ~itsp |
| 06:50.29 | 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. |
| 06:51.00 | fury_ | ~itsplist-us |
| 06:51.00 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.broadvoice.com, http://www.jnctn.com, http://www.sipstation.com, http://vitelity.net, http://voip.ms and http://flowroute.com |
| 06:51.18 | fury_ | thanks that'll be useful |
| 06:51.20 | Penguin | The term "SIP provider" would be the equivalent to saying "HTTP provider" when you're talking about your ISP giving you the ability to surf the web. |
| 06:51.51 | fury_ | good analogy, and now I feel like an idiot ;) |
| 06:52.11 | Penguin | You'll get over that. |
| 06:53.04 | Penguin | Anyway, your dial format for your ITSP could be something like 011 or 00 followed by the area code and phone number. |
| 06:53.12 | Penguin | Mine will accept both. |
| 06:53.42 | fury_ | yeah I gather that, kinda like physically dialing on a phone |
| 06:55.44 | Penguin | If I wanted to call a particular company in the UK, I might use 00 44 1352 761687 or 011 44 1352 761687. |
| 06:56.11 | Penguin | SIP/itsp/00441352761687 |
| 06:56.15 | Penguin | SIP/itsp/011441352761687 |
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| 07:04.11 | Penguin | When you create your dialplan, accept the international numbers in that same format and pass the call to your ITSP using the same extension. |
| 07:15.38 | fury_ | I have a whole dial plan, I'm just using a hack-ish way to initiate an outbound call |
| 07:15.44 | fury_ | if you have any suggestions I'm all ears |
| 07:16.25 | fury_ | the hack-ish way is literally "bash -c 'asterisk -x \'originate ...\''" |
| 07:21.20 | Penguin | Why the bash -c? |
| 07:30.59 | Penguin | You're taking too long to answer, and I'm heading for bed. So I'm just going to leave you with this thought and head out... If you're doing it from a shell script or cron, just specify #!/bin/bash and then use /usr/sbin/asterisk -rx "channel originate SIP/peer/extension whatever whatever" |
| 07:31.13 | Penguin | You don't need all the quoting and escaping of quoting. |
| 07:31.41 | Penguin | Double quotes around the command for asterisk to execute should be enough. |
| 07:31.48 | Penguin | And I'm out! |
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| 08:21.32 | fury_ | Penguin: I'm so sorry for taking that long, I got called away on an important personal phonecall. :( thanks for your help :) |
| 08:22.44 | fury_ | Penguin: also the extra quoting and escaping was also due to the fact that it's executed on-the-fly in python, and I was also escaping it so *you* could parse it |
| 08:23.07 | fury_ | eg if you take everything but the double quotes and paste it in a bash prompt it would work |
| 08:23.38 | fury_ | and the 'bash -c' was to sort of imply that I was calling bash manually and not in a script (which would normally use a shebang like you suggested) |
| 08:23.55 | fury_ | I do wonder what -r is though, first glance it's probably helpful |
| 08:24.31 | fury_ | ah nope, just "connect to asterisk on this machine" |
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| 09:41.25 | EmleyMoor | Are there any significant deprecations between Asterisk 1.8 and 11? |
| 09:45.59 | EmleyMoor | (anything that will break, rather than warn) |
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| 11:12.08 | EmleyMoor | I've upgraded my asterisk to 11.13, but my Jabber/XMPP connection isn't coming up any more and I can't see why... |
| 11:15.58 | EmleyMoor | It's within the same host... any way to "trigger" it to see why it's not working? |
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| 13:02.13 | StaRetji | hello |
| 13:03.46 | StaRetji | I need help, operator is playing IVR every 3-4 calls asking to press digit in order to proceed with the call, if wrong code, they will hangup. Now, problem is, they charge this IVR, which is unacceptable. I need help to prevent this |
| 13:04.19 | StaRetji | is there a way to detect ivr and keep in no-answer session |
| 13:04.37 | StaRetji | for me is even better to hangup if I can detect ivr |
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| 13:14.54 | SpaceAceX | So how do i setup an analog outside line to an inside analog line for a house usin asterisk as the answering machine and other cool features? |
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| 13:45.49 | WIMPy | SpaceAceX: Get a card with a FXS and a FXO module. If you want it cheap you can use an ATA with both FAX and FXO as well. |
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| 14:18.10 | EmleyMoor | How do I trace why my asterisk setup is no longer connecting to my XMPP server? I can't see any reason in the log, but it hasn't worked since I upgraded earlier. I have checked the config. |
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| 15:29.40 | Penguin | fury_: I looked at the man page and it kind of appears that -x implies -r. Not totally sure on that. Historically, we always used -rx to have asterisk execute a command, but it looks as if -x is enough nowadays. |
| 15:31.23 | fury_ | Penguin: yeah, wasn't what I was looking for. I'm looking for something to make it return from asterisk -x immediately, I'm not sure that's really a thing. what would you recommend for remotely communicating with asterisk? |
| 15:31.43 | fury_ | if I needed to trigger an originate from a remote machine for example, or wanted to avoid asterisk -x |
| 15:32.01 | Penguin | fury_: I use a shell script and call asterisk -rx most of the time. There's also AMI if you like that sort of thing. |
| 15:32.54 | Penguin | AMI, or the manager interface, can do neat things that asterisk CLI cannot do. |
| 15:33.14 | Penguin | And, if you're in the coding mood, why not just create call files with your python script? |
| 15:33.25 | Penguin | ~call file |
| 15:33.25 | infobot | ACTION looks around and then screams out file as loudly as possible |
| 15:33.34 | Penguin | ~call files |
| 15:33.34 | infobot | ACTION looks around and then screams out files as loudly as possible |
| 15:33.41 | Penguin | gah! |
| 15:35.25 | file | raises yebeor |
| 15:35.27 | file | or eyebrow |
| 15:35.30 | Penguin | ~callfile |
| 15:35.30 | infobot | it has been said that callfile is a text file that when placed in the correct directory makes Asterisk make an outgoing call. See http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out#Callfiles |
| 15:35.55 | fury_ | ah |
| 15:36.11 | fury_ | just place it in /var/spool/asterisk/outgoing/ and I'm done |
| 15:36.17 | Penguin | Don't point your yebeor at me! |
| 15:36.53 | WIMPy | No. Don't write it there. Write is somewhere else on the same FS and when done, move it there. |
| 15:37.26 | Penguin | I generally write the file in something like /var/tmp/ and then move it to the outgoing spool dir. |
| 15:38.05 | WIMPy | That would fail here. |
| 15:38.25 | Penguin | Every system is potentially different. |
| 15:38.34 | WIMPy | I prefer to use /var/spool/asterisk/tmp. That way it's obvious. |
| 15:38.41 | Penguin | I didn't use /tmp because /tmp and /var might not be the same fs. |
| 15:38.56 | fury_ | yeah I'm aware of my system layout hehe |
| 15:39.01 | fury_ | it's almost always different (it's not a linux) |
| 15:39.07 | Penguin | /var/spool/asterisk/tmp could also be reasonable. |
| 15:39.41 | fury_ | why are we doing that, though, by the way? creating it then moving it? |
| 15:40.12 | WIMPy | Otherwise there's a good chance of it being read before it's complete. |
| 15:40.24 | WIMPy | Thus not doing what you want. |
| 15:40.36 | WIMPy | The burden of multitasking. |
| 15:40.38 | fury_ | er |
| 15:40.41 | WIMPy | :-) |
| 15:40.43 | fury_ | yeah, didn't think of that. |
| 15:46.50 | Penguin | It should have been mentioned in that page infobot linked to. |
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| 15:56.23 | WIMPy | You can edit it. |
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| 16:55.53 | fury_ | Penguin: turns out python has a nice little API for asterisk called pyst. got that working and can now originate calls perfectly using AMI without needing file writing hackery. here's some python: http://dpaste.com/1PF2F73 |
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| 17:10.16 | EmleyMoor | What has changed in the xmpp/jabber suuport between 1.8 and 11? |
| 17:13.37 | EmleyMoor | (something must have, because the upgrade stopped it working, with no obvious errors) |
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| 17:26.54 | file | are you using autoload? |
| 17:33.50 | EmleyMoor | Er... I either always was or never was - not sure |
| 17:34.20 | EmleyMoor | Yes, I am using autoload |
| 17:34.37 | file | res_jabber was replaced by res_xmpp which is a drop-in replacement |
| 17:35.00 | EmleyMoor | ... |
| 17:35.58 | EmleyMoor | It's loaded, it's just not working - kills off the call in which I try to use it |
| 17:36.17 | file | what's the complete console output from start to failure? |
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| 17:37.45 | EmleyMoor | Hold on - will present that if I still need it, shortly |
| 17:38.51 | EmleyMoor | Ah - looks like a conf file name change was the problem |
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| 17:44.18 | EmleyMoor | Indeed it was |
| 17:44.45 | EmleyMoor | Not so much drop in as "drop in and twist slightly" |
| 17:45.07 | file | you shouldn't have had to change the file name |
| 17:45.13 | file | provided an xmpp.conf is not present it will use jabber.conf |
| 17:45.38 | EmleyMoor | file: Well, that would explain it then - the package installed an xmpp.conf |
| 17:47.23 | file | that would do it |
| 17:49.31 | EmleyMoor | It hasn't solved anything I was trying to solve by the upgrade, though... was hoping to be able to do faxes in a logical fashion |
| 17:52.21 | EmleyMoor | In fact I think 1.8 was better at faxes |
| 17:56.03 | ChannelZ | Death to faxes! |
| 17:57.00 | EmleyMoor | ChannelZ: Ideally, yes. Unfortunately I occasionally have to deal with the NHS |
| 17:58.24 | file | nothing beats an analog line for faxing |
| 17:58.31 | file | except email. |
| 17:58.45 | EmleyMoor | file: Which they don't accept |
| 18:22.07 | coppice | ChannelZ: is that volume 2 of an encyclopedia? |
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| 18:36.38 | acovrig | Is it possible to have my server use GoogleVoice/Motif as a trunk and still have my cellphone/gmail ring (as-in ring all instead of having Asterisk catch the call instantly)? |
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| 18:48.24 | [TK]D-Fender | * catches the call when you tell it to |
| 18:49.26 | [TK]D-Fender | (answer) |
| 18:53.21 | acovrig | [TK]D-Fender: it seems to all the time, if I set the XMPP Priority to 127 my cell doesnât know about any inbound calls; if I set it to 0, Asterisk doesnât know about any inbound calls; is it possible to have them both ring at the same time? |
| 18:57.19 | [TK]D-Fender | no idea |
| 18:58.43 | acovrig | [TK]D-Fender: also, any idea if I have my asterisk server with a priority of 127 so it grabs the calls instantly, what happens if the server goes down, is it possible to restore GV to my phone if I loose access to my asterisk server? |
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| 19:54.59 | StaRetji | operator is playing IVR every 3-4 calls asking to press digit in order to proceed with the call, if wrong code, they will hangup. Now, problem is, they charge this IVR, which is unacceptable. I need help to prevent this |
| 19:55.20 | StaRetji | is there a way to detect IVR and hangup? |
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| 20:02.55 | StaRetji | http://www.voip-info.org/wiki/view/Asterisk+cmd+AMD |
| 20:03.32 | StaRetji | maybe I can detect any IVR and hagup, if there's IVR, else allow call |
| 20:03.51 | StaRetji | hehe, monologue, but it helps |
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| 21:19.01 | apb1963 | I would like to do an alias like this: test carrier1=originate SIP/900807 extension <EXT><phone>@external ...... first, please confirm syntax is correct? Second, how would I pass in <EXT>, and possibly also <phone>, from the CLI ? |
| 21:20.59 | Penguin | I have no idea what <EXT> and <phone> are. |
| 21:23.40 | fury_ | is it possible to generate a dialplan on-the-fly, and use it with asterisk AMI? in other words, without needing the dial plan in extensions.conf? |
| 21:23.42 | Penguin | The syntax is: channel originate <tech/data> extension [exten@][context] |
| 21:23.49 | apb1963 | Penguin: they're data I pass in |
| 21:23.59 | apb1963 | parameters |
| 21:24.28 | Penguin | In your example, 900807 nees to be the name of a SIP device on your asterisk. |
| 21:24.30 | apb1963 | yes, <tech/data> and exten |
| 21:24.46 | apb1963 | yes, that's the sip device |
| 21:24.47 | Penguin | And I have no idea what <EXT> and <phone> are. |
| 21:25.06 | apb1963 | those are my own names for <tech/data> and exten |
| 21:25.12 | Penguin | But extension needs to have extension@context. |
| 21:25.48 | apb1963 | so, how would I pass those 2 items into the alias from the CLI? |
| 21:25.55 | apb1963 | So that it substitutes |
| 21:26.16 | Penguin | argv[1-4] perhaps |
| 21:26.35 | Penguin | $1, $2, etc. |
| 21:26.59 | Penguin | That's what I would do with a shell script. |
| 21:27.05 | apb1963 | so.... either a C reference or a shell var |
| 21:27.35 | apb1963 | heh. I just crashed * ... it doesn't like my alias |
| 21:32.05 | apb1963 | and that's w/out any vars :( |
| 21:33.29 | apb1963 | oh interesting... it only didn't like it when I did a core reload |
| 21:33.38 | apb1963 | but a fresh startup, it's fine and works |
| 21:36.23 | apb1963 | hmmm now core reload works. weird |
| 21:39.07 | apb1963 | Penguin: Nope, neither $1 {$1} nor argv[1] worked for me. Thank you anyway :) |
| 21:39.57 | apb1963 | waits for inspiration |
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| 22:18.08 | apb1963 | checks his watch... doesn't look like inspiration is coming. Stood up again! |
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