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| 04:01.15 | deal | hey folks. ive just done some simple stuff with asterisk (through freepbx) but now looking to do some more interesting stuff |
| 04:01.53 | deal | can i get some advice on which direction to go in order to do things like displaying incoming calls and then maybe sending those calls to p[articular extensions |
| 04:02.20 | deal | ideally some sort of web interface so some person in the office can see the calls and make decisions based on who is calling |
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| 04:33.28 | drmessano | FOP2 |
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| 05:03.16 | deal | cool ill check it out |
| 05:03.21 | deal | any alternatives to that? |
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| 05:58.17 | drmessano | I wouldnt even look |
| 05:58.32 | drmessano | FOP2 is pretty much the beez kneez |
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| 06:50.12 | wyoung | .o/ |
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| 09:08.00 | TheGallopingFox | at the moment my setup uses 2x voices, 1 for the menu and 1 for voicemail etc, i wonder if there are enough samples to create a user menu in asterisk core sounds so i can use a single voice |
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| 09:27.18 | TheGallopingFox | i wonder if there is a way to use googletts.agi for all voicemail voices etc |
| 09:30.19 | TheGallopingFox | wow |
| 09:30.20 | TheGallopingFox | https://github.com/Uskur/Asterisk-TTS |
| 09:31.08 | TheGallopingFox | pity its old |
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| 09:40.25 | TheGallopingFox | doesnt work, would need to create a script what converts all the samples provided in core-sounds-en.txt using googletts |
| 09:45.37 | TheGallopingFox | then voicemail sounds will be consistent with the IVR menu |
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| 10:07.12 | TheGallopingFox | problem solved, enough samples in core extra sounds to create IVR menu! |
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| 11:26.08 | wyoung | TheGallopingFox: nice |
| 12:24.14 | TheGallopingFox | when i get a voicemail, there is a big delay before my polycom LED starts to blink alerting there is a new message, is there a way to speed that up? |
| 12:27.58 | fauxalliance | Set pollmailboxes = yes |
| 12:28.04 | fauxalliance | <PROTECTED> |
| 12:28.05 | fauxalliance | reload |
| 12:28.25 | fauxalliance | (vm configuration) |
| 12:29.17 | TheGallopingFox | fauxalliance: can that be put in voicemail.conf ? |
| 12:29.24 | fauxalliance | it should be |
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| 12:29.30 | TheGallopingFox | thanks |
| 12:29.52 | fauxalliance | no sweat |
| 12:29.55 | fauxalliance | send coffee |
| 12:30.04 | TheGallopingFox | []) |
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| 12:44.37 | TheGallopingFox | strange, pollfreq= has no effect |
| 12:44.50 | Samot | OK |
| 12:44.53 | Samot | JFC. |
| 12:45.25 | Samot | When a *NEW* message is left in the INBOX, Asterisk generates a NOTIFY for MWI |
| 12:45.29 | Samot | IMMEDIATELY |
| 12:46.04 | TheGallopingFox | i deleted the message, but the phone is still alerting me there is a new voicemail message |
| 12:46.11 | TheGallopingFox | that will go off but not for a good while |
| 12:46.19 | Samot | RIGHT |
| 12:46.25 | Samot | Because your are SUBSCRIBED to it |
| 12:46.34 | Samot | *you |
| 12:46.57 | TheGallopingFox | i thought the pollfreq would stop that |
| 12:48.01 | TheGallopingFox | maybe pollfreq is in minutes not seconds |
| 12:48.08 | Samot | pollmailbox is for when the voicemail is modified not through app_voicemail |
| 12:48.17 | Samot | I told you yesterday to READ the samples and the wiki |
| 12:48.17 | TheGallopingFox | oh |
| 12:48.26 | TheGallopingFox | ok will have a look |
| 12:48.28 | fauxalliance | like when I dump a message in every useers vmbox |
| 12:48.36 | TheGallopingFox | oh i see |
| 12:48.40 | Samot | When we went round and round about your voicemail formats |
| 12:48.54 | fauxalliance | was not privy to that shenanigans |
| 12:50.02 | Samot | Well... |
| 12:50.11 | Samot | Your suggestion of pollmailboxes was incorrect as well |
| 12:50.18 | Samot | Part of me jumping in here. |
| 12:51.16 | fauxalliance | VoicemailRefresh, whatever. |
| 12:51.47 | fauxalliance | BLF/MWI/ Good Luck. |
| 12:53.48 | Samot | TheGallopinFox: How long is this "delay |
| 12:54.52 | TheGallopingFox | its about 10-15 minutes |
| 12:55.35 | TheGallopingFox | i will time it properly right now |
| 12:56.48 | TheGallopingFox | email alert works straight away, timing now |
| 12:57.44 | TheGallopingFox | might need these options: |
| 12:57.45 | TheGallopingFox | https://www.voip-info.org/wiki/view/Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk |
| 12:58.12 | TheGallopingFox | most are on already |
| 12:59.43 | TheGallopingFox | but im missing: msg.mwi.1.subscribe="" |
| 12:59.55 | Samot | No. |
| 13:00.04 | Samot | It uses the host |
| 13:00.08 | TheGallopingFox | oh ok |
| 13:00.19 | Samot | OK |
| 13:00.21 | Samot | Listen.. |
| 13:00.33 | Samot | You SUBSCRIBE to voicemail notices. |
| 13:00.54 | Samot | When app_voicemail handes a new voicemail a NOTIFY is generated and sent |
| 13:01.24 | Samot | When you SUBSCRIBE a NOTIFY is sent to give you an update |
| 13:01.37 | Samot | How are you deleting the voicemails? |
| 13:01.46 | TheGallopingFox | through the menu 7 |
| 13:01.50 | Samot | OK |
| 13:02.18 | TheGallopingFox | i left a new voicemail at: 13:56, LED not flashing yet |
| 13:02.26 | TheGallopingFox | im timing how long it takes |
| 13:02.45 | Samot | Uhm. |
| 13:02.58 | Samot | WTF are you not looking at SIP DEBUGS? |
| 13:03.06 | TheGallopingFox | ok will do! |
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| 13:43.00 | jamesaxl | hi |
| 13:43.45 | jamesaxl | is it possible to call angent from asterisk command line and when he/she hang up, he/she will listen a playback? |
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| 13:52.39 | fauxalliance | jamesaxl: the console driver maybe and the record function |
| 13:53.26 | fauxalliance | then a callfile with the playback fucntion |
| 13:54.16 | fauxalliance | my thinking is dated... |
| 13:54.22 | fauxalliance | i'm sure there are other options |
| 13:57.24 | Samot | Well they are two different things. |
| 13:57.32 | Samot | Calling the agent is an Originate command |
| 13:57.51 | Samot | Having the agent listen to a playback after the call is hungup is a Dial() option. |
| 13:57.56 | fauxalliance | having the pbx call the agent and replay the recording |
| 13:58.12 | Samot | Calling the agent is an Originate command |
| 13:58.14 | Samot | Having the agent listen to a playback after the call is hungup is a Dial() option. |
| 13:58.40 | fauxalliance | ^^what he said |
| 13:59.20 | Samot | Dial() has options to let either/and caller and callee continue in the dialplan |
| 13:59.41 | Samot | Or send the caller to a specific context or the callee to a specific context after the call is hung up. |
| 14:02.22 | jamesaxl | Samot: could I do it using AMI command ? |
| 14:02.32 | jamesaxl | s/command/commands |
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| 14:09.00 | Samot | Yes. |
| 14:09.02 | Samot | Originate. |
| 14:09.32 | Samot | Which will fire off a diaplan context that has your Dial() string in it. |
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| 14:17.29 | jamesaxl | Samot: thank you very much |
| 14:22.40 | TheGallopingFox | at last got voicemail working! |
| 14:22.59 | TheGallopingFox | i was missing @default in sip.conf mailbox settings |
| 14:23.31 | TheGallopingFox | led shows straight away when new message arrives and it goes as soon as i delete the message |
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| 14:25.12 | TheGallopingFox | i also added: subscribemwi=yes not sure if that helped too |
| 14:28.27 | TheGallopingFox | i also added the voicemail number to polycom mwi subscribe |
| 14:29.02 | Samot | So basically... |
| 14:29.12 | Samot | Not reading any of the samples to figure out what to setup |
| 14:29.32 | TheGallopingFox | google helped for some of it |
| 14:29.38 | TheGallopingFox | but it works now how it should |
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| 14:30.48 | TheGallopingFox | so the last thing to do is create new samples for a IVR menu using asterisk extra core sounds, ill save that for another day... |
| 14:31.21 | TheGallopingFox | then i can get my call centre up and running |
| 14:31.29 | TheGallopingFox | (only joking..) |
| 14:32.50 | TheGallopingFox | ""Claim Your Mis-Sold PPI - Beat The PPI Deadline" |
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| 18:36.17 | TheGallopingFox | i only have g.722 asterisk sounds, however, when i call from outside, the sounds still work? |
| 18:36.30 | TheGallopingFox | i thought g.722 would only work internally |
| 18:36.52 | Samot | You're transcoding then. |
| 18:37.03 | TheGallopingFox | oh so asterisk does do transcoding |
| 18:37.05 | TheGallopingFox | i didnt know what |
| 18:37.16 | Samot | Of course it does |
| 18:37.21 | Samot | It's a B2BUA |
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| 18:37.26 | Samot | That's a key component |
| 18:37.27 | TheGallopingFox | would it be better to also place some alaw sounds inside it too so asterisk doesnt need to transcode it |
| 18:37.46 | TheGallopingFox | and that might improve quality? |
| 18:37.48 | Samot | Do you really need someone to say yes? |
| 18:37.56 | TheGallopingFox | ok, ill take that as a yes :D |
| 18:38.13 | TheGallopingFox | first i need to double check what codec is used from outside with sip debug |
| 18:38.22 | Samot | No. |
| 18:38.25 | Samot | You do not |
| 18:38.33 | Samot | What do you have set on your provider trunk? |
| 18:38.44 | TheGallopingFox | in iax.conf ? |
| 18:38.53 | Samot | Oh FFS. |
| 18:38.58 | Samot | Your provider supports IAX?! |
| 18:39.02 | TheGallopingFox | yes |
| 18:39.08 | TheGallopingFox | thats what i use for outside line |
| 18:39.18 | Samot | Jesus. |
| 18:39.31 | Samot | It would be in the peer settings in iax.conf |
| 18:39.37 | TheGallopingFox | ok let me check |
| 18:39.39 | Samot | You still need to tell it what codec to use. |
| 18:39.47 | Samot | And they should tell you what codec's they support |
| 18:39.50 | Samot | They did right? |
| 18:40.41 | TheGallopingFox | there isnt much information about that on their website |
| 18:40.49 | TheGallopingFox | in iax.conf i use this: disallow=all / allow=g722 / allow=alaw / allow=ulaw / allow=gsm |
| 18:40.56 | TheGallopingFox | but im pretty sure they dont support g722 |
| 18:41.02 | Samot | So take it out of there |
| 18:41.06 | TheGallopingFox | ok |
| 18:41.16 | Samot | which means they are doing on of the three. |
| 18:41.24 | TheGallopingFox | yeah but what one? |
| 18:41.33 | Samot | a/ulaw are basically the same |
| 18:41.47 | TheGallopingFox | alaw is used in Europe where im from |
| 18:41.48 | Samot | You could ask them. |
| 18:41.51 | TheGallopingFox | ok will check |
| 18:41.58 | Samot | So then take ulaw out of there |
| 18:42.09 | Samot | Don't offer codecs the other side doesn't support |
| 18:42.17 | TheGallopingFox | gsm isnt a codec, im not sure what that means? |
| 18:42.23 | Samot | Yes. |
| 18:42.26 | Samot | Yes it is a codec. |
| 18:42.29 | TheGallopingFox | oh it is |
| 18:42.50 | Samot | Is this for a company PBX? |
| 18:44.00 | Samot | I'm just curious... |
| 18:44.21 | TheGallopingFox | voiptalk |
| 18:44.21 | Samot | Because it's sounding like something more pointy/clicky GUIish would have been the better choice. |
| 18:44.30 | TheGallopingFox | https://www.voiptalk.org/products/index.html |
| 18:44.36 | TheGallopingFox | i cant find anything about codecs on that site |
| 18:45.30 | Samot | JFC. |
| 18:45.44 | Samot | They are a VoIP Provider |
| 18:45.50 | Samot | Not an ITSP. |
| 18:46.14 | TheGallopingFox | is that not a good thing? |
| 18:46.40 | Samot | Oh wait..they have instructions |
| 18:48.21 | TheGallopingFox | maybe should swap to a proper ITSP at some point |
| 18:49.25 | Samot | Well... |
| 18:49.45 | Samot | Considering it appears that not only are they an Asterisk shop but they offer Asterisk training... |
| 18:50.08 | Samot | And their Asterisk config samples are both horrible and from Asterisk 1.6.x |
| 18:50.32 | TheGallopingFox | yeah |
| 18:50.43 | Samot | You would figure an Asterisk shop, offering Asterisk training would have not only current but, I don't know, more detailed samples. |
| 18:51.49 | Samot | They are basically the European version of a client I'm about to drop. |
| 18:52.06 | Samot | A lot of money backing Asterisk knowledge that stopped pre-1.8 era |
| 18:52.40 | Samot | And a refusal to modernize in anyway because it's "beyond him" and he's a "hands on" guy. |
| 18:55.48 | TheGallopingFox | right done a debug from a outside call |
| 18:55.50 | TheGallopingFox | requested format = alaw |
| 18:56.21 | TheGallopingFox | so will now download the alaw samples |
| 18:57.26 | TheGallopingFox | i think i should still allow ulaw just incase i have a international call? |
| 18:57.45 | Samot | ulaw is a standard PSTN codec. |
| 18:58.03 | Samot | It's why they are only sending that instead of g722 or another codec. |
| 18:58.10 | Samot | So *they* don't have to transcode |
| 18:58.31 | TheGallopingFox | oh ok |
| 18:58.45 | Samot | You, no matter what, will always be transcoding calls to/from the PSTN |
| 18:59.01 | Samot | Because your trunk will be alaw and your devices are g722 |
| 18:59.10 | Samot | Immediate transcoding |
| 18:59.41 | Samot | Unless you make sure alwa is part of the offering for the devices and on the phone so it can be chosen for external calls. |
| 18:59.50 | Samot | *alaw |
| 19:00.37 | Samot | So you can have it that when Device to Device it's g722 |
| 19:00.59 | Samot | And when it is Device <--> PSTN it can use alaw |
| 19:01.42 | TheGallopingFox | yes thats what im changing |
| 19:01.48 | TheGallopingFox | g722 works internally |
| 19:01.51 | TheGallopingFox | nice quality |
| 19:04.23 | TheGallopingFox | works |
| 19:04.25 | TheGallopingFox | <PROTECTED> |
| 19:04.41 | TheGallopingFox | now try internal |
| 19:05.38 | TheGallopingFox | Playing 'vm-intro.g722' (language 'en') |
| 19:05.41 | TheGallopingFox | perfect |
| 19:05.56 | TheGallopingFox | so all i need is g722 and alaw |
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| 20:45.33 | fauxalliance | ...To Be Continued... |
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| 22:29.05 | sibiria | how can i, in a dialplan during an incoming call, pick up the IP number of the caller? |
| 22:29.09 | sibiria | using pjsip |
| 22:29.48 | sibiria | CHANNEL(recvip) and peerip are "Unknown or unavaiable" |
| 22:45.37 | sibiria | CHANNEL(pjsip,remote_addr) was it |
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