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| 00:59.16 | snadge | blocking a call based on callerid.. is that something i do in extensions.conf ? |
| 00:59.30 | snadge | i know how to block numbers based on the dialled number |
| 01:00.05 | WIMPy | yes |
| 01:00.42 | WIMPy | You can do it as part of the extension, seperating it with an "/" or using function CALLERID. |
| 01:01.06 | snadge | the dialplan is huge though.. so you could check the callerid before every single DIAL |
| 01:01.28 | snadge | or maybe have a function to do that.. but that seems like an inefficient way to do that |
| 01:02.20 | WIMPy | Creating an efficient dialplan can take som thought. |
| 01:02.25 | WIMPy | +e |
| 01:03.25 | snadge | i found an _X .. so maybe i can use that.. i didn't write this monstrosity.. inherited it |
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| 01:42.41 | clever | i'm trying to debug a google-talk connection within asterisk, any attempt to call out results in the error "Unable to create Jingle session on endpoint 'myendpoint' as no capable resource for target '....'" |
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| 02:28.33 | lvlinux | has application originate been changed in v13?? the module is loaded but when I try to run it it says there isn't any such command |
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| 02:38.00 | lvlinux | how can I diagnose the problem there? |
| 02:43.41 | lvlinux | evidently * isn't reading my cli_aliases file for some reason... |
| 02:51.33 | lvlinux | And the reason was that someone blacklisted the res_clialiases.so module... :-( |
| 02:56.50 | wyoung | hi gang! |
| 02:56.55 | wyoung | snadge: yes |
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| 04:15.44 | Lope | is there any good reason that I should enable g722? I've got alaw, ulaw and gsm enabled. My MOH is flawless with those codecs. But it seems nomatter what I do, g722 is always choppy, poppy, blippy and crappy when playing my MOH |
| 04:16.34 | Lope | I've coded my MOH into g722 and confirmed asterisk is serving the g722 file. I've played the file on my PC, and it's flawless. It's somehow the transmission, or the implementation of g722 that sucks on the softphones I've tested. |
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| 05:14.21 | Lope | what sample rate is used by the asterisk speex codec? |
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| 05:52.02 | Lope | okay I see it supports speex (8) and speex16 and speex32. |
| 05:52.12 | Lope | Now i just wanna find out what extension to save my speex files in. |
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| 06:05.29 | Lope | I've got a sound file with a .spx extension, yet asterisk doesn't use it when my caller connects with the speex codec? I've also tried .speex ... so far I've not had any problems with .alaw .ulaw .g722 .gsm etc |
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| 06:20.52 | Lope | `core show file formats` speex is missing? but I have the speex module loaded? |
| 06:20.59 | Lope | the codec module |
| 06:28.20 | cmendes0101 | Lope: You have speex installed on that machine? |
| 06:28.54 | Lope | yes |
| 06:29.25 | Lope | I also loaded func_speex.so incase it would help, but it has not. |
| 06:31.37 | Lope | wikipedia and another random site said the speex file extension is .spx :/ |
| 06:31.38 | cmendes0101 | Is the call using speex as the select codec? If the call falls to ulaw then it would be trying to convert and play ulaw I believe |
| 06:32.06 | cmendes0101 | i mean it would grab the best format first |
| 06:32.07 | Lope | Yes, the call is being made with the speex codec, it says so on my softphone, and I've disabled all other codecs in sip.conf. |
| 06:32.17 | Lope | allow=!all,speex |
| 06:32.23 | Lope | that's how I test diff codecs |
| 06:33.10 | Lope | `core show file formats` why is speex is missing, I have the speex codec loaded? |
| 06:35.57 | cmendes0101 | It just errors and says audio file not found? |
| 06:36.24 | cmendes0101 | theres 3 version of speex but would have different error mentioning that |
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| 06:45.35 | Lope | No error is shown at all. When I make the call it says Playing 'my-moh.alaw' (language 'my-lang') |
| 06:46.26 | Lope | When I connect the call as .g722 for example, it will say "Playing 'my-moh.g722' (language 'my-lang')" for gsm it will say "Playing 'my-moh.gsm' (language 'my-lang')" |
| 06:46.52 | Lope | I've got debug and verbose 10 in the asterisk terminal |
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| 06:51.19 | Lope | Now I removed all other formats. and I left 4 copies of the file in with these extensions: .spx .spx8 .speex .speex8 and it gave an error: ast_openstream_full: File my-moh does not exist in any format. app_playback.c:494 playback_exec: Playback failed on SIP/testsip-00000003 for my-moh |
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| 06:52.15 | cmendes0101 | The extension I've seen is .spx. I would say check to make sure the codec_speex was compiled but if your call is connected in speex that should be good |
| 06:52.54 | Lope | `core show codecs` speex 8, 16 and 32khz is there |
| 06:53.19 | Lope | yeah, I mean when I have the alaw MOH file there, asterisk transcodes it to speex. |
| 06:53.25 | Lope | I just wanted to provide it in speex format. |
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| 06:59.29 | Lope | I guess there's no file format for speex? https://wiki.asterisk.org/wiki/display/AST/File+Format+Drivers |
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| 07:05.07 | cmendes0101 | oh interesting |
| 07:07.05 | Lope | very weird :/ |
| 07:08.39 | cmendes0101 | I know this might be pointing in a different direction but the ispeech plugin uses speex. Maybe digging through will help find something for what you need. Of course it be speex is on the ASR side which won't be anything useful |
| 07:08.50 | cmendes0101 | https://github.com/zaf/asterisk-ispeech |
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| 08:11.47 | seik0 | Hi! I am concerned about many (about 20 in 10 minutes with ~45 simultaneous sip calls) "__sip_autodestruct: Autodestruct on dialog ...". What can I try to debug the issue? |
| 08:12.08 | seik0 | Because I think that it causes sip to lag |
| 08:12.32 | seik0 | (Another possibility is that something that causes sip to lag causes autodestruct) |
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| 08:24.16 | seik0 | (I mean, something causes sip to lag, that causes to some misbehave in sip dialogs and that causes dialogs to hang and thus to autodestruct) |
| 08:28.19 | seik0 | btw, all dialogs are in BYE, like: WARNING[1868]: chan_sip.c:4057 __sip_autodestruct: Autodestruct on dialog '3076ce734523dfb613fd235f208ec8ddb9a58e16@123.456.789.101' with owner SIP/sip_perm_in-000003da in place (Method: BYE). Rescheduling destruction for |
| 08:28.19 | seik0 | <PROTECTED> |
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| 09:37.56 | Zogot | is it correct, that a Dial(peer,u(on_answer)), that the u(on_answer) from that Dial happens to the channel that ran PickupChan? even though PickupChan didnt (and cant) define u(on_answer) |
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| 10:46.28 | jkroon | hi all, does anybody know of a way to use asterisk as a SBC? Ie, when a client registers to us, perform a forward REGISTER, and when a client de-REGISTERs do the same towards upstream? unfortunately using the regcontext in combination with DUNDi is not a viable option in this case. |
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| 12:31.25 | geek-man | Hi All. I'm looking for some advice running SIP realtime. Using Asterisk 11 latest against MySQL/Maria DB Galera using ODBC. We're getting a lot of segfaults when running in prod, with about 150+ simultaenous registrations -- segfaults are ranging from libc, to asterisk to chan_sip. They can vary from daily to once a month, so I doubt just submitting a bunch of bug reports is going to be a |
| 12:31.25 | geek-man | 100% fix at this point. |
| 12:31.40 | geek-man | I'm confident the segfaults are due to the high number of registrations. We have 2x PBX nodes. The primary node segfaults a lot. The secondary node with < 10 registrations has not segfaulted at all. Also, both PBX node has its own sipregs table, which I've read is buggy. But I can't figure out how to avoid this, if both nodes are on the same sipregs table each node mistakenly thinks all |
| 12:31.40 | geek-man | devices are local to it (Using IsChanAvail) and I cannot route calls between PBXes properly. |
| 12:31.53 | geek-man | t this point, I'm considering A) Disabling sipregs entirely, with registrations stored on the local astdb ONLY. _But_ still keep sipusers in realtime, somehow (this is important). Or, B) Figuring out a way to put all sipregs under sipusers (which seems recommended for stability) but avoid issues with IsChanAvail somehow -- using a view? |
| 12:31.58 | geek-man | Both solutions, I'm not sure how to achieve. Any advice on this, or any other suggestions, would be appreciated. |
| 12:37.16 | catphish_ | geek-man: i use chan_sip and realtime mysql, i upgraded from 1.8 to 13 towards the end of last year, hen downgraded again because of ongoing segfaults :( |
| 12:37.28 | jkroon | geek-man, I just did benchmarks with 60k registered clients in the week, no problems. |
| 12:37.47 | jkroon | however, i'm not using sipregs, only sippeers |
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| 12:38.02 | catphish_ | i dont think the number of clients is important really, but there are (or were) definitely issues |
| 12:38.14 | jkroon | not arguing, just sharing my experience. |
| 12:38.35 | catphish_ | i'm planning to move away from chan_sip and realtime and try asterisk 13 with pjsip and config files for peers |
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| 12:41.00 | jkroon | that's probably longer term what I need to do as well, that should get me 500k SIP users/server hopefully. which would help me a lot. |
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| 12:41.48 | catphish_ | i'm busy writing an ARI backend for most of my routing / logging / basically everything |
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| 13:20.02 | danielyk | It is able to set max_retries of an pjsip Trunk to infinity? |
| 13:22.53 | newtonr | If you can, you probably set it to 0 to accomplish that |
| 13:27.08 | danielyk | @newtonr @file has written at https://community.asterisk.org/t/pjsip-outbound-registration-max-retries/49060 that it is not currently possible. I think there has nothing done yet... |
| 13:28.07 | newtonr | danielyk, sounds like an answer. :) |
| 13:28.16 | file | knows nothing |
| 13:28.50 | danielyk | I will write a feature request |
| 13:32.01 | newtonr | danielyk, feature requests are not accepted on the bug tracker without a patch. So write a patch with it! :) |
| 13:33.11 | danielyk | @newtonr thanks for this info, i will try it. |
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| 15:09.48 | Starstorm | Hiya. I inherited a 1.8 installation. The console is reporting what looks to be an error, and whenever it pops up the fop2 blinks offline. The 2 lines are: DEBUG[31756]: sig_pri.c:1016 sig_pri_request: sig_pri_request 18 AND sig_pri.c:6591 sig_pri_call: CALLER NAME: NUM: [MainPhonelineNumber] Would anyone know what to do or could tell me where I could find a fix ? |
| 15:10.10 | Lope | jkroon: have you tried compiling with intel's compiler? |
| 15:11.43 | jkroon | Lope, no. |
| 15:11.46 | Lope | hey guys i struggled for about 2 hours with codecs and formats. first I discovered, it seems that while asterisk 13 supports encoding/decoding speex, it doesn't support speex as a file-format. |
| 15:12.20 | Lope | which means my * box must transcode all my audio for every call that uses speex. |
| 15:14.25 | Lope | second issue, I've had a lot of trouble trying to convert wav to ilbc. it's not natively supported by sox. it's theoretically possible to compile avconv (libav) with external libilbc library, but it won't compile, code is broken, I've tried various branches of each codebase, based on changelogs etc. |
| 15:16.01 | Lope | Then I thought maybe I can use asterisk to convert wav to ilbc. I dug up an old voip info page, that said you can run this in asterisk cli `file convert /tmp/foo.wav /tmp/foo.ilbc` but `file convert` is not a recognized command in asterisk 13. |
| 15:18.23 | [TK]D-Fender | check that the sytax hasn't changed |
| 15:18.28 | [TK]D-Fender | CLI has help..... |
| 15:19.28 | Lope | Hi TKD: I tried `core show help file` `core show help file convert` |
| 15:19.51 | [TK]D-Fender | start with TAB |
| 15:20.48 | [TK]D-Fender | that does look right... |
| 15:20.53 | [TK]D-Fender | start by showing us the attempt |
| 15:21.00 | Lope | oh wow, I didn't know about tab |
| 15:21.19 | Lope | okay, nothing jumping out at me there, 1 sec. |
| 15:21.30 | [TK]D-Fender | file <tab> |
| 15:22.15 | Lope | ! acl aoc bridge cc cdr cel channel cli config core data database devstate dialplan dnsmgr features group http indication keys local logger manager mixmonitor module moh no queue rtcp rtp say sip stun timing udptl voicemail xmldoc |
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| 15:22.28 | Lope | I don't have file |
| 15:22.38 | file | sometimes I don't even have myself |
| 15:23.29 | Lope | *CLI> core show help file .. No such command 'file'. |
| 15:25.12 | Lope | asterisk 13.6 |
| 15:25.47 | Lope | I had another look inside menuselect (the build tool) I don't see any options for "file" stuff? |
| 15:28.26 | jkroon | Lope, asterisk -rx "help" | grep file ?? |
| 15:31.37 | Lope | jkroon: it should be: `asterisk -rx "core show help" | grep file` but there's nothing there that starts with file or does conversion stuff. |
| 15:32.23 | jkroon | interesting. |
| 15:32.29 | Lope | I also grep'd for conv and trans |
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| 15:34.12 | Lope | The original hack idea I had was to use mixmonitor to record the call in ilbc format with separate a/b streams, then call my * box from a softphone with sound muted (depending if the streams are really recorded separate) and wait for the whole moh track to play out, then use the recording. |
| 15:34.56 | Lope | but I'd much prefer a cleaner solution? |
| 15:35.12 | shido6_ | log |
| 15:35.20 | shido6_ | history |
| 15:35.37 | Lope | shido6_: can you please elaborate> |
| 15:37.47 | Lope | TKD: you know asterisk backwards, any ideas regarding `file convert` in asterisk 13? or what versions support file convert? |
| 15:38.12 | file | "file convert" is in the res_convert module, if that's not built or loaded then it won't be present |
| 15:39.29 | Lope | file: genius you are. I have the .so. will load it. |
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| 15:41.14 | shido6 | Lope - I was looking for retroactive submissions to better understand your query |
| 15:41.35 | shido6 | the first part I read was âThe original hack idea..." |
| 15:41.52 | Lope | Brilliant! it seems to have worked, or at least not complained and said it worked :) |
| 15:42.15 | Lope | shido6: okay |
| 15:44.59 | Lope | brilliant, it's working! |
| 15:45.22 | Lope | thank you file! classic that that's your nick. |
| 15:45.34 | file | it's many things |
| 15:45.42 | file | classic may be a word to describe it |
| 15:46.55 | shido6 | its got some years on it , definitely. |
| 15:47.33 | file | Registered : Oct 29 19:18:09 2001 (14y 25w 2d ago) |
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| 18:25.45 | lvlinux | What's the difference between Dial(SIP/provider/number) and Dial(SIP/number@provider)? |
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| 18:34.12 | newtonr | IIRC the first is used a peer defined in sip.conf the second is using an explicit SIP URI to dial an endpoint not specified in sip.conf |
| 18:34.39 | newtonr | for PJSIP I wrote a page a while back |
| 18:34.39 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels |
| 18:34.54 | newtonr | but PJSIP does not equal chan_sip |
| 18:35.20 | lvlinux | newtonr: thanks! I'm actually using PJSIP. |
| 18:35.45 | newtonr | Good! |
| 18:37.09 | lvlinux | Yeah, I decided to force myself to take the plunge on this new system I'm setting up :-) |
| 18:37.44 | lvlinux | Well "plunge" is probably over-reacting lol. So far I like it, and the extra config actually seems to make better sense to me than sip.conf |
| 18:43.51 | lvlinux | For sip providers that have multiple POPs to send me calls, with chan_sip, I had to create multiple sip peers. With PJSIP, can I just create one endpoint with multiple AORs?? Or do I need to create multiple endpoints? |
| 18:45.18 | file | if you are doing IP based matching you need one identify section |
| 18:45.28 | file | but you can specify as many matches as you want, and they'll all go to the single endpoint |
| 18:45.56 | file | if it's an outbound registration you can try line support which will try to automatically associate incoming calls with an endpoint without further configuration |
| 18:46.24 | lvlinux | It's IP based auth. |
| 18:46.52 | file | then a single identify with as many match as you need |
| 18:47.01 | lvlinux | SAWEEEET! thanks |
| 18:47.30 | lvlinux | the aor is for outbound, and the identify is inbound, correct? |
| 18:48.38 | file | yes |
| 18:48.55 | file | and just as an fyi - https://www.asterisk-blog.com/2016/01/27/the-pjsip-outbound-registration-line-option/ there's a blog post on the line option |
| 18:51.35 | lvlinux | Ok, thanks I will definitely give that a read. Now, on the contact item in the AOR, and match in identify, can I put an FQDN, or only an IP address? |
| 18:52.13 | file | you can put a FQDN or IP address |
| 18:52.28 | lvlinux | great |
| 18:52.29 | file | in the case of an identify it wil get all A records, but it won't do SRV or NAPTR |
| 18:53.05 | lvlinux | ok. Is SRV coming or is there a reason it won't do that? |
| 18:54.15 | file | noone is currently working on it |
| 18:55.04 | lvlinux | Well what I meant was is there a particular reason that and SRV or NAPTR shouldn't be used for that. |
| 18:55.18 | lvlinux | And by your answer I'm assuming no :-) |
| 18:55.33 | file | it could be in 14 |
| 18:55.44 | file | 13 not easily |
| 18:56.48 | lvlinux | K. |
| 18:57.15 | lvlinux | thanks file! |
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| 21:03.12 | swimmercol | Hi folk |
| 21:03.19 | swimmercol | I have lot of retransmission timeout reached on the CLI |
| 21:04.18 | swimmercol | what I understand is...it happens when Asterisk send a re-invite to an endpoint o provider by SIP and didn't get an answer |
| 21:05.00 | swimmercol | I wonder why since the re-invite is being sent and Asterisk is not getting the confirmation the call still stablished |
| 21:05.02 | swimmercol | thanks ! |
| 21:05.29 | lvlinux | network issues maybe? have you tested for packet loss? |
| 21:05.51 | swimmercol | lvlinux: sure..NAT issues |
| 21:05.58 | swimmercol | but the call is never dropped |
| 21:06.16 | swimmercol | my question is trying to understand why the call is still up |
| 21:06.46 | lvlinux | well if Asterisk doesn't get the confirmation that the call is over, it won't kill it. |
| 21:06.57 | lvlinux | For a while anyway. |
| 21:07.19 | swimmercol | is there a timer for that? |
| 21:07.40 | lvlinux | I'm sure there is somewhere, but I'm not familiar with it. Probably in your chan_sip or pjsip config. |
| 21:07.51 | swimmercol | I think is Asterisk detects that there is not RTP traffic it kills that session |
| 21:08.50 | lvlinux | Depends on if Asterisk is processing the RTP itself I guess. |
| 21:09.02 | lvlinux | If not, then it wouldn't know about the RTP. |
| 21:09.14 | swimmercol | that's right |
| 21:09.36 | swimmercol | regarding my question...what SIP RFC says about those re-invite |
| 21:09.37 | swimmercol | hehehe |
| 21:09.49 | swimmercol | seems like I have to read that RFC lol |
| 21:10.36 | lvlinux | lol it probably wouldn't hurt :-) It's 3261 i think |
| 21:10.57 | swimmercol | it is |
| 21:11.02 | swimmercol | :) ha |
| 21:11.29 | lvlinux | "the failure of the re-INVITE does not cause the existing call to fail - the session continues using the previously negotiated characteristics. |
| 21:11.48 | lvlinux | that's from page 15 |
| 21:13.25 | swimmercol | that's exactly what I wanted to "hear" |
| 21:13.30 | swimmercol | thanks sir ! |
| 21:13.35 | lvlinux | :-) |
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| 21:32.16 | polysics | hey! long time no see, #sterisk :) |
| 21:32.36 | polysics | question for y'all: is it possible to effectively spoof a phone number using Caller ID? |
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| 21:35.06 | swimmercol | polysics: what u mean? |
| 21:35.41 | polysics | swimmercol: a company starts using our software but still wants to call out using their existing PSTN number which they cannot port |
| 21:38.35 | swimmercol | okay so.. |
| 21:39.25 | swimmercol | you can on Asterisk but......our provider (the one you use to call put) MUST allow you to set the CallerID you want otherwise they are going to rewrite whatever CallerID you use on Asterisk |
| 21:39.40 | swimmercol | *call out |
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| 21:41.04 | lvlinux | polysics: yes it's very simple to do but your provider has to allow it. It's legal in the US as long as it's not done with intent to deceive. |
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| 21:41.18 | lvlinux | Anveo Direct, voip.ms both support it. |
| 21:41.27 | lvlinux | and didlogic, if you request it. |
| 21:41.27 | polysics | do you know of any US providers that would allow me to do that? Flowroute or Voxbone maybe? |
| 21:41.47 | lvlinux | I think flowroute will if you proove to them that you own the other number. |
| 21:41.57 | polysics | yeah, there would be no intent to deceive, the same business owns the numbers |
| 21:42.26 | lvlinux | then should be no problem, as long as the voip account is in their name (with Flowroute). |
| 21:42.26 | swimmercol | vitelity also does |
| 21:42.30 | polysics | lvlinux: you seem to be well versed in providers :) who would you recommend for getting about 300 trunks? |
| 21:43.31 | lvlinux | depends on several factors, like usage, CID support, outbound CNAM, price, etc. |
| 21:43.44 | lvlinux | if you need outbound CNAM, you get limited real fast. |
| 21:43.57 | lvlinux | as not many providers support it. |
| 21:44.15 | swimmercol | that's right |
| 21:44.47 | lvlinux | Any by "trunks" do you mean "channels" or separate trunks (to be used individually by clients or whatnot)? |
| 21:45.23 | lvlinux | In other words, do you need 300 channels, all used by one place, or different trunks, with differnt auth/ip endpoints, etc. |
| 21:45.23 | polysics | each number will be assigned to a different office and call out only 1 to 2 customers at a time |
| 21:45.36 | polysics | all used by one Asterisk box |
| 21:45.43 | polysics | possibly 2 or 3 but you get the idea |
| 21:45.57 | polysics | most calls will be outbound, I plan to just dial the old number on inbound |
| 21:46.09 | lvlinux | k then really you only need 300 channels and one trunk (or 2 or 3) |
| 21:46.38 | polysics | and I need Outbound CNAM, I would say |
| 21:47.15 | lvlinux | qualify "I would say" because it could force you to pay around double the per-minute price :-) |
| 21:47.37 | lvlinux | I pay $0.004/min for Anveo. |
| 21:47.49 | lvlinux | Anveo Direct that is. |
| 21:47.50 | polysics | or whatever it is between Outbound and Inbound CNAM that allows me to "Spoof" the old number |
| 21:47.55 | polysics | I need that. |
| 21:48.40 | lvlinux | That's just callerid setability. Outbound CNAM is where the called party sees the number AND your business name, not just the number. |
| 21:49.27 | lvlinux | But actually, if you are setting the callerid to a number that is on another provider (which may have CNAM already) then you don't need it, as that is handled by the provider that has the DID you will spoof. |
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| 21:50.00 | polysics | so I just need caller ID set-ability, right? |
| 21:50.06 | lvlinux | yup |
| 21:50.26 | polysics | I had good experiences with Flowroute and Vitelity so far |
| 21:50.34 | polysics | any recommendations? |
| 21:50.44 | lvlinux | Yes they are both excellent. |
| 21:50.56 | lvlinux | I use them, but also LOVE Didlogic and Anveo direct |
| 21:51.13 | lvlinux | and voip.ms is good too, but I haven't used them a lot. |
| 21:51.24 | polysics | so any of the "big names" |
| 21:51.42 | lvlinux | IMHO you can't go wrong with any of those. |
| 21:51.45 | polysics | I curiously yet have to find anyone who has worked with Voxbone, even if they are big |
| 21:52.05 | lvlinux | That's because they are so big you have to have a spending commitment with them. |
| 21:53.04 | polysics | oh, I see |
| 21:53.05 | lvlinux | Anveo Direct is my main service that I use. dirt cheap per minute rates. No problems. They just work great. BUT support is limited, and you pay extra for it. |
| 21:55.02 | lvlinux | DIDlogic is similar, but they do have offer support. $0.007 for them. |
| 21:55.37 | lvlinux | Flowroute and voip.ms are $0.01 or so, Vitelity $0.014 |
| 21:55.51 | lvlinux | Support gets better as you pay more for the minutes. |
| 21:55.53 | polysics | with some luck there should not be too much need for support |
| 21:56.13 | polysics | Flowroute support is very good |
| 21:56.19 | lvlinux | I've never needed it with Anveo. As long as you know what you are doing and set things up correctly, most of the time stuff just runs. |
| 21:56.27 | lvlinux | Yes Flowroute is great. |
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| 21:56.37 | polysics | and this is a very simple setup, with one box and one trunk |
| 21:57.21 | polysics | how are Value routes on Anveo? |
| 21:57.38 | polysics | I am usually wary of anything called "value" :P |
| 21:57.47 | pvoigt | I am having my asterisk configured for two providers. One is used for all outgoing calls while the second one is used only if I use a special prefix, e.g. "*1" in my case. Disadvantage is that I cannot use my phonebook stored on the sip phone. I am looking for more flexible way easy way to switch the provider. I would appreciate any help on this. |
| 21:58.52 | lvlinux | polysics: I'm not a big usage guy, but I haven't noticed any issues at all with quality or connections, and I have my setup to use the cheapest routes first. |
| 21:58.58 | WIMPy | Use a 2nd account (AKA "line") |
| 21:59.36 | lvlinux | pvoigt: what WIMPy says is probably the easiest and simplest way. |
| 21:59.50 | polysics | lvlinux: so, on Anveo, I would get normal per minute DIDs (no need for free inbound as ideally no one should be calling those) and set outgoing up as Value? |
| 22:00.36 | polysics | 0.19 per month per number is CHEAP. |
| 22:01.01 | lvlinux | You don't even have to get a DID. But they are cheap. You can set it up to use LCR or Premium first or however you want. |
| 22:01.09 | pvoigt | WIMPy, lvlinux: Thanks, I am rather new to Asterisk and do not fully understand, because I HAVE already configured both providers. Could you please explain what to you mean with "2nd account"? |
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| 22:01.39 | WIMPy | ... on the phone. |
| 22:01.44 | polysics | if you do not get a DID, you just spoof the Caller ID and that's it? |
| 22:01.57 | lvlinux | yup |
| 22:02.11 | polysics | wow, I think I am going to try them first :D |
| 22:02.32 | WIMPy | Create two accounts for the phone and register bot so that you can select the accoutn on the phone. Then put them in defferent contexts dialling different providers. |
| 22:02.39 | pvoigt | OK, get this point, but... |
| 22:02.41 | polysics | it's either Anveo for super prices or Flowroute because I know they kick butt in support |
| 22:03.09 | polysics | lvlinux: thanks, you told me all that I needed to know. Really, thanks! |
| 22:03.25 | pvoigt | WIMPy, lvlinux: Ah, now its clear. Thanks for this advice. Will think about it in more detail. |
| 22:03.40 | lvlinux | polysics: np let us know how it comes along :-) |
| 22:03.56 | polysics | I'll whip up a box tomorrow monrning and go to town :) |
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| 22:13.01 | pvoigt | I have configured Voicemail. Currently a caller gets information about the reciving interal extension via voice prompt. But I would prefer the caller to re-tell the number he has dialed instead of the internal extension. Is there a way to configure voicemail this way? |
| 22:14.01 | pvoigt | I should add that I would like to avoid recording a custom voice prompt. |
| 22:14.25 | WIMPy | He actually is played the number (or rather name) of the mailbox he has reached. |
| 22:14.57 | pvoigt | WIMPy: Yeah, can this be changed easily? |
| 22:14.58 | WIMPy | That need not be the dialled number not the extension. |
| 22:16.34 | WIMPy | You chose the names. They can be whatever you like. |
| 22:16.51 | WIMPy | They don't even have to be numeric. |
| 22:17.08 | WIMPy | But I wouldn't expect the playback to work on anything else. |
| 22:19.59 | pvoigt | OK, but it is probably not a good idea to name my mailbox with my phone number just to have a corresponding voice prompt. |
| 22:20.17 | WIMPy | Why not? |
| 22:21.53 | pvoigt | Because I have several phone numbers...but wait: You just recommende to set up a second line - and that will get an mailbox name which equals my second phone number and so on. |
| 22:22.15 | pvoigt | Should work this way. |
| 22:25.00 | WIMPy | Not that there's any realtion :-) |
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