| 00:07.41 | sofltech | TK: are you suggesting something like this? http://pastebin.com/giCyHgjP |
| 00:11.57 | [TK]D-Fender | sofltech, http://pastebin.com/rSHZhAYi |
| 00:20.27 | sofltech | TK: thanks for that, I think I understand, the local channel that initiated the call is where the answered caller will get dumped and that's where I run my extended dial plan. however, if they hang up in there, aren't I in the same situation (if they didn't confirm) that I don't know the status? |
| 00:27.33 | [TK]D-Fender | no |
| 00:27.45 | [TK]D-Fender | there are TWO sets of diaplan |
| 00:27.53 | [TK]D-Fender | that is NOT where they get dumped. |
| 00:27.58 | [TK]D-Fender | that is the part that PLACES the call... |
| 00:28.05 | [TK]D-Fender | where they dumped is a SECOND place |
| 00:33.58 | sofltech | ok... I don't get and I'm not sure what I'm not understanding. I'm looking at your updated pastebin (thank you) and I don't see how I would restructure that to do what you're suggesting. |
| 00:40.19 | [TK]D-Fender | loca channel has 2 lines |
| 00:40.30 | [TK]D-Fender | the dial and the userevent saying that nothing answered |
| 00:40.53 | [TK]D-Fender | the OTHER part is the same as you used to have doing whatever you were doing before |
| 00:40.59 | [TK]D-Fender | plus a message saying "success |
| 00:41.32 | [TK]D-Fender | Stop thinking of this is 1 piece of dialplan. |
| 00:41.33 | [TK]D-Fender | IT ISN"T |
| 00:41.42 | [TK]D-Fender | You are doing an Originate |
| 00:41.49 | [TK]D-Fender | you call the CHANNEL: side. |
| 00:41.59 | [TK]D-Fender | when that answers it gets dumped into the dilaplan where you tell it |
| 00:42.08 | [TK]D-Fender | so your Channel: in this case will be MORE DIALPLAN |
| 00:42.16 | [TK]D-Fender | bu att that does is the actual DIAL |
| 00:42.36 | [TK]D-Fender | Because when that gets answered it gets bridged into the dialplan where the originate points to |
| 00:42.59 | [TK]D-Fender | that dialplan is just doing the dirty work instead of you saying "SIP/provider/number" |
| 00:43.09 | [TK]D-Fender | it has LOGIC around it to do your ADADITIONAL stuff |
| 00:43.14 | [TK]D-Fender | ADDITIONAL* |
| 00:43.23 | [TK]D-Fender | Which in this case is 1 line |
| 00:44.36 | sofltech | TK: thanks, digesting. |
| 00:46.02 | sofltech | TK: yes, I got it, thank you so very much. however, this brings me back to my original concern. |
| 00:46.41 | sofltech | Whent he call gets bridged, the originate local channel dialplan will execute (got it) |
| 00:47.02 | sofltech | That dialplan will have logic that requires the user to ack (press 1) |
| 00:47.32 | sofltech | If the user in fact acks, a userevent(success) is generated. |
| 00:48.37 | sofltech | If the user simply hangs up, no fail message? |
| 00:49.06 | sofltech | Unless a hangup will return them to the 2nd line after the DIAL command? |
| 00:55.03 | sofltech | I'll have a pastebin for you shortly, thanks |
| 00:56.51 | sofltech | TK: http://pastebin.com/ACpnFfVP is this on the right track? |
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| 01:02.57 | Katty | Boop. |
| 01:03.09 | Katty | Beepbeep |
| 01:03.11 | file | bop |
| 01:03.14 | Katty | Whrrrrr |
| 01:06.21 | superscrat | badabeep. |
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| 01:34.23 | Milos|Work | meow |
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| 01:51.02 | [TK]D-Fender | <sofltech> If the user simply hangs up, no fail message? <- on THAT end you'd deal with "h" to send a "end no answer" response |
| 02:03.38 | Katty | hey fenderbender. howre you dear |
| 02:08.14 | [TK]D-Fender | doing alright. Place is clean. I should have guests over more than every other year! ;) |
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| 02:49.36 | sofltech | TK: thanks for that info, going to try it your way, much appreciate it. |
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| 03:25.04 | sofltech | TK: getting error ... unable to call channel local ... no such extension/context (i do have it) - where do i look for this? |
| 03:25.23 | sofltech | TK: to resovle this issue - if you can point me in the right direction... thanks. |
| 03:26.49 | sofltech | basic code for extension... |
| 03:26.50 | sofltech | exten => alert,1,Noop(Alert notification) |
| 03:26.50 | sofltech | exten => alert,n,Playback(pressOneToAcknowledge) |
| 03:26.50 | sofltech | exten => alert,n,UserEvent(AlertStatus,Ack: confirmed) |
| 03:26.57 | AnonGirl | win 74 |
| 03:27.02 | AnonGirl | whups |
| 03:37.51 | sofltech | nm, figured this one out... |
| 03:40.42 | [TK]D-Fender | "you don't" is the answer. |
| 03:40.57 | [TK]D-Fender | When it says "not found".... it doe mean it |
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| 03:47.19 | sofltech | tk: thanks, yep, had an extra character in there. |
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| 07:04.47 | awk | http://pastebin.com/kVXJpNSG |
| 07:05.03 | awk | Please can somebody check that post... |
| 07:05.07 | awk | presencestate pri |
| 07:05.07 | awk | pbx01*CLI> pri show spans |
| 07:05.07 | awk | PRI span 1/0: Up, Active |
| 07:05.07 | awk | PRI span 2/0: In Alarm, Down, Active |
| 07:05.07 | awk | PRI span 3/0: In Alarm, Down, Active |
| 07:05.09 | awk | PRI span 4/0: In Alarm, Down, Active |
| 07:05.23 | awk | It just says the number is not answering, but the call isn't going out. |
| 07:16.11 | WIMPy | Apart from the fact that your caller ID isn't valid, it doesn't looke like you're talking to anything there. |
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| 07:36.25 | awk | Executing [s@macro-dialout-trunk:22] Dial("SIP/1420-00000035", "DAHDI/g1/1023,300,Tt") dialing 1023 via DAHDI g1 ? |
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| 07:46.07 | cyford | awk i dont know about pri, but i had a simular issue that turned out to be syntax errors causing trunks to fail without any good errors |
| 07:48.51 | r00f | you are not getting replies, all your pri messages are outgoing. it would be helpful to see debug from the other side, maybe asterisk thinks that messages are being sent but driver just drops them. or drops replies from other party (if any) |
| 07:49.35 | r00f | actually i hate pri. so many debug efforts without easy tracing tools, like tcpdump for sip |
| 07:57.25 | awk | thats, let me try re-create this |
| 07:57.30 | awk | thats/thanks |
| 08:04.01 | awk | one thing, if I put on a debug and I call in, I get "no information" |
| 08:04.09 | awk | could this mean there is a fault with the BRI |
| 08:04.21 | awk | why i ask, is that the other 2 BRI's are down, and last week it was flapping |
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| 10:13.07 | WIMPy | Well, if the line wasn't ok, it shouldn;t even try to send anything. |
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| 10:13.33 | WIMPy | But I don't know what the reality looks like with dahdi and activating lower layers. |
| 10:14.28 | WIMPy | And r00f: That's Asterisk issues, really. |
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| 10:15.22 | madduck | what may be a reason that voicemailmain does not accept a mailbox password listed exactly like that in secret.conf in the mailbox dir? |
| 10:15.57 | madduck | <PROTECTED> |
| 10:16.02 | WIMPy | What secret.conf? |
| 10:16.13 | madduck | jugband:/srv/asterisk/spool/voicemail/madduck/2# grep password secret.conf |
| 10:16.13 | madduck | password = 2781 |
| 10:17.23 | WIMPy | Did I miss simething? I have never seen that file. |
| 10:17.46 | madduck | it gets created if you set the password in voicemail.conf to the name of the box |
| 10:18.24 | madduck | passwordlocation=spooldir |
| 10:19.38 | WIMPy | I see. Never tried that. |
| 10:20.07 | WIMPy | Permissions? |
| 10:20.53 | madduck | -rw-rw---- 1 asterisk asterisk 214 Feb 23 11:11 secret.conf |
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| 14:10.04 | mookins | Hi, does anyone know of a channel to talk about hardware related to Asterisk? |
| 14:10.17 | [TK]D-Fender | Generally here. |
| 14:12.18 | mookins | awesome. I want to run my Asterisk installation in a VM. I need to connect 4 PSTN provided by our telco so the system and I think a SIP to PSTN gateway is what I am after. Jus twondering if it is that straight forward or am I missing something? |
| 14:12.52 | [TK]D-Fender | Nope, that's it |
| 14:12.55 | mookins | I am really need to VoIP systems and am learning on the run. I got VoIP setup on a LAN and calling between soft phones working. Now looking to push it to real phone lines. |
| 14:13.14 | mookins | [TK]D-Fender: Any brands I should take a good look at, or avoid? |
| 14:13.23 | mookins | I see quite a large price range with some of these units |
| 14:13.40 | [TK]D-Fender | http://www.telephonydepot.com/Catalog/Analog-Gateways/AudioCodes-MediaPack-MP-114-FXO |
| 14:14.50 | [TK]D-Fender | http://www.telephonydepot.com/Catalog/Analog-Gateways/Mediatrix-C730 |
| 14:15.09 | [TK]D-Fender | 2 reputable makes right there |
| 14:15.28 | mookins | Is echo cancellation a big problem? |
| 14:15.34 | [TK]D-Fender | It can be |
| 14:15.50 | [TK]D-Fender | proper interfaces do a good job of dealing with it |
| 14:16.32 | mookins | that Mediatrix, $220USD, or $625 in Australia LOL |
| 14:16.59 | [TK]D-Fender | yup, import tax is just stupid down under |
| 14:18.10 | mookins | yeah and all the grubby little hands it passes through on its way here |
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| 14:23.02 | mookins | ordered the Audiocodecs |
| 14:23.05 | mookins | thanks for the help :) |
| 14:25.39 | [TK]D-Fender | You're welcome |
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| 15:07.17 | babadoo | does asterisk remove a dot from the EXTEN variable f.e. in a call to firstname.lastname@someasterisk.net |
| 15:07.38 | babadoo | is it bad practice to have a dot in peer descriptors? |
| 15:09.46 | babadoo | my problem is, that in the channel EXTEN becomes firstnamelastname instead of firstname.lastname |
| 15:09.47 | [TK]D-Fender | EXTEN != peer descriptor |
| 15:10.20 | [TK]D-Fender | Show us the call. |
| 15:10.23 | [TK]D-Fender | ~pb |
| 15:10.23 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:10.25 | [TK]D-Fender | ^^^ |
| 15:10.29 | [TK]D-Fender | "sip set debug on" |
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| 15:15.45 | yun1989 | hello |
| 15:16.52 | yun1989 | when I exit a softphone in asterisk appeers extensions is connected in asterisk but in really it is not connected |
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| 15:17.45 | yun1989 | how can you ensure that in fact the extension is on? |
| 15:17.56 | [TK]D-Fender | "on" is not a thing |
| 15:18.26 | [TK]D-Fender | If your softphone doesn't UNREGISTER when it exits then there is not * can do except timeout on them. |
| 15:19.03 | yun1989 | in session-expires ? |
| 15:19.47 | yun1989 | which set up a time out session? |
| 15:19.55 | [TK]D-Fender | QUALIFY <------------ |
| 15:20.06 | [TK]D-Fender | this is not a "session" |
| 15:20.34 | [TK]D-Fender | Set a low registration timeout, or wait till they fail a qualify timeout |
| 15:22.34 | babadoo | [TK]D-Fender: thanx for the hint to use the sip debug option. it helped me to see that its the voip client zoiper, which removes the dot. in linphone the setup works. does it make sense that the dot is removed? should i have only user names without dot? |
| 15:23.11 | [TK]D-Fender | Clearly zoiper is doing it... perhaps a dialplan issue. Or it just doesn't play nice. |
| 15:23.38 | babadoo | is it better to have usernames without dots? |
| 15:24.16 | [TK]D-Fender | babadoo: You've already mixed up EXTEN vs USERNAME twice now... you should clarify this first.... |
| 15:24.17 | babadoo | i could not find something in the rfcs regarding that point. maybe i missed something. |
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| 15:25.28 | Ice_Strike | Anyone know any good alaternative of Zoiper? |
| 15:25.34 | Ice_Strike | Lightweight |
| 15:26.01 | babadoo | when a call arrives in the dialplan from the sip channel, i thought the EXTEN variable is showing the user part of the sip uri. |
| 15:26.33 | [TK]D-Fender | yes. Do not call that a USERNAME. |
| 15:28.52 | babadoo | i have "friends" registered in sip.conf. so i would call them friends. of course i can dial them with their sip uri which has a user part. does it makes sense this way? |
| 15:30.56 | [TK]D-Fender | babadoo: No. |
| 15:31.48 | [TK]D-Fender | babadoo: You do not dial THEM. You dial an EXTENSION. This is dialplan. Dialplan determines what happens when a call is placed. By no means is there any automatic association between what you dial and some DEVICE definition. |
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| 15:32.31 | [TK]D-Fender | babadoo: I could have a dialplan that never ever leads to a call out to another device. |
| 15:33.01 | Valduare | hows it going [TK]D-Fender |
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| 15:34.19 | [TK]D-Fender | babadoo: And almost everyone uses numeric extensions only. Many use mixed alpha device names. so I may have a [john] in sip.conf with that as a username .... but nobody is dialing "john" on a phone to call them. They have NUMBERED extens leading to dialing actual devices./ |
| 15:36.04 | Valduare | can you add smiley faces to extens might help fight a case of the âmondaysâ if you have to dial âsmiley face - extensionâ |
| 15:36.16 | babadoo | [TK]D-Fender: i understand. i allready use numeric extensions on which users can sign on with a hotdesk dialplan. But i wanted to let the users be able to call each other by "firstname.lastname@myasterisk.net" |
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| 15:36.47 | babadoo | of course only IF they have a client with alphnumeric input |
| 15:36.49 | cyford | Ice_Strike i use x-lite, good if u only need 1 line |
| 15:36.50 | [TK]D-Fender | that's a LOT of crap to dial vs "100" |
| 15:37.09 | [TK]D-Fender | And the fact of using a shitty soft-phone. |
| 15:37.12 | babadoo | its better to rememeber |
| 15:37.28 | Ice_Strike | For some reason, it is getting delayed to hear myself until I see "Unknown RTP codec 126 received from ..." |
| 15:37.32 | [TK]D-Fender | What, they don't have a speed-dial list? |
| 15:39.03 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 15:39.17 | cyford | lol interesting, i get that too from zoiper mobile client.. but it still works |
| 15:41.16 | cyford | are you using zoiper on windows phone or tablet Ice_Strike? |
| 15:41.53 | Ice_Strike | windows phone |
| 15:42.41 | Ice_Strike | I cant hear antthing until I see: Unknown RTP codec 126 received from |
| 15:42.52 | Ice_Strike | What is causing this? |
| 15:42.57 | cyford | the latest update has a fix for zoper codecs, |
| 15:43.40 | cyford | before u can set codecs in zoiper and it wouldnt really work, the update repaired that |
| 15:43.40 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-mwyauxzaainqofcy) |
| 15:44.25 | cyford | and zoiper for windows phone is new and combats alot of windows limitations, there is no other softphone for it atm |
| 15:44.31 | Ice_Strike | I meant windows |
| 15:44.35 | Ice_Strike | not for windows phone |
| 15:44.57 | cyford | ohh, that chnges everything lol |
| 15:46.17 | cyford | but i still think zoiper is trying to use a unsupported codec before it uses the correct one |
| 15:46.33 | cyford | try removing all except ulaw |
| 15:48.23 | Ice_Strike | ok |
| 15:48.43 | babadoo | [TK]D-Fender: thank you for your hints. speed-dial is interesting. but nevertheless it would be nice to be able to call a "friend" directly by their registered user name. do you think this is bad practice? |
| 15:49.51 | robmal | babadoo: What phones do you use? |
| 15:50.15 | [TK]D-Fender | babadoo: Naming them the same at all invites hack attempts. GEnerally your SIP account names should be harder to guess or sniff. Making them the same as what you dial is unnecessary exposure. |
| 15:50.28 | [TK]D-Fender | babadoo: Generally bad policy |
| 15:51.27 | cyford | hmm, i want hack babadoo.. lets try that username first ... |
| 15:51.29 | babadoo | ok, if you tell me that all kind of evil will come from this, i will of course stick to another scheme. |
| 15:52.08 | babadoo | nice try, cyford, but its babadoo.asterisknoob |
| 15:52.11 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:248:82fa:5bff:fe0a:dfef) |
| 15:53.10 | cyford | well asterisk is a given, and if that works noob would be the next step lol jkin |
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| 16:29.17 | *** join/#asterisk RadJackson (~Rad@cpe-xd001622.cust.jaguar-network.net) |
| 16:30.11 | RadJackson | Hello ive got a problem concerning Playback cmd , when ive got less than 50 simultenous calls , it works , once it surpasses certain number , like 100/200 , it starts to fail (WARNING[9109][C-0000025e]: app_playback.c:493 playback_exec: Playback failed o) , my file is WAV format. |
| 16:33.58 | *** join/#asterisk zapata (~zapata@2001:470:1f0b:11bc:c1c1:a559:75d3:c663) |
| 16:37.42 | mjordan | RadJackson: show the entire call leading up to that point. |
| 16:49.28 | *** join/#asterisk u0m3 (~u0m3@109.96.133.187) |
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| 17:01.37 | *** join/#asterisk h8 (~puck@unaffiliated/go) |
| 17:01.40 | h8 | hi everyone |
| 17:01.47 | h8 | does anyone run asterisk 11 and pfsense 2.2? |
| 17:17.23 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:20.04 | WIMPy | h8: Do you? |
| 17:20.30 | jeev | i run them both but not anywhere near eachother |
| 17:23.10 | h8 | yeah stupid question, asterisk behind pfsense, that is |
| 17:23.48 | cyford | asterisk in dmz or bypass |
| 17:24.54 | cyford | i ran it behind untangle.. bypassed filters and gave it priotiy in QoS |
| 17:27.49 | h8 | registration works, incoming call deosn't, as in, it doesn't get to the context, this is the debug: http://pastebin.com/xUHvjuK8 can anyone advise on what am i pucking up here? |
| 17:28.19 | h8 | I can see the incoming call in debug, but it doesn't show me anything of the likes of "call failed because there's no such context" |
| 17:29.51 | h8 | I do see SIP/2.0 401 Unauthorized tho |
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| 17:37.16 | h8 | ah nvm, figured it out, puck you insecure=very |
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| 17:59.10 | h8 | is there a solution to show the caller id if a call is forwarded? |
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| 18:57.34 | h8 | is there a solution to show the caller id if a call is forwarded? (sorry for double post) |
| 18:59.36 | file | I don't understand the question. |
| 19:01.42 | robmal | Someone outside PBX calls some ext. The ext dosn't answer so it goes to the users cell no. By default - the call on the cell is presented with the outgoing trunk number. Can this behaviour be changed? |
| 19:04.20 | file | Asterisk does not implicitly enforce that behavior. It's up to either the dialplan logic in use, or the underlying technology/provider of the outgoing call. |
| 19:05.10 | robmal | Thank you, that's the answer i was hoping for. |
| 19:28.40 | *** join/#asterisk babadoo (~babadoo@planeshift/art/associate/kinea) |
| 19:30.03 | babadoo | which voip clients play nice with video codecs of asterisk? i see linphone f.e. has VP8, which i cannot use in vanilla asterisk. |
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| 19:49.25 | [TK]D-Fender | babadoo: * never transcodes and only really passes through H.264, H.263, and H.263p |
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| 19:50.54 | babadoo | so linphone uses vp8 and theora. ekiga uses theora and h261. |
| 19:51.13 | babadoo | then i could only use ekiga. |
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| 19:52.04 | babadoo | what abouth261? |
| 19:55.33 | babadoo | oh i see linphone has H.263 and H.264 |
| 19:56.59 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Asterisk+video |
| 19:57.03 | [TK]D-Fender | yes, 261 as well |
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| 22:43.32 | mookins | Hi, just a quick question about IP phones and what not. Things like directory services, showing current line status, etc, are those all outside the scope of standard protocols and are provided by vendor specific and/or custom programmed interfaces? Or are there standards for these types of services that exist. |
| 22:45.11 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 22:45.49 | robmal | mookins: Provider specific address books, sorry. Also - at all cost avoid OpenStage by Siemens. |
| 22:48.47 | [TK]D-Fender | mookins, Address-books depends on the phones. There is no fixed standard |
| 22:49.10 | [TK]D-Fender | mookins, Line status = Presence. Asterisk already supports this |
| 22:50.04 | robmal | [TK]D-Fender: I feel somewhat ignored here, do i get bonus points for [] in my nickname? ;-) |
| 22:52.45 | [TK]D-Fender | Not sure what you mean.... |
| 22:53.40 | mookins | Thanks. Clears that up. I already avoid Siemens with regard to our industrial systems, don't need to tell me twice to convince me to skip over them in telephony too. |
| 22:54.43 | robmal | [TK]D-Fender: Our answers our usally simillar but somehow mine go unnoticed. I'd like to know why, if you're some asterisk guru - great. Otherwise - why? ;-) |
| 22:54.57 | *** join/#asterisk roler (~roler@unaffiliated/roler) |
| 22:54.59 | robmal | Our answers are* |
| 22:55.29 | roler | Does asterisk support RNIE? Looking to transfer external caller phone number to a forwarded number when they aren't at their desk. |
| 22:56.31 | [TK]D-Fender | robmal, He didn't answer me before you seemed oto think I hadn't been. |
| 22:58.04 | [TK]D-Fender | robmal, Our answers in this case were quite diferent |
| 22:59.14 | robmal | Please explain. |
| 22:59.50 | robmal | I like to be wrong as long someone explains to me why. |
| 23:00.04 | [TK]D-Fender | "Provider specific address books, sorry" <- What does sorry mean here? Too vague> You're aplogizing for addressbooks? Doesn't answer if there is a standard. Doesn't say if there is any option at all let alone how they are handled |
| 23:00.37 | [TK]D-Fender | And I answered his presense question which you didn't |
| 23:00.50 | [TK]D-Fender | So what I said and what you said are 2 very different things |
| 23:01.34 | [TK]D-Fender | Was that "Sorry" as in you're saying you don't know? As I said, that could mean almost anything. |
| 23:01.44 | robmal | Ok, i stand corrected. Thank you. |
| 23:02.22 | robmal | Sorry ment it's not that easy as you would expect it to be. |
| 23:03.08 | [TK]D-Fender | That wording did not really convery a meaningful answer |
| 23:04.20 | robmal | You're right. |
| 23:05.40 | [TK]D-Fender | You can be "sorry, I don't know" (confirm you don't have the answer), or "sorry you are having trouble" (condolences with no advice added), or "sorry, what you were hoping exist does not exist". |
| 23:06.02 | [TK]D-Fender | dusts off his Grammar Rangers suit... |
| 23:08.32 | robmal | Can we make it a BOFH answer class? I know i lost, but i didn't even start fighting. |
| 23:08.46 | robmal | Let's go with 1. |
| 23:10.17 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
| 23:12.21 | [TK]D-Fender | He'll take door #1 Bob! |
| 23:12.25 | [TK]D-Fender | WINNER! |
| 23:12.46 | [TK]D-Fender | \o/ |
| 23:13.42 | robmal | If i could switch - i would. |
| 23:14.17 | [TK]D-Fender | From what to what? |
| 23:14.39 | mookins | hahaha |
| 23:14.45 | mookins | gimme the cleaver |
| 23:14.56 | [TK]D-Fender | Door #2? There be lions behind said door! |
| 23:15.08 | [TK]D-Fender | grawr |
| 23:15.37 | robmal | I was looking for universal address book :-( |
| 23:16.55 | robmal | I'm making a web interface for AB for polycoms and i'd like to make it available for other phones. |
| 23:17.14 | robmal | EXCEPT OpenStage. |
| 23:17.57 | [TK]D-Fender | Polycom support 2 immediate forms. #1: Local text file and #2: LDAP |
| 23:18.08 | robmal | Making provisioning from SAIL to OpenStage was enough for this year. |
| 23:18.09 | [TK]D-Fender | What OTHER phones support... depends on them. |
| 23:19.34 | robmal | I'm adding a web interface for those text files. Also, our clients want a corporate addressbook not connected to LDAP. I don't know why, but that's not such a bad idea. |
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| 23:21.59 | robmal | I'll put it somewhere public once done, maybe it'll spin off to something useful. |
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| 23:42.14 | roler | I am trying to setup caller id forwarding, that is RNIE or RDNIS - time warner says I need to pass a "redirect CID" along with the external caller id to get this feature to work... Does asterisk support this? |
| 23:57.51 | [TK]D-Fender | Depends on the formatting |
| 23:58.11 | [TK]D-Fender | UYou'll have to gt a proper description from them as to EXACTLY which headers they are expecting |