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| 00:32.12 | *** join/#asterisk DanFromUK (DanFromUK@2.27.7.98) |
| 00:35.06 | DanFromUK | Hi all, does anyone know what variable contains the sip peer name? I would use callerid(num) but it changes. i need to be guaranteed that the varible contains the sip peer name. |
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| 01:04.51 | ariel_ | see if callerid(ani) or callerid(all) has it. |
| 01:05.16 | ariel_ | you can use NoOp() in your dial plan to see where it's at |
| 01:07.24 | DanFromUK | ok, i'll take a look. thanks |
| 01:08.27 | drynish | Ah so difficult |
| 01:09.28 | carrar | SIPCHANINFO(peername) also has it |
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| 02:14.37 | drynish | Got it at last! |
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| 09:18.09 | lcx | hi, have kind of a blackout ... there is a term for the fraud when the call is being answered, a soundfile played where you here people talking or something like "come to the phone, there is someone calling you". |
| 09:19.34 | KNERD | huh??? |
| 09:20.14 | lcx | never had that issue ? :) vendors charge interconneciton but have 0 costs |
| 09:23.30 | KNERD | What are you talking about? |
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| 12:00.32 | *** join/#asterisk elkluto (~roffer@fix.mix.barbra.triks.oblix.no) |
| 12:02.48 | elkluto | hello, i try to use a billion isdn kart to get incomming and outgoing to work, when i try to call inn to box it rings but i get Channel 0/2, span 1 got hangup, cause 27 and it hangs up, i use asterisk 1.6.2.16.1 |
| 12:04.28 | WIMPy | ... when you try to call out. ? |
| 12:04.53 | elkluto | im trying to call from my mobile to the box |
| 12:05.48 | WIMPy | If you see the call on Asterisk, but it doesn't get through, it's almost vertainly your dialplan. |
| 12:06.33 | elkluto | but i rings on my x-lite, and i can answer but it hangs up |
| 12:08.42 | WIMPy | Ah, ok. |
| 12:10.20 | elkluto | <PROTECTED> |
| 12:10.30 | WIMPy | Turn up debug ond verbose and put up the log aof a call attempt on pastebin. |
| 12:11.02 | WIMPy | That does not look like ringing. |
| 12:16.15 | elkluto | hers the output http://pastebin.com/6njip4g2 |
| 12:18.51 | WIMPy | Uh. Hard to read FreePBX stuff :-( |
| 12:19.04 | WIMPy | I haven't seen that before. |
| 12:19.12 | elkluto | ahh oki |
| 12:19.28 | WIMPy | Might be a misconfiguration in chan_dahdi. |
| 12:19.52 | WIMPy | Have you checkeced that interface type and signalling protocol are correct? |
| 12:20.23 | elkluto | i have tried but dont know what the values shuld be |
| 12:21.04 | elkluto | signalling=fxs_ks |
| 12:22.06 | WIMPy | It's either bri_cpe or bri_cpe_ptmp. |
| 12:22.30 | WIMPy | Do you know it you have a line in point-top-point or point-to-multipoint configuration? |
| 12:23.03 | elkluto | not shure, need to test, its not ordenary isdn, its emulatd from my broadband supplier |
| 12:23.44 | WIMPy | Oh, SIP via S0? |
| 12:23.46 | *** join/#asterisk hipitihop (~denis@202.153.70.15) |
| 12:24.03 | WIMPy | That's most probably ptmp. |
| 12:24.14 | elkluto | heh again im not shure,, need to investigate mroe i guess |
| 12:24.51 | WIMPy | And switchtype=euroisdn those two are the important bits. |
| 12:26.22 | elkluto | signalling=ptmp ? |
| 12:34.42 | elkluto | i now have signalling=bri_cpe_ptmp |
| 12:34.42 | elkluto | switchtype=euroisdn |
| 12:34.44 | WIMPy | bri_cpe_ptmp |
| 12:34.58 | WIMPy | Sound good. |
| 12:36.28 | elkluto | but still same problem, hers output of chan_dahdi http://pastebin.com/MAf7ysjA |
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| 12:38.08 | WIMPy | Yes. That's a FreePBX thing again. |
| 12:38.12 | WIMPy | ~freepbx |
| 12:38.12 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 12:38.36 | WIMPy | You need to look in one of the included files. |
| 12:38.44 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 12:39.34 | elkluto | but they are # out, and if i try to anable them, noting happens :) |
| 12:40.59 | WIMPy | No, # is not a comment. It's the syntax of the inclued. |
| 12:41.25 | WIMPy | -d |
| 12:42.17 | elkluto | ahh ok,, didnt know that,, so dahdi-channels.conf is adde to config |
| 12:42.33 | WIMPy | yes |
| 12:43.06 | elkluto | but ; unkomments it ? |
| 12:43.25 | WIMPy | ; is for comments, yes |
| 12:44.02 | [sr] | hi WIMPy |
| 12:44.33 | WIMPy | Hi [sr]. Got some interesting features again? ;-) |
| 12:45.06 | [sr] | WIMPy: eheh..not this time... no IT for today :p |
| 12:46.10 | elkluto | bah, you were right WIMPy , just changed signalling = bri_cpe to signalling = bri_cpe_ptmp and it works |
| 12:46.33 | WIMPy | great |
| 12:46.45 | elkluto | i was shure that when you add # in front of any include it was not added.. |
| 12:47.14 | elkluto | well i learn every day.. tnx WIMPy |
| 12:47.20 | WIMPy | It's the same syntax, C preprocessor directives use. |
| 12:50.05 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 12:51.30 | WIMPy | elkluto: Maybe you can find ouy your VoIP account data and directly configure them in Asterisk, bypassing the IAD. |
| 12:51.55 | WIMPy | s/ouy/out/ |
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| 14:46.02 | IPNixon | is there any way to get asterisk to tell a spa3102 to send a hook flash over the line port? |
| 14:49.03 | IPNixon | i don't even know if this is possible...i want to get asterisk to send the hook flash so it can dial a nortel feature code using a nortel ata |
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| 14:54.26 | *** join/#asterisk DanFromUK (DanFromUK@2.27.2.120) |
| 14:57.01 | DanFromUK | hi, ive been struggling with a dialplan. I need to retrieve the sip peer name, but all I can find is the CALLERID function which doesn't always contain the sip peer name. Are there any other variables that contain the sip peer name? |
| 14:58.20 | russellb | yup |
| 14:58.38 | russellb | ${CHANNEL(peer)} I think ... check *CLI> core show function CHANNEL |
| 14:58.52 | russellb | there should be a section that shows chan_sip specific args to CHANNEL() |
| 15:00.10 | russellb | IPNixon: there is a Flash() application that _might_ do it ... I can't remember. It's possible to send a Flash command over the RTP stream, though. I just don't remember how to invoke it from within asterisk. Flash() might be DAHDI only |
| 15:00.23 | russellb | disappears again |
| 15:00.27 | IPNixon | ohh, alright. thanks |
| 15:03.36 | vsniney | has anyone experienced cisco 7960s refusing to register to asterisk 1.6.2.19? |
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| 15:07.45 | DanFromUK | russellb: anything with CHANNEL() seems to come back as nothing. |
| 15:07.50 | DanFromUK | tried this, |
| 15:07.54 | DanFromUK | exten => 558,n,Verbose(${CHANNEL(peer)}) |
| 15:07.56 | DanFromUK | and also this |
| 15:08.01 | DanFromUK | exten => 558,n,Verbose(${CHANNEL(peername)}) |
| 15:08.06 | DanFromUK | Both came back blank |
| 15:08.42 | DanFromUK | Asterisk 1.4.42 |
| 15:10.14 | vsniney | huh. well, my issue is fixed |
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| 15:16.19 | Kobaz | is there a discount link for the astricon hotel? |
| 15:19.42 | DanFromUK | ${CHANNEL} returns something, but it doesnt contain the extension that's dialling. |
| 15:20.09 | Kobaz | CHANNEL won't necessarily have anything to do with the extension |
| 15:20.13 | Kobaz | unless it's a local channel |
| 15:21.20 | DanFromUK | here's the problem. for billing, i need to know which extension is dialling out. usually, CALLERID(num) contains the SIP Peer name. |
| 15:22.15 | DanFromUK | but when a call comes in, and the operator wants to perform an attended transfer to an external number, CALLERID(num) doesnt contain the SIP Peer name. Its either blank, or the CALLERID from the original incoming caller. |
| 15:24.54 | DanFromUK | is there any way to dump all the variables that are currently in use? |
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| 15:36.59 | vsniney | has anyone had any experience (good or bad) with asterisk + bluetooth ? |
| 15:46.43 | Iamnacho | vsniney: I've had mixed results. sometimes I can get it working, other times not. It also seems to depend on the phone model you use |
| 15:47.00 | Kobaz | DanFromUK: DumpChan() |
| 15:47.26 | Kobaz | DanFromUK: use SetVar in your sip config |
| 15:47.35 | Kobaz | actually setvar |
| 15:47.48 | Iamnacho | vsniney: when I can get everything working and if I don't make any changes it seems to work well, but if I make changes i seem to break things |
| 15:47.51 | Kobaz | setvar=phone_exten=1234 |
| 15:48.31 | Kobaz | on that sip peer.... so when that peer makes a call, you will have the info and not rely on callerid |
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| 15:48.49 | Kobaz | the setvar only applies to calls *from* that devices |
| 15:53.52 | DanFromUK | Looking at DumpChan, the SIP Peer extension is not available. |
| 15:54.44 | DanFromUK | I thought about setting my own variable to track which extension currently has the incoming call, but if the calls gets transferred internally, i don't know how to update the variable. |
| 16:15.24 | vsniney | Iamnacho: thanks. have always wondered about the support |
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| 16:42.45 | DanFromUK | hi all. |
| 16:42.59 | DanFromUK | regarding the SIP Channel name, is the ID always 8 characters? |
| 16:43.27 | DanFromUK | I want to strip out the SIP Peer name from SIP/PEERNAME-xxxxxxxx |
| 16:48.53 | Gugge | cut it by the - char |
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| 17:02.09 | DanFromUK | How do I work out the location of the - char? |
| 17:02.43 | DanFromUK | If i'm using ${CHANNELNAME:4:INDEX_OF_MINUS_SYMBOL} |
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| 17:35.46 | *** join/#asterisk Noah7372 (needle@ultra30.tptp.cc) |
| 17:36.34 | Noah7372 | Hi guys. This may sound like a stupid number, although how do I specify on a phone what exension it is, and how do I tell the asterisk server what phone is what extension? |
| 17:37.10 | Noah7372 | I am just trying to setup an asterisk server so I can trunk to google voice. |
| 17:37.46 | Noah7372 | And it looks like I am at a point for the extensions.conf rules of the DID provider, that I need to specify an extension of sorts. |
| 17:38.17 | Noah7372 | exten => SIP-ID,1,Dial(SIP/Extension) <-- instead of extension you should define the corresponding peer |
| 17:38.29 | Noah7372 | I don't really understand what this means..... |
| 17:40.46 | p3nguin | It means that SIP-ID is the extension in that example. SIP/PEER-NAME is how you define a device to dial. |
| 17:41.10 | Noah7372 | so I can do like SIP/172.17.1.7 ? |
| 17:42.01 | p3nguin | You should be using peer names as you have them defined in sip.conf. |
| 17:42.12 | p3nguin | SIP/jack would be [jack] from sip.conf |
| 17:43.08 | p3nguin | Reading your initial question, it is apparent to me that you do not realize that phones are not extensions and extensions are not phones. |
| 17:43.29 | p3nguin | Phones are not extensions, phones are devices. |
| 17:44.32 | Noah7372 | Right. Well I am not very familiar w/ asterisk or the phone system. |
| 17:44.57 | p3nguin | You do not specify "on a phone," as you put it, what "extension it is." |
| 17:44.59 | Noah7372 | I just want the DID that the asterisk will register to, to ring through to my wifi/sip phone. |
| 17:45.54 | Noah7372 | The only other way ot get what I am trying to get accomplished would be by spending a monthly fee that I don't have. I have an edge machine that is sparcv9 doing little more than routing, so it should be able to do this well for me too. |
| 17:46.34 | p3nguin | Create a register statement for your ITSP account in sip.conf. Create a peer entry in sip.conf for your ITSP. When a call is sent in, if the call matches the peer entry, it will be sent into the context that you define in the peer entry. At that point, it's all dial plan the rest of the way. Dial plan is configured in extensions.conf. |
| 17:46.46 | Noah7372 | I already made the register statement. |
| 17:46.54 | Noah7372 | I am on this extensions part, this peer part is throwing me off. |
| 17:47.17 | p3nguin | What extension do calls get sent to? A good ITSP will send calls to your phone number. |
| 17:47.24 | Noah7372 | THe DID provider has the sip.conf and extensions.conf settings for the register and the dial rules. |
| 17:47.31 | Noah7372 | it doesn't have any info about peers. |
| 17:47.36 | Noah7372 | is there somewhere I can read on it? |
| 17:47.40 | p3nguin | ~book |
| 17:47.40 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 17:47.50 | Noah7372 | That's not very sufficient or efficient. |
| 17:48.01 | Noah7372 | I was thinking more a specific section of a website. Although thanks. |
| 17:48.09 | p3nguin | It's plenty sufficient. |
| 17:48.09 | KNERD | ~burn book |
| 17:48.09 | infobot | ACTION pours gasoline all over book, ignites the fire, and then enjoys some toasty marshmallows with the glorious blaze |
| 17:48.28 | p3nguin | There's far more info in the book than you're asking for, so it is sufficient. |
| 17:49.17 | Noah7372 | It may be sufficient as a book on asterisk, although it is not sufficient for my current purposes. |
| 17:49.25 | Noah7372 | Which includes a time and hassle limit! |
| 17:49.28 | p3nguin | It's very sufficient. |
| 17:49.32 | Noah7372 | =:-] |
| 17:49.49 | p3nguin | You've confused lack of sufficiency with laziness. |
| 17:50.14 | Noah7372 | The only laziness is the book and its lack of fitness for my particular purpose. |
| 17:50.41 | p3nguin | All of the information is in there. How is that not fit for your purpose? |
| 17:50.51 | Noah7372 | Time! |
| 17:50.59 | p3nguin | Time is money. |
| 17:52.17 | Noah7372 | looks. |
| 17:52.40 | *** join/#asterisk dlynes (~dlynes@bas6-hamilton14-1176141096.dsl.bell.ca) |
| 18:01.29 | p3nguin | After you read the relevant bits of the book, if there is a part that you don't understand, I'll be happy to try to clarify it. |
| 18:03.48 | Noah7372 | Woo. I got it. I didn't need to. |
| 18:03.54 | Noah7372 | I think I got it. AFK a min! |
| 18:04.04 | Noah7372 | Phone has registered to asterisk and asterisk has registered to DID |
| 18:04.14 | p3nguin | You're half way there. |
| 18:04.47 | p3nguin | Now you have to have a peer definition for the ITSP to match incoming calls, and then have extensions to do something useful with the calls. |
| 18:07.45 | Noah7372 | Ah wait no the phone isn't registering. The DID registered though. |
| 18:08.00 | p3nguin | Do you have a peer definition for the phone? |
| 18:08.21 | Noah7372 | [Jul 3 14:07:12] NOTICE[13005]: chan_sip.c:21768 handle_request_register: Registration from '"anonymous" <sip:utstarcom@172.17.0.1>' failed for '172.17.1.7' - Peer is not supposed to register |
| 18:08.57 | Noah7372 | [Jul 3 14:07:58] ERROR[13005]: chan_sip.c:13206 register_verify: Peer 'utstarcom' is trying to register, but not configured as host=dynamic |
| 18:09.02 | Noah7372 | It's not static. Meh. |
| 18:09.15 | p3nguin | You must have incorrectly specified the phone's IP address instead of host=dynamic. |
| 18:09.46 | Noah7372 | The phone is static, it is 172.17.1.7 and that is what I put host= |
| 18:09.51 | Noah7372 | I will justs put dynamic |
| 18:10.02 | p3nguin | host=dynamic is for any device that must register to Asterisk. This is not a reflection of network configuration, i.e static vs. dhcp IP addressing. |
| 18:10.11 | Noah7372 | k |
| 18:11.56 | *** join/#asterisk boazb (~b@46-116-200-22.bb.netvision.net.il) |
| 18:12.48 | Noah7372 | Hmm ok it registers. Next step. I tried placing a call and received this error. |
| 18:12.49 | Noah7372 | [Jul 3 14:12:10] NOTICE[13113]: chan_sip.c:20281 handle_request_invite: Call from '1205463e0' to extension 'njm' rejected because extension not found in context 'sipgate_in'. |
| 18:13.06 | Noah7372 | I will just rename the extension njm |
| 18:13.12 | Noah7372 | shrugs |
| 18:13.36 | p3nguin | The call was sent to extension njm. Did you make that call over the PSTN via Sipgate? |
| 18:13.43 | Noah7372 | via my cell |
| 18:13.45 | Noah7372 | pstn.... |
| 18:14.05 | p3nguin | But it went through sipgate into Asterisk... to extension njm. |
| 18:14.25 | Noah7372 | Right. I'm not even going to care as to why. |
| 18:14.30 | Noah7372 | I am just going to rename the extension. |
| 18:14.46 | p3nguin | It could be because you added /njm onto your register statement. |
| 18:15.05 | Noah7372 | that is my ininitials, it might have something to do w/ the sipgate act too. I don't know. |
| 18:15.09 | Noah7372 | I don't really care. |
| 18:16.07 | p3nguin | If you're willing to accept calls to njm, then extension njm in the sipgate_in context needs to do something useful. |
| 18:18.27 | Noah7372 | ok |
| 18:21.00 | Noah7372 | This should suffice, after naming the peer 'njm' in sip.conf? exten => SIP-ID,1,Dial(SIP/njm) |
| 18:21.14 | p3nguin | no |
| 18:21.27 | Noah7372 | ok |
| 18:21.36 | p3nguin | That call went to extension njm. You have to have extension njm in extensions.conf. |
| 18:22.15 | p3nguin | What is the name given to your phone/device? |
| 18:22.37 | Noah7372 | utstarcom |
| 18:23.04 | p3nguin | exten => njm,1,Dial(SIP/utstarcom,30) |
| 18:23.24 | p3nguin | extension njm dials the utstarcom phone. |
| 18:23.31 | p3nguin | It'll time out after 30 seconds. |
| 18:23.37 | p3nguin | Adjust as needed. |
| 18:26.49 | Noah7372 | Yay. Phone call accepted. THanks p3nguin. |
| 18:27.20 | Noah7372 | My outbound is not working. |
| 18:27.33 | Noah7372 | piddles. Thanks p3nguin. |
| 18:27.53 | p3nguin | Do you have credits with sipgate to allow outbound calling? |
| 18:28.38 | *** join/#asterisk cyborg-one (1000@85-238-109-104.broadband.tenet.odessa.ua) |
| 18:29.16 | Noah7372 | [Jul 3 14:27:04] NOTICE[13267]: chan_sip.c:20281 handle_request_invite: Call from 'utstarcom' to extension '18005696972' rejected because extension not found in context 'default'. |
| 18:29.23 | Noah7372 | I am trying 1800 numbers. |
| 18:29.37 | p3nguin | The phone's peer entry needs a context. |
| 18:29.38 | Noah7372 | I really don't need it. I use google voice =:-] |
| 18:29.42 | Noah7372 | is poor.... |
| 18:29.42 | p3nguin | default is not correct. |
| 18:30.16 | p3nguin | If so, start with the context assigned to the phone. The call comes from the phone, matches the peer, goes into the assigned context, matches the called extension, then does something useful. |
| 18:30.35 | Noah7372 | Right. It doesn't no the outbound dialplan to use I am guessing. |
| 18:30.45 | p3nguin | In your case, the extension would need to Dial(SIP/sipgate), assuming your sipgate peer is named 'sipgate'. |
| 18:30.56 | Noah7372 | It is. |
| 18:31.05 | Noah7372 | Will do. Thanks. |
| 18:31.10 | Noah7372 | I think I can handle that. |
| 18:31.16 | p3nguin | Well, Dial(SIP/sipgate/${EXTEN}) to be more accurate |
| 18:32.35 | p3nguin | exten => _800XXXXXXX,1,Dial(SIP/sipgate/${EXTEN}) |
| 18:32.46 | p3nguin | or exten => _800XXXXXXX,1,Dial(SIP/sipgate/1${EXTEN}) |
| 18:32.54 | p3nguin | or exten => _1800XXXXXXX,1,Dial(SIP/sipgate/${EXTEN}) |
| 18:33.07 | Noah7372 | How can I just add everything from the phone? I have to set it context= in the peer define? |
| 18:33.08 | p3nguin | Some variation of that, depending on what dial strings sipgate accepts. |
| 18:33.26 | Noah7372 | I'll just add that. Thanks. |
| 18:33.37 | Noah7372 | 1888 too..... |
| 18:34.00 | p3nguin | In the [utstarcom] peer definition, you have to specify a context, such as context=phones, that has a "path" to the extension you're dialing on the phone. |
| 18:34.28 | p3nguin | I use a phones context, then include a few other contexts within phones. One of the included contexts is an outbound_calls context. |
| 18:35.13 | Noah7372 | Right. |
| 18:35.18 | Noah7372 | I see. |
| 18:36.35 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 18:39.39 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 18:40.05 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 18:41.11 | Noah7372 | Hmm it just rings and then goes to a busy signal. |
| 18:41.13 | Noah7372 | [sipgate_out] |
| 18:41.13 | Noah7372 | exten => _X.,1,Set(CALLERID(num)=njm) |
| 18:41.13 | Noah7372 | exten => _X.,2,Dial(SIP/${EXTEN}@sipgate,30,trg) |
| 18:41.13 | Noah7372 | exten => _X.,3,Hangup |
| 18:41.26 | Noah7372 | I have this. And context set to sipgate_out |
| 18:41.31 | Noah7372 | No errors any longer :| |
| 18:41.38 | p3nguin | I see errors. |
| 18:41.49 | Noah7372 | You do? |
| 18:41.58 | Noah7372 | the CALLERID? |
| 18:42.37 | p3nguin | njm is not a valid caller ID number. SIP/sipgate/${EXTEN} is the syntax. And you shouldn't be using options t, r, and g unless you know what they are for and have a reason to use them. |
| 18:42.51 | p3nguin | And you should be using n instead of 2 and 3. |
| 18:42.58 | Noah7372 | That is fresh from sipgate. |
| 18:43.21 | Noah7372 | That is what they say to do. |
| 18:43.27 | Noah7372 | I will change it. |
| 18:43.34 | p3nguin | If you would have asked me before you pasted if ITSPs ever know how to configure Asterisk, the answer would have been, "No." |
| 18:44.11 | p3nguin | It's rather silly that I've never seen a single ITSP provide *GOOD* example configs to users. |
| 18:44.11 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 18:45.14 | Noah7372 | I believe it. |
| 18:45.42 | p3nguin | By hanging out here, I've seen a lot of ITSP example configs, too. |
| 18:46.01 | p3nguin | Some of them don't even work AT ALL. |
| 18:47.08 | Noah7372 | LoL it works now. Wow. |
| 18:47.10 | Noah7372 | Amazing. |
| 18:47.35 | Noah7372 | There is another one to add to the list. |
| 18:48.30 | p3nguin | I don't find it to be an unreasonable expectation that a person's ITSP provide good, working, competent sample configs. |
| 18:48.59 | Noah7372 | Indeed. |
| 18:49.02 | Noah7372 | By all means. |
| 18:49.11 | p3nguin | A company in the industry should be able to give a customer a drop-in config. Many of them try, but few succeed. |
| 18:52.20 | carpe_diem | when will asterisk start using ptlib instead of pwlib for h323 support?? |
| 18:52.54 | Noah7372 | Hmm well strange. The 1800 numbers still don't actually ring, they give a busy signal, but when I try a local number I get the good ol' sipgate credit error. |
| 18:53.01 | Noah7372 | That's sort of odd. |
| 18:53.16 | Noah7372 | Like I said, I use google voice, so it really doesn't matter. That's gonna be the largest pain to setup. |
| 18:53.52 | p3nguin | If direct calling using gtalk is still working, it's probably easier to set up than the sipgate stuff you just went through. |
| 18:54.47 | Noah7372 | Well, there is no reason it shouldn't work.... in some sense anyways... I have gvoice the python google voice client installed on this machine and it still works |
| 18:55.05 | Noah7372 | And sipgate just issues a system() to that afaik. |
| 18:57.29 | Noah7372 | Yeh I just used it man. Forced a call from the sip phone to my cell from a sparcv9 nix command line =D |
| 18:57.36 | Noah7372 | munks w/ that now.. |
| 18:58.12 | Noah7372 | That is the part that you can't get for free. |
| 18:58.22 | p3nguin | What's not free? |
| 18:58.26 | Noah7372 | gvoice trunking where you don't have to get on the computer to making a call. |
| 18:58.31 | Noah7372 | well unless you host it yourself |
| 18:58.39 | Noah7372 | pbxes.org will trunk to google voice... it aint free though. |
| 18:58.55 | p3nguin | Asterisk does it just fine all by itself. |
| 18:59.04 | Noah7372 | I know. That is why I am setting up asterisk. |
| 18:59.14 | Noah7372 | I am sick of making calls from windows. |
| 18:59.21 | Noah7372 | I want to pick up the damn phone and make a call. |
| 18:59.59 | Noah7372 | Google voice uses the jabber protocol, not sip, so you can't just easily connect to it. |
| 19:00.07 | p3nguin | false |
| 19:00.13 | p3nguin | You can very easily connect to it. |
| 19:00.15 | Noah7372 | no, true, read on it man. |
| 19:00.25 | Noah7372 | the only way asterisk does it is by issuing system() to the gvoice python client |
| 19:00.34 | Noah7372 | which just logs into the website more or less and issues a call |
| 19:00.36 | Noah7372 | all through scripting |
| 19:00.40 | p3nguin | I don't need to read on it. I've set up direct calling between Google Voice and Asterisk. It worked just fine. |
| 19:00.46 | p3nguin | No scripting required. |
| 19:00.58 | Noah7372 | Are you sure you don't mean google talk? |
| 19:01.28 | Noah7372 | If you know something I don't. It'd sure help. |
| 19:01.32 | p3nguin | Well, to be fair, Google Voice is using gtalk in some form, which Asterisks must connect with. |
| 19:01.35 | *** join/#asterisk jjohnson (~jjohnson@officegateway.ethostream.com) |
| 19:01.51 | Noah7372 | right gtalk it is jabber/xdcmp or whatever |
| 19:01.58 | Noah7372 | there IS a module efor asterisk for it |
| 19:02.00 | p3nguin | The Asterisk wiki has a step-by-step guide on setting up gtalk and jabber in Asterisk for direct calling. |
| 19:02.12 | Noah7372 | well it is for gtalk |
| 19:02.15 | p3nguin | It's very simple. |
| 19:02.16 | Noah7372 | i looked at that module |
| 19:02.22 | p3nguin | It works well. |
| 19:02.24 | Noah7372 | it showed how to call people that are using gtalk |
| 19:02.29 | Noah7372 | it doesn't say anything about calling a phone number |
| 19:02.31 | p3nguin | It only takes a few minutes to set up. |
| 19:02.32 | Noah7372 | which is what google voice does |
| 19:02.57 | Noah7372 | will looks again. |
| 19:03.06 | p3nguin | The wiki page shows configuration for inbound and outbound direct calling. |
| 19:03.15 | Noah7372 | yea to gtalk users |
| 19:03.18 | Noah7372 | not to phone numbers |
| 19:03.18 | Noah7372 | afaik |
| 19:03.22 | Noah7372 | maybe I'm wrong |
| 19:03.42 | p3nguin | You're wrong. |
| 19:04.28 | Noah7372 | Is there any possibility you could point me at this wiki page? |
| 19:04.37 | p3nguin | Yes, give me just a moment to search it. |
| 19:04.48 | Noah7372 | http://www.voip-info.org/wiki/view/Asterisk+Google+Talk |
| 19:04.52 | Noah7372 | This is whwat you are talking about, no? |
| 19:05.12 | p3nguin | https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 19:05.22 | p3nguin | "the Asterisk wiki" |
| 19:06.08 | p3nguin | I've followed this and configured it myself. It works as desired for inbound and outbound PHONE CALLS to PHONE NUMBERS. |
| 19:06.44 | Noah7372 | Hmm I wonder if I would have to compile thoes modules. If so. I am probably not up for it today. |
| 19:06.48 | Noah7372 | looks if he has em. |
| 19:06.50 | Noah7372 | I doubt it. |
| 19:07.10 | Noah7372 | Wooo I do have it. |
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| 19:17.13 | jc319 | Is anyone using LDAP address book on Cisco 7940/7960s? I've set up the system with openldap and it works with a small set of entries (<=130 entries), but given CXML error message if I add one more entry into the LDAP container. Anyone using this with a directory with more than 130 entries? |
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| 19:46.35 | Noah7372 | p3nguin: I dunno man... I set it all up, and finally got it to authenticate, it seems it wants my email password and not my google voice password (which is google apps) |
| 19:46.51 | Noah7372 | Well I got it to log in and it won't let me make a call, it says there are not gtalk capable clients |
| 19:47.05 | Noah7372 | It isn't even my google voice password I used to log in, its the apps password for gtalk and stuff. |
| 19:47.22 | Noah7372 | JABBER: asterisk INCOMING: |
| 19:47.22 | Noah7372 | [Jul 3 15:43:29] ERROR[14639]: chan_gtalk.c:934 gtalk_alloc: no gtalk capable clients to talk to. |
| 19:47.23 | Noah7372 | [Jul 3 15:43:29] WARNING[14639]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'gtalk' (cause 0 - Unknown) |
| 19:47.38 | Noah7372 | That's what I get when I try making a phone call w/ the setup on the site. |
| 19:48.32 | p3nguin | I, and most other people I know who have configured this, use a gmail account and associated gvoice account -- the user and password are the same. |
| 19:48.57 | p3nguin | So my gmail email and password essentially are put in jabber.conf. |
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| 19:52.12 | Noah7372 | That isn't the problem.... |
| 19:52.27 | Noah7372 | I can use the command line gvoice client no different, it uses my google voice login. |
| 19:52.52 | Noah7372 | This wants the l ogin for the mail and apps and gtalk and stuff, which IS all the same. |
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| 19:53.12 | Noah7372 | It loggs in..... |
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| 20:12.26 | DanFromUK | Hi all, quick question. How do you force hangup after a successful dial command has been completed and the conversation has ended? Currently, the dialplan continues to voicemail if the sip extension hangs up before the caller. |
| 20:13.39 | Noah7372 | exten => njm,n,Hangup |
| 20:13.42 | Noah7372 | like that? |
| 20:14.33 | DanFromUK | yes, but at the moment, ive got:- |
| 20:14.46 | DanFromUK | exten => xxx,1,Dial(etc) |
| 20:14.56 | DanFromUK | exten => xxx,n,Voicemail |
| 20:15.16 | DanFromUK | i need it to jump over the voicemail line and go to hangup if the call was successful |
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| 20:15.29 | p3nguin | With a dial plan like that, what would you expect to happen? |
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| 20:15.55 | p3nguin | What Dial options are you using? |
| 20:16.02 | DanFromUK | 1sec. its in mysql |
| 20:16.32 | DanFromUK | just "T" |
| 20:17.45 | DanFromUK | something a little more unusual, is that sometimes, when the sip extension hangs up, the caller gets music on hold. which is not part of the dial plan at all. |
| 20:19.18 | p3nguin | Sounds like a phone I have on an ATA. A hook flash puts the call on hold, so dropping the handset can cause it to bounce and trigger a hook flash. |
| 20:20.47 | p3nguin | As for the hangup or voicemail, I use a Goto() along with DIALSTATUS to determine where the call goes after the Dial() is complete. |
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| 20:25.24 | DanFromUK | Happens when using a softphone or desktop sip phone. |
| 20:25.30 | DanFromUK | no ata's involved. |
| 20:25.44 | DanFromUK | also happens with incoming calls from the sip provider. |
| 20:25.56 | DanFromUK | the goto is a good idea. |
| 20:26.01 | DanFromUK | i'll add that. |
| 20:26.03 | DanFromUK | thanks |
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| 20:50.39 | DanFromUK | Does anyone have a professional recording saying "Your call is being transferred. Please hold" ? |
| 20:51.11 | p3nguin | There's probably one from Allison. |
| 20:55.36 | DanFromUK | I thought that but its not in the asterisk sound files. |
| 20:55.45 | DanFromUK | checked extras too |
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| 20:58.19 | AviMarcus | I'm curious.. where do folks go for A-Z termination? |
| 20:58.39 | AviMarcus | flowroute and voip.ms and the reasonable places for domestic USA have kinda horrible a-z rates |
| 20:59.08 | p3nguin | danfromuk: There's "Please hold while I try that extension." That's called "transfer." |
| 21:00.11 | DanFromUK | If the agent transfers a call to a cellphone, there is a long pause before the cellphone starts ringing, so i want to ensure the caller doesnt hang up. |
| 21:00.45 | p3nguin | You could try playing a ringing sound. |
| 21:02.44 | DanFromUK | i know |
| 21:02.48 | DanFromUK | ok, thanks |
| 21:03.32 | DanFromUK | how can I start MusicOnHold before hitting the dial command? |
| 21:03.58 | DanFromUK | I use this dial plan for other things, and only want to play music on hold when a transfer is occuring. |
| 21:04.19 | DanFromUK | if its just a standard outgoing call, i want it to behave like normal, so cant put the music on hold option into the dial cmd |
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| 21:05.26 | p3nguin | You can use StartMusicOnHold() |
| 21:06.27 | DanFromUK | is that in 1.4? its not listed on voip-info |
| 21:06.34 | p3nguin | Yes it is. |
| 21:06.55 | DanFromUK | will it stop automatically when a call is answered? |
| 21:07.00 | p3nguin | I have no idea. |
| 21:07.31 | DanFromUK | ok. thanks. i'll give it a go |
| 21:09.22 | AviMarcus | So.. anyone have a favorite a-z terminator? |
| 21:09.59 | p3nguin | Any ITSP out there is going to vary widely on their USA vs. international rates. |
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| 21:41.17 | KNERD | huh?? |
| 21:59.06 | p3nguin | huh what? |
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