| 00:15.56 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 00:28.37 | pabelanger | Having troubling finding out if GROUP_COUNT can be run from the AMI? |
| 00:35.16 | pabelanger | Guess not. What about running a dialplan function from AMI? |
| 00:42.13 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 00:42.57 | *** join/#asterisk bittwist (~USER@unaffiliated/bittwist) |
| 00:43.34 | *** join/#asterisk coppice (~chatzilla@210.17.255.115) |
| 00:47.07 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 00:48.44 | Kobaz | so is polycom blf still really limited, or is it just me? |
| 00:52.41 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 01:00.55 | Beltechs | where do i get a ast-playline in file?\ |
| 01:01.04 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 01:02.18 | pabelanger | Any pointers on using ${OUTBOUND_GROUP_ONCE} ? |
| 01:02.32 | pabelanger | specifically setting it, and using group_count |
| 01:13.37 | *** join/#asterisk coppice (~chatzilla@m121-202-100-173.smartone-vodafone.com) |
| 01:20.51 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 01:24.24 | pabelanger | So, apparently this is how you use it |
| 01:24.26 | pabelanger | <PROTECTED> |
| 01:24.26 | pabelanger | <PROTECTED> |
| 01:32.16 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 01:38.06 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
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| 01:52.58 | pabelanger | Kobaz: ping? |
| 01:53.37 | pabelanger | 302 redirect to #asterisk-dev |
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| 02:43.47 | *** join/#asterisk luckman212_phone (~luckman21@pool-173-77-253-145.nycmny.fios.verizon.net) |
| 02:44.24 | luckman212_phone | hello guys |
| 02:46.33 | luckman212_phone | anybody in here running 1.8.2-rc1? |
| 02:47.14 | WIMPy | ~ask |
| 02:47.15 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 02:51.23 | luckman212_phone | I've got an asterisk 1.8.2rc1 server set up here at home, and it works fine. the thing is I can't get any audio if I call from a softphone via 3g. If I call from that same softphone but while connected to a wifi spot (any wifi spot) it works. so I am thinking AT&T might blocking RTP?? |
| 02:51.46 | luckman212_phone | tried both Bria and Acrobits |
| 02:52.19 | luckman212_phone | I can register my extension and even initiate calls.. so the sip signalling is getting thru |
| 02:52.41 | WIMPy | Probably a NAT issue. |
| 02:54.19 | luckman212_phone | seems like it right? but I've tried nat=yes, nat=no, canreinvite=yes/no/nonat |
| 02:54.27 | luckman212_phone | and it does work via wifi |
| 02:55.05 | luckman212_phone | I am not even calling out via trunks, just intercomming my IVR for example |
| 03:03.00 | luckman212_phone | so I'm stumped |
| 03:05.40 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 03:06.36 | *** join/#asterisk lfod- (~email@unaffiliated/lfod-) |
| 03:09.49 | *** join/#asterisk RickCogley (~anonymous@EM114-48-32-108.pool.e-mobile.ne.jp) |
| 03:11.58 | drmessano | luckman212_phone, is your externhost/externip and localnet set correctly? |
| 03:12.26 | luckman212_phone | drmessano: yes it is |
| 03:12.52 | luckman212_phone | my pbx is at 10.10.10.248 I have localnet as 10.10.10.0/24 |
| 03:13.15 | luckman212_phone | externhost is my dyndns hostname |
| 03:14.01 | luckman212_phone | it really baffles me, I've been able to muddle thru most of my other asterisk problems with the help of google |
| 03:14.22 | luckman212_phone | but this one's got me stuck, for days now |
| 03:15.59 | sbrath | Any idea why this call: System(grep ${AMPUSER} /etc/asterisk/pinset_1) sometimes results in a false answer? |
| 03:18.01 | sbrath | I have that as a check in my outbound route, and the pinset_1 file only has like 2 lines in it, and if I go the "grep" in question on the command line, it reports no match. But sometimes in the asterisk-logs it matches when it shouldn't and then people get requested to enter an outbound pin-access-code... |
| 03:21.24 | sbrath | luckman212_phone: when you are on wifi are you talking to the inside address of the asterisk box? |
| 03:22.01 | *** join/#asterisk Preytell (~jerry.win@24-171-5-61.dhcp.stls.mo.charter.com) |
| 03:22.12 | luckman212_phone | <PROTECTED> |
| 03:22.31 | luckman212_phone | just doesnt work over 3G |
| 03:23.02 | sbrath | What softphone client & Phone & Carrier? |
| 03:23.06 | luckman212_phone | tried various codecs too-- g729, ulaw, g722 even |
| 03:24.06 | luckman212_phone | mostly Bria (counterpath) but also Acrobits , using iPhone 3GS on AT&T |
| 03:24.56 | sbrath | I use the 3CX client with no issues... But you are on AT&T, and god only knows what they do on their network. |
| 03:25.37 | sbrath | I know it might sound stupid, but the iPhone has a nice VPN solution, setup a vpn point on your Firewall, and VPN in over AT&T, and then AT&T can't do jack about you :) |
| 03:25.40 | luckman212_phone | yeah AT&T is a f___g nightmare |
| 03:26.16 | sbrath | I dropped them ... ( as in calls ) ... for Sprint... Saved a ass load of $$ on service and got a nice phone ( HTC EVO ) |
| 03:26.39 | sbrath | But seriously... Try the VPN... What is your firewall at the house, or is the Asterisk on the net raw? |
| 03:26.57 | sbrath | And if it's behind a firewall, how are you proxing the SIP ? |
| 03:27.21 | luckman212_phone | no, asterisk is on a small box sitting behind the nat, which is running Tomato |
| 03:27.39 | luckman212_phone | tomato forwards the sip (5060) to the pbx |
| 03:28.36 | sbrath | note this I'm guessting:: http://www.thriftyfun.com/tf91534303.tip.html |
| 03:29.37 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
| 03:30.05 | luckman212_phone | uhmmm...no :/ |
| 03:30.29 | sbrath | So Tomato is a replacement firmware.... Does it Proxy SIP or just Port Forward? |
| 03:31.15 | luckman212_phone | it doesn't proxy sip, I just port forward 5060 udp to the pbx |
| 03:31.57 | luckman212_phone | also 4569 |
| 03:32.01 | sbrath | ok |
| 03:33.27 | sbrath | I've used: http://www.zeroshell.net/eng/ as a firewall when I don't just make a full linux box... PS. The VPN I used with my iPhone was StrongSwan ( it dosen't do OpenVPN- Dumb Apple phone ) |
| 03:34.21 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 03:35.51 | luckman212_phone | hmm |
| 03:37.02 | sbrath | luckman212_phone: Just in case nobody warned you... SIP is un-encrypted, and over a public wifi is easy to sniff.. And some people have ended up with 100$'s phone bills as their asterisk was compromised..... I only use VPN connections from Sip phones to asterisk to protect this... |
| 03:37.38 | sbrath | iPhone will only do an IPSec VPN, and it's easy to setup on the phone.. |
| 03:38.05 | luckman212_phone | I'm looking into TLS & zrtp |
| 03:38.32 | sbrath | good deal. |
| 03:38.51 | luckman212_phone | maybe I need to replace my tomato router with so |
| 03:39.04 | luckman212_phone | something nerdier |
| 03:39.09 | luckman212_phone | nerdier |
| 03:39.22 | luckman212_phone | beefier I was trying to write |
| 03:39.39 | luckman212_phone | stupid iPhone keeps correcting my words |
| 03:39.59 | sbrath | It might be chalanging to get an iPhone SIP client compatible with zrtp.. |
| 03:40.18 | luckman212_phone | Acrobits does it |
| 03:40.31 | sbrath | You're IRC'ing on the iPhone :) I hope using Speech Reckognition ! |
| 03:40.56 | luckman212_phone | nah... keyboard |
| 03:41.07 | sbrath | oldschool... |
| 03:41.38 | luckman212_phone | was gonna do a pfsense |
| 03:42.17 | luckman212_phone | but zeroshell seems nice too just not familiar with it |
| 03:52.56 | sbrath | pfsense is nice.... But to be geeky you need to compile your firewall from source ( Gentoo++ ) |
| 03:54.10 | luckman212_phone | that might be above my pay grade |
| 03:55.32 | sbrath | so nobody has any idea why sometimes System(grep ) reports an invalid response? |
| 03:59.02 | *** join/#asterisk chopp (~chopp@unaffiliated/chopp) |
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| 04:46.15 | *** join/#asterisk raz (~raz@unaffiliated/raz) |
| 04:51.33 | raz | hrm |
| 04:51.53 | raz | is there a way to have the internal moh *not* start from the beginning for each call? |
| 04:52.18 | raz | i know mpg123 does that, but that seems to randomly freeze for me, with no music at all afterwards |
| 04:52.33 | raz | the internal moh works, but always starts from the beginning of the files |
| 04:54.29 | sbrath | the only way I can think of is to do MOH from /dev/audio or something and then have something else play the MOH into the mic ?? |
| 04:55.29 | raz | ew ;) |
| 04:55.42 | raz | i guess i'll give mpg123 another shot |
| 04:55.55 | raz | i think it's perhaps one of my mp3 files causing trouble, i think it worked fine for a long time |
| 04:56.22 | raz | and then again, silence is perhaps better than giving the callers the same song snippet over and over |
| 05:02.37 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
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| 05:40.12 | hatsisz | is anyone available to give some insight on how to get a Sangoma A200 series FXO card to work w/ AsteriskNOW ? |
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| 05:53.48 | sol326 | I'm giveing asterisk gui a go and can't seem to create a new SIP.... can you have more than one SIP available? |
| 05:54.15 | sol326 | I have one asterisk already registered with my ISP want to use this as my back up |
| 06:08.08 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 06:08.15 | *** part/#asterisk raz (~raz@unaffiliated/raz) |
| 06:09.27 | hatsisz | here's a pastebin of my issue. |
| 06:09.30 | hatsisz | http://paste.wm161.net/show/2326 |
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| 08:02.20 | schmidts | good morning |
| 08:07.59 | *** join/#asterisk lftsy (~lftsy@install.deckpoint.ch) |
| 08:08.11 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 08:08.41 | wdoekes2 | morning |
| 08:11.36 | sol326 | evening :D |
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| 09:06.12 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 09:44.33 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 09:45.27 | z4nD4R | hi all |
| 09:48.46 | z4nD4R | my asterisk is successfuly registered on remote server ( 2 users ). But i cant call this user on remote server... asterisk tell me |
| 09:48.46 | z4nD4R | [Jan 14 09:47:42] WARNING[16175]: chan_sip.c:13392 check_auth: username mismatch, have <trunk-01>, digest has <222361103> |
| 09:48.46 | z4nD4R | My settings looks like http://www.voiptalk.org/products/asterisk-sip-trunk.html.. SOMEBODY TO HELP?? |
| 09:52.40 | Tozz_ | u also have insecure=very? |
| 09:53.05 | z4nD4R | Tozz_: yes in each trunk section |
| 09:53.13 | wdoekes2 | hai broersma ;) |
| 09:53.22 | Tozz_ | lol |
| 09:53.37 | Tozz_ | z4nD4R: post your sip.conf somewhere |
| 09:53.45 | *** join/#asterisk RickCogley (~anonymous@EM114-48-93-29.pool.e-mobile.ne.jp) |
| 09:53.51 | wdoekes2 | you don't need insecure, if the rest is set up correctly |
| 09:53.57 | kaldemar | z4nD4R: the call matches to a peer that expects authentication by username "trunk-01", but the caller authenticates as "222361103". |
| 09:54.19 | Tozz_ | perhaps a missing auth= |
| 09:54.34 | tuxx- | Tozz_ ? :-o |
| 09:54.44 | Tozz_ | oh potver. Zit echt iedereen hier ;) |
| 09:54.50 | tuxx- | :D |
| 09:54.51 | wdoekes2 | haha |
| 09:54.58 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 09:55.56 | tuxx- | tijd voor taart. |
| 09:58.40 | Tozz_ | omnomnom |
| 10:01.01 | z4nD4R | kaldemar: but trunk-01 is name off section [trunk-01], and in this section is definedt one user to connect on remote server |
| 10:01.33 | kaldemar | z4nD4R: anyway, the incoming call matches that. |
| 10:02.07 | z4nD4R | kaldemar: her is my part of sip.conf |
| 10:02.17 | z4nD4R | http://pastebin.com/tgTqz4ie |
| 10:03.34 | Tozz_ | z4nD4R: try auth= |
| 10:03.36 | Tozz_ | auth = <authname> : Value assigned to the Digest username= SIP header |
| 10:03.56 | kaldemar | z4nD4R: incoming calls from that host match that peer. your numbered peer definitions are invalid, they don't have a host defined. |
| 10:04.40 | z4nD4R | Tozz_: in wich user? |
| 10:05.16 | z4nD4R | kaldemar: i muss define the 222361103 peer to? |
| 10:05.23 | kaldemar | auth= is for outbound calls. |
| 10:05.27 | Tozz_ | oh |
| 10:05.29 | Tozz_ | my bad ;) |
| 10:05.51 | *** join/#asterisk tyman (~tyler@173-12-219-177-Fresno.hfc.comcastbusiness.net) |
| 10:06.01 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 10:06.34 | wdoekes2 | I wonder why authentication is attempted when insecure=very |
| 10:06.54 | *** join/#asterisk RickCogley_ (~anonymous@EM111-188-128-124.pool.e-mobile.ne.jp) |
| 10:06.58 | wdoekes2 | pb a sip trace (sip set debug on) |
| 10:07.03 | kaldemar | z4nD4R: depends on what you want. since that call is coming from an ISP, you might want to match all calls from them by host as you do now. |
| 10:07.19 | kaldemar | wdoekes2: he probably has a version where "very" is not a valid option anymore. |
| 10:07.25 | wdoekes2 | mm :) |
| 10:07.29 | kaldemar | wdoekes2: good catch, btw. :) |
| 10:08.05 | wdoekes2 | I would suggest fixing the peer to send From: <sip:trunk-01@...> and using RPID for callerid |
| 10:08.06 | kaldemar | z4nD4R: change insecure=very to insecure=port,invite on trunk-01. |
| 10:08.16 | wdoekes2 | or that |
| 10:08.46 | z4nD4R | kaldemar: port means 5060 or 5061 ? |
| 10:08.50 | wdoekes2 | no |
| 10:08.52 | wdoekes2 | literally |
| 10:08.56 | z4nD4R | ok |
| 10:08.59 | z4nD4R | i try it |
| 10:09.51 | z4nD4R | kaldemar: [Jan 14 10:18:38] WARNING[16175]: chan_sip.c:25135 set_insecure_flags: Unknown insecure mode 'port,invite on trunk-01' on line 108 |
| 10:10.07 | z4nD4R | op |
| 10:10.17 | wdoekes2 | which asterisk version? |
| 10:10.31 | z4nD4R | 1.8.0 |
| 10:10.38 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 10:10.45 | wdoekes2 | insecure=port,invite |
| 10:10.51 | wdoekes2 | not insecure=port,invite on trunk-01 |
| 10:16.18 | z4nD4R | wdoekes2: kaldemar thx... incoming call works fine... |
| 10:16.43 | *** join/#asterisk dimm (~guest@unaffiliated/dimm) |
| 10:16.45 | z4nD4R | but i thig i have problem with codecs to pstn |
| 10:18.44 | merlin8282 | Hi guys, i've a problem setting up an IAX2 trunk between 2 asterisk servers. |
| 10:19.36 | merlin8282 | to test, I have set up a sip account on my *2, but I don't know where to put / how to use the "switch =>" statement. |
| 10:20.24 | kaldemar | merlin8282: why do you want to use the switch statement? |
| 10:21.09 | merlin8282 | kaldemar: *1 has an ISDN card in it, and *2 is a VM that does only sip/iax. |
| 10:21.45 | kaldemar | still no need to use the switch statement |
| 10:22.37 | merlin8282 | then I don't understand what I'm doing :/ |
| 10:22.38 | kaldemar | a plain Dial command with a proper pattern is enough. |
| 10:22.48 | merlin8282 | I tried to set up with example 3 on http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers |
| 10:23.08 | *** join/#asterisk RickCogley_ (~anonymous@EM111-188-28-175.pool.e-mobile.ne.jp) |
| 10:23.10 | kaldemar | merlin8282: see "Configuring IAX Trunks Between Asterisk Systems" in http://ofps.oreilly.com/titles/9780596517342/ch07.html#OutsideConnectivity_id291235 |
| 10:26.29 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 10:27.15 | z4nD4R | codec G.711a means in asterisk allow=alaw? |
| 10:27.29 | IsUp | yes |
| 10:27.59 | z4nD4R | hm.... but if i call from pstn... asterisk tell me [Jan 14 10:34:12] NOTICE[16175]: chan_sip.c:8419 process_sdp: No compatible codecs, not accepting this offer! |
| 10:28.43 | kaldemar | z4nD4R: how do you define codecs? pastebin. |
| 10:30.39 | z4nD4R | kaldemar: fixit... thx :D:D:D |
| 10:31.05 | merlin8282 | kaldemar: ok, but this doesn't tell me anything on my error I make. |
| 10:31.18 | IsUp | hello kaldemar |
| 10:31.18 | *** join/#asterisk jetlag (~jetlag@pool-173-61-239-155.cmdnnj.east.verizon.net) |
| 10:31.30 | kaldemar | merlin8282: neither do you. :) |
| 10:31.36 | kaldemar | IsUp: hello. |
| 10:31.52 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 10:34.08 | merlin8282 | kaldemar: sorry for my poor english, I don't understand what you mean. |
| 10:35.15 | kaldemar | merlin8282: you haven't shown an error to look at. anyway, you don't need the switch statement. if you know what numbers to send to the box with the ISDN card, just use an extension. |
| 10:36.36 | merlin8282 | so simply dial(isdnbox) and i'll be able to dial a number as if I would be on the isdnbox ? |
| 10:36.58 | kaldemar | merlin8282: yes. |
| 10:37.02 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:21b:63ff:feb5:e13b) |
| 10:37.20 | merlin8282 | ok. What does the switch, then ? I don't understand what it is for. |
| 10:37.28 | merlin8282 | (and didn't find any doc about it :/) |
| 10:38.07 | kaldemar | afaik the switch makes additional queries in IAX2 to find out whether the other box has a matching extension. |
| 10:45.10 | *** join/#asterisk shtoom (shtoom@14.96.171.199) |
| 10:46.44 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 10:48.01 | *** join/#asterisk RickCogley (~anonymous@ntkngw479129.kngw.nt.ftth.ppp.infoweb.ne.jp) |
| 10:49.31 | *** join/#asterisk chopp (~chopp@unaffiliated/chopp) |
| 11:08.49 | IsUp | if someone got a softphone, can we test something? |
| 11:09.00 | IsUp | i have troubles when i am connecting from my home computer to my PBX |
| 11:09.11 | IsUp | i got no audio when i make a call, but Echo works fine |
| 11:11.31 | merlin8282 | Ok, now my IAX dial is working. It does "Playing 'tt-weasels' (language 'de')", but I hear it in english ! And when I call 666 directly from a device connected to the isdnbox, it works (also -> in german) |
| 11:12.01 | merlin8282 | I checked, everywhere I can find "language=" it's set to "de". |
| 11:12.09 | merlin8282 | (on both boxes) |
| 11:13.17 | merlin8282 | with the echo-test it's the same: <IAX2/ast-10134> Playing 'demo-echotest' (language 'de') but playing it in english. |
| 11:13.59 | IsUp | something wrong with your sound file? |
| 11:14.10 | merlin8282 | FYI, the isdnbox is an old Asterisk 1.4.24.1-BRIstuffed-0.4.0-RC3e |
| 11:14.24 | merlin8282 | IsUp: no, it works when I call it directly from the isdnbox. |
| 11:15.04 | merlin8282 | and the second box is Asterisk 1.6.2.9-1~bpo50+3 |
| 11:15.34 | kaldemar | so it works on a SIP channel but not on an IAX one. |
| 11:15.45 | merlin8282 | kaldemar: right. |
| 11:16.37 | kaldemar | if you put a Set(CHANNEL(language)=de) before the playback, does it still play it in english? |
| 11:16.51 | kaldemar | are SIP and IAX using the same codec? |
| 11:17.34 | merlin8282 | no, iax uses gsm |
| 11:18.54 | kaldemar | do de sound files exist in gsm format? |
| 11:20.23 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:20.46 | IsUp | does PHPAGI works with 1.8? |
| 11:20.54 | merlin8282 | yes. I'm trying to listen to it, maybe it's in english... I see the problem. Thanks kaldemar |
| 11:22.05 | *** join/#asterisk schmidts (~schmidts@213.235.212.193) |
| 11:27.31 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 11:31.31 | merlin8282 | that's it, kaldemar, I lowered gsm codec priority and now with alaw it works. |
| 11:31.53 | kaldemar | were the gsm files in english? |
| 11:33.21 | merlin8282 | I can't read it, but I suppose yes. |
| 11:34.02 | merlin8282 | I saw in the console log that when it used gsm, it was english, and when using alaw -> DE. |
| 11:37.15 | merlin8282 | Ok. What I want/need to do is using the second asterisk server (the not-isdn box) for connecting all sip-hardphones, and forwarding all calls to the isdnbox. The isdnbox will do all the work, in fact. |
| 11:38.11 | merlin8282 | I want to do it this way because then I can use an up to date version of asterisk for the sip phones. The isdnbox needs an older version of asterisk. |
| 11:39.09 | WIMPy | Hmm. I's actially say that current Asterisk versions like ISDN much better. |
| 11:39.11 | WIMPy | d |
| 11:39.13 | merlin8282 | And that's the reason why I wanted to set up an IAX trunk. |
| 11:39.40 | merlin8282 | WIMPy: do you know if it works well with junghanns cards ? |
| 11:39.55 | WIMPy | Sure |
| 11:40.00 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 11:40.09 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN Just updated the feature list tonight. |
| 11:41.02 | WIMPy | There are three ways to do it with 1.8. |
| 11:41.13 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 11:47.32 | merlin8282 | that would be ? dahdi, misdn2 and visdn ? |
| 11:47.48 | WIMPy | dahdi, misdn1 or misdn2 |
| 11:47.56 | merlin8282 | ah ok. |
| 11:48.14 | WIMPy | See the end of that link. |
| 11:48.53 | tzafrir | WIMPy, about your patch: I'm trying to figure out if it should go in |
| 11:49.14 | WIMPy | Which one? |
| 11:49.19 | tzafrir | The main point: if the card does provide NT/TE, I don't want the parameter to be required |
| 11:49.36 | tzafrir | (and I actually have such a card in my test box) |
| 11:50.06 | WIMPy | Ah, it's not my patch. I just changed it to work with durrent dahdi. |
| 11:50.15 | tzafrir | https://issues.asterisk.org/view.php?id=17694 |
| 11:50.57 | WIMPy | I think it would be nice it it would only have to be specified in chan_dahdi.conf. |
| 11:51.18 | WIMPy | But that's obviousely not the way dahdi does it. |
| 11:57.16 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 12:02.33 | *** join/#asterisk joejoejoe (~marrr@dslb-188-104-183-127.pools.arcor-ip.net) |
| 12:04.14 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 12:05.50 | joejoejoe | hi! when i dial by peer, like Dial(0123456@arcor-out), asterisk puts the IP of the server in the invite/To-field instead of the hostname, which i think causes the 403. is it possible to put the domain there? |
| 12:09.40 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 12:16.13 | *** join/#asterisk s519 (~steve@119.42.87.13) |
| 12:24.21 | *** join/#asterisk kombi (~kombi@port-92-198-15-96.static.qsc.de) |
| 12:25.45 | kombi | I hear our own MOH when calling out an being put on hold, wicked! How does that work? |
| 12:36.58 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 12:41.05 | merlin8282 | Kobaz: maybe you are using isdn ? |
| 12:41.14 | merlin8282 | oops, sorry... kombi gone... |
| 12:49.20 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 12:54.16 | joejoejoe | hi! when i dial by peer, like Dial(0123456@arcor-out), asterisk puts the IP of the server in the invite/To-field instead of the hostname, which i think causes the 403. is it possible to put the domain there? here are two logs, the asterisk one does not work: http://pastebin.de/13830 |
| 12:58.05 | schmidts | joejoejoe it should work with both but if you really want to use the dns name in the request uri you could try the following: dial(01234@arcor-out!01234@sip.arcor.de) or whatever this domain should look like |
| 12:59.00 | joejoejoe | ok thanks, i'll try it... but what else could cause a 403? |
| 13:02.09 | joejoejoe | the To: field is correct now (with domain), the INVITE is still with IP... still 403 Forbidden :/ |
| 13:04.11 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 13:10.10 | *** join/#asterisk Faithful (~Faithful@202.189.73.144) |
| 13:16.03 | *** part/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 13:21.44 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 13:22.28 | schmidts | 403 forbidden normally means that your user is not allowed to. maybe you have a wrong password? |
| 13:24.36 | joejoejoe | the servers answer on the working phone is "SIP/2.0 407 Proxy Authentication Required", instead of forbidden... so I assume, authentication follows after that? |
| 13:25.27 | WIMPy | correct |
| 13:25.46 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:25.59 | joejoejoe | so there must be anything in the INVITE packet that the server doesnt like... i even changed the user agent :) |
| 13:26.11 | WIMPy | There is a parameter fromdomain. |
| 13:26.31 | joejoejoe | already set, just like fromuser |
| 13:26.49 | WIMPy | ok |
| 13:26.50 | joejoejoe | you've seen the paste above, wimpy? |
| 13:27.28 | WIMPy | no |
| 13:27.41 | joejoejoe | take a look please: http://pastebin.de/13830 |
| 13:28.15 | joejoejoe | meanwhile the To: header is the same too |
| 13:33.16 | WIMPy | Have you tried to set outboundproxy? |
| 13:34.47 | joejoejoe | yes, already set :/ now i tried to set the rtp and sip port's the same, still doesn't work |
| 13:35.15 | joejoejoe | somehow the branch= and tag= parts have a different length. does that matter? |
| 13:36.09 | WIMPy | I wouldn't think so. BTW: Are you trying to hack your phone``line''? |
| 13:37.35 | joejoejoe | what you mean? i'm just trying to make an outgoing call with asterisk... my router box (easybox) does it fine :/ |
| 13:38.40 | WIMPy | A simple "yes" would have been fine :-) Isn't there any documentation for Arcor lines with Asterisk around? |
| 13:40.28 | joejoejoe | there are some forum posts and it worked for those guys... but i didn't find any hint what could cause this problem. i'm trying to figure this out for some hours now :( |
| 13:42.51 | joejoejoe | you mean asterisk is to mighty for my phone line? ;) |
| 13:42.59 | WIMPy | Yes, that stuff can be quite annoying. |
| 13:43.42 | WIMPy | Well, when I tried the same thing for KDG, it took me quite a while to figure out why the exaple configs I found didn't work for me. |
| 13:44.01 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:44.06 | WIMPy | It turned out that I had to disable several features I enabled in the general section. |
| 13:44.50 | WIMPy | So if you have chaned anything there, it might be worth to try with a new sip.conf. |
| 13:44.54 | WIMPy | Or even to try an older Asterisk. |
| 13:46.00 | joejoejoe | ye, maybe... still it would be interesting what in this little packet is forbidden by the provider ;) i'll try to tune the general section |
| 13:46.31 | joejoejoe | maybe to throw out some "Allow:"-header parts |
| 13:47.22 | kaldemar | joejoejoe: does your router have some kind of an ALG enabled? |
| 13:48.35 | joejoejoe | i don't think so, i've natted all ports through... the REGISTER works, but not the invite |
| 13:50.01 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:52.48 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85cc.bcn.adamo.es) |
| 13:55.30 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
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| 14:01.52 | schmidts | Yeah found it again, the best use for a snom phone: http://twitpic.com/z8n36 :D :D :D |
| 14:02.13 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 14:03.06 | WIMPy | They may not be ideal, but they are the best I found so far. |
| 14:03.07 | *** join/#asterisk kaii (~kaii@port-87-193-161-154.static.qsc.de) |
| 14:04.18 | WIMPy | And they have good documentation. |
| 14:04.56 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:05.38 | ariel_ | tzafrir: are you around for some twinstar driver and questions? |
| 14:05.46 | ariel_ | Hello everyone |
| 14:08.37 | *** join/#asterisk moy_ (~moy@CPE003048b11058-CM0026f396812d.cpe.net.cable.rogers.com) |
| 14:09.14 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:21b:63ff:feb5:e13b) |
| 14:10.50 | schmidts | wimpy hate it or love it ;) i choosed to not love it |
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| 14:29.15 | w0ls0n | having a strange issue with dtmf/monowall |
| 14:29.16 | w0ls0n | inside the network it works fine but when I dial in from home any keypresses don't work |
| 14:29.20 | w0ls0n | any ideas? |
| 14:39.37 | puzzled | hi |
| 14:40.40 | TimeRider | w0ls0n open some more ports: , rtp, you probably only got 5060 open |
| 14:41.01 | w0ls0n | I have 5060 and 10K - 20K |
| 14:41.26 | TimeRider | ehm.... dunno then, sounds like the usual audio over nat problem somewhere though |
| 14:41.39 | TimeRider | your phone registers okay though yes? |
| 14:41.55 | w0ls0n | yes |
| 14:42.09 | w0ls0n | I did a bit of googling and some people reverted to 1.22 and said it worked fine |
| 14:42.09 | TimeRider | and does it ring if someone at the pbx calls you when you're at home? |
| 14:42.14 | w0ls0n | yep |
| 14:42.38 | TimeRider | and can you hear each other? |
| 14:42.41 | w0ls0n | we can |
| 14:43.08 | w0ls0n | been testing it the past 2-3 nights |
| 14:43.08 | TimeRider | hmmm, not sure then dude... |
| 14:43.23 | w0ls0n | yea I can revert to monowall 1.22 and see if the problem goes away |
| 14:43.39 | TimeRider | I think that's probably a V good idea, even just to test |
| 14:43.43 | w0ls0n | right |
| 14:43.52 | w0ls0n | I can do that this weekend when nobody is around |
| 14:44.00 | TimeRider | Have to let us know if it works, and maybe but the info on monowall site/let them know :) |
| 14:45.30 | w0ls0n | yea I had another issue with NAT when nobody from the outside could dial in at all |
| 14:45.42 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 14:46.16 | w0ls0n | externip=66.181.120.94 ; currently the ip for nospam.mine.nu |
| 14:46.16 | w0ls0n | localnet=192.168.9.0/255.255.255.0 |
| 14:47.04 | w0ls0n | but the problem is nospam is a DHCP addy so when the IP changes I will have to update the conf |
| 14:47.13 | w0ls0n | maybe that will go away if I downgrade monowall |
| 14:48.14 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 14:49.51 | tzafrir | ariel_, yes |
| 14:49.56 | TimeRider | not sure, there are ways online of altering things that have dynamic IP's - have a look on the pbxiaf pages - sure there's something there |
| 14:50.25 | TimeRider | get a static IP - it's nice :) |
| 14:50.30 | ariel_ | tzafrir: can I pm you about the twrinstar setup? |
| 14:50.44 | tzafrir | yes |
| 14:51.02 | w0ls0n | my BS isp wants $30 a month for a static IP |
| 14:52.16 | w0ls0n | guy in #freebsd suggested pfsense instead of monowall |
| 14:54.40 | kaldemar | w0ls0n: how does your setup deliver DTMF? |
| 14:55.00 | kaldemar | w0ls0n: i.e. which dmtfmode have you configured? |
| 14:56.30 | *** join/#asterisk JustinCampbell (~JustinCam@dsl092-226-147.phl1.dsl.speakeasy.net) |
| 14:57.10 | JustinCampbell | anyone here use puppet or chef for asterisk deployment? |
| 15:04.39 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-nmzfrtnbxoicsqdc) |
| 15:07.20 | *** join/#asterisk JustinCampbell (~JustinCam@dsl092-226-147.phl1.dsl.speakeasy.net) |
| 15:08.34 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:08.34 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:16.46 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:17.34 | merlin8282 | Is there any good debian package with asterisk 1.8 ? Lenny-backports "only" has 1.6. |
| 15:18.15 | JustinCampbell | merlin8282: make from source? |
| 15:19.31 | merlin8282 | JustinCampbell: yes, that's the idea, if there's no better alternative ;) |
| 15:19.54 | *** join/#asterisk Jasnejac (~kvirc@81.91.106.59) |
| 15:20.50 | TimeRider | merlin, I have an install script I just completed for it - tho it install freepbx with it, you could easy comment that bit out |
| 15:21.08 | TimeRider | http://timerider.co.uk/download/freedeb.tar |
| 15:21.24 | TimeRider | feedback very much appreciated :) |
| 15:22.14 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:22.39 | kaii | Is there any plausible reason why Asterisk 1.8 translates a Dial() targeting non-existent peer SIP/10 to IP address 0.0.0.10 which of course causes sip_xmit errors "returned -1: Invalid argument"? |
| 15:23.18 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 15:23.18 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:23.19 | kaii | "sip_xmit of 0xDEADBEEF (len 916) to 0.0.0.10:5060 returned -1: Invalid argument" |
| 15:23.26 | Tozz_ | if a peer does not exists it tries to connect to a hostname. SO I can imagine that if u only use numbers it will try to translate it into an IP address |
| 15:23.47 | TimeRider | merlin8282 : script compiles asterisk from source, freepbx bit is towards the end if you wanna do away with that bit |
| 15:24.03 | merlin8282 | TimeRider: thanks, i'll give it a try ;) |
| 15:24.05 | kaii | Tozz_: thats my understanding too, but it should only do this when given a valid IP i think. |
| 15:24.43 | Tozz_ | Asterisk disagrees ;) |
| 15:24.45 | TimeRider | merlin8282 : Make sure you give me some feedback, whether good or bad - worked in all tests I did, on latest Debian and Ubuntu :) |
| 15:24.52 | kaii | Tozz_: Asterisk 1.4 just gave "no peer found" and a congestion |
| 15:25.02 | Tozz_ | mm |
| 15:26.06 | merlin8282 | TimeRider: ok. Maybe you could put in the README an url where the script can be downloaded from ? |
| 15:26.28 | TimeRider | I could, but, I only just completed it - and even then will change soon most likely |
| 15:26.54 | TimeRider | if you do skip the freepbx bit, be sure to leave the security parts in near end, where it sets passwords etc for asteriskuser |
| 15:27.08 | merlin8282 | yes |
| 15:27.25 | TimeRider | Most of the install relies on BLANK passwords during install, then sets the passwords at the end for you :) |
| 15:27.52 | TimeRider | so when mysql asks you 4 times for passwd - just hit ok - then it gets set at the end |
| 15:30.57 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 15:33.05 | TimeRider | merlin8282 : i just updated it some more :p see what I mean! |
| 15:33.17 | TimeRider | one yo uhave will work fine tho if yo ualready started, otherwise grab the new one |
| 15:33.50 | merlin8282 | I've already begun with the first... |
| 15:34.10 | TimeRider | that's cool |
| 15:34.12 | TimeRider | it'll work fine |
| 15:34.18 | TimeRider | just some tweaks, nothing much :) |
| 15:35.02 | *** join/#asterisk Kedare (~Kedare@onyxia.netyxia.net) |
| 15:35.12 | merlin8282 | ok. Why is this "adduser asterisk blah" in the FreePBX installation block ? It should be done before beginning with FreePBX, no ? |
| 15:37.09 | TimeRider | I have NO idea, lol - it is a bit messy, with old code commented out etc, but it works... so... |
| 15:37.23 | merlin8282 | ok. |
| 15:37.58 | *** join/#asterisk atan3 (~atan@unaffiliated/atan) |
| 15:38.17 | *** join/#asterisk sourcode (~code@ppp-58-8-233-88.revip2.asianet.co.th) |
| 15:38.33 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:39.31 | TimeRider | it's in the freepbx bit, cos freepbx is all in the script that uses apache - thus created the asterisk for apache |
| 15:39.37 | TimeRider | but still a mess :p |
| 15:40.29 | TimeRider | make sure you get the right addons in the menuconfig bit - since addons is included in asterisk1.8 now |
| 15:40.41 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
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| 15:45.50 | *** join/#asterisk chadkouse (~chadkouse@75-93-182-214.day.clearwire-dns.net) |
| 15:46.03 | chadkouse | hey guys -- i have a weird problem that just "started" happening |
| 15:46.18 | chadkouse | I have 2 phone lines through my cable company -- running into a digium card using the zaptel drives |
| 15:46.37 | chadkouse | suddenly any calls coming in on those lines just keep ringing even though someone has answered on the asterisk server |
| 15:46.57 | chadkouse | as in -- the person calling from the outside continues to hear ringing |
| 15:47.06 | JustinCampbell | ha |
| 15:47.12 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 15:47.18 | JustinCampbell | no dialplan changes caused it? |
| 15:47.37 | chadkouse | nothing changed in the server config in the last 3-4 moths |
| 15:47.41 | chadkouse | months* even :) |
| 15:48.11 | chadkouse | my guess is the cable company changed something on their end and I'm not sending the right signal that "I've answered this call, stop playing ring.wav" :D |
| 15:51.08 | kaldemar | what kind of lines? |
| 15:52.39 | chadkouse | what do you mean ? |
| 15:53.04 | kaldemar | the used technology. are they analog lines? |
| 15:53.24 | chadkouse | well they are analog from the cable companies modem to my server. |
| 15:53.36 | chadkouse | then they go out over the cable companies voip network |
| 15:54.26 | kaldemar | and they administrate the modem? |
| 15:55.03 | kaldemar | as in firmware updates and configuration? |
| 15:56.11 | merlin8282 | TimeRider: /var/lib/asterisk/bin/module_admin is also freepbx stuff ? |
| 15:56.55 | *** join/#asterisk coppice (~chatzilla@210.17.255.115) |
| 15:57.40 | TimeRider | yes merlin |
| 16:02.23 | merlin8282 | bye all, I'm back monday. |
| 16:02.58 | TimeRider | ya out? it work? |
| 16:03.41 | merlin8282 | not yet tested, I'll do it monday, maybe. |
| 16:03.54 | TimeRider | Okies, let me know :) |
| 16:03.56 | merlin8282 | I'm also not finished modifying it. |
| 16:04.08 | TimeRider | No worries, when ya done, send me a copy :) |
| 16:04.13 | merlin8282 | TimeRider: yes. If I forget, whisper me ;) |
| 16:04.41 | TimeRider | no probs :) address in the readme... have a great weekend :) |
| 16:04.53 | merlin8282 | thanks ! Bye ;) |
| 16:13.56 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-158-17.dsl.stlsmo.sbcglobal.net) |
| 16:16.13 | LemensTS | Maybe developers here can help me. I am using svn on a project, I work in the trunk directory. I just did "svn copy trunk tags/1.0.0" to start creating versions. I was telling my users to just checkout the trunk, should I start telling them to check out versions in "tag"? Reason I see checking out the versions instead of using trunk, is if they do svn update, it may break something if they havent done it in a long time and I |
| 16:16.48 | JustinCampbell | i like git |
| 16:16.48 | LemensTS | And if they do a new version it can have an update script to update from the last couple versions. |
| 16:16.56 | JustinCampbell | but other than that i have nothing constructive to add |
| 16:18.29 | LemensTS | Yea I have used git also, im using sourceforge and they use svn. |
| 16:19.04 | coppice | you can use GIT at sourceforge |
| 16:20.43 | LemensTS | coppice: oh didn't know that. seems alot of people are using GIT instead of svn anymore. |
| 16:21.32 | *** join/#asterisk theHub (~karl@69.177.93.21) |
| 16:27.16 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 16:31.22 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:31.22 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:31.25 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-158-17.dsl.stlsmo.sbcglobal.net) |
| 16:32.19 | *** join/#asterisk mechbangirc (~mechbangi@mbl-82-58-183.dsl.net.pk) |
| 16:32.47 | mechbangirc | hi what kind of cpu do i need to handle 1000 SIP concurrent calls on asterisk 1.4 without transcoding? |
| 16:33.55 | Tozz_ | we have intel core2duo 6600, works as a charm for over 1000 calls |
| 16:34.07 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 16:34.39 | mechbangirc | Tozz_: can you tell me the load |
| 16:35.12 | Tozz_ | i'm afraid not, as we dont usually have that many calls |
| 16:35.37 | Tozz_ | but its basicly pushing around packets. Asterisk doesnt have to do very much if u dont have to do transcoding |
| 16:36.06 | mechbangirc | i agree, transcoding is the real killer |
| 16:36.29 | Tozz_ | jep |
| 16:36.31 | Qwell | media passing through Asterisk too |
| 16:36.40 | Qwell | if media is external, 1k calls is nothing |
| 16:36.52 | _Corey_ | I was going to say, if Asterisk is in the media path it's going to pull that number down a lott |
| 16:37.42 | *** join/#asterisk Defraz (~Defraz@63-226-95-152.dia.static.qwest.net) |
| 16:38.47 | Tozz_ | ofcourse it also depends on your dialplan. If it does lots of tricks with AGI scripts and what not |
| 16:40.36 | mechbangirc | i plan to use a2billing which makes use of AGI dont know how much that gonna pull down concurrent calls number |
| 16:41.44 | *** join/#asterisk millerx (~millerx@rrcs-24-73-84-250.se.biz.rr.com) |
| 16:41.52 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 16:42.45 | Qwell | mechbangirc: if it uses FastAGI, then it would be extremely minimal. Otherwise, consider that you're spawning a new instance of perl/php/whatever with every AGI() call. |
| 16:43.48 | mechbangirc | Qwell: no it uses deadagi. gotta calculate duration blabla billing stuff |
| 16:44.55 | *** join/#asterisk chasing`Sol (~rc4@smtp.master-zone.net) |
| 16:46.02 | _Corey_ | probably should move away from deadagi |
| 16:46.06 | _Corey_ | it's deprecated |
| 16:46.38 | Qwell | No it isn't. |
| 16:46.46 | _Corey_ | 1.6+ |
| 16:47.07 | mechbangirc | deprecated in 1.6? |
| 16:47.31 | _Corey_ | yeah |
| 16:47.40 | _Corey_ | you can do the same thing with AGI |
| 16:48.11 | Qwell | You're just supposed to use AGI() now, is all. |
| 16:48.11 | Qwell | The concept of it still exists |
| 16:48.21 | JustinCampbell | "dead" |
| 16:48.52 | _Corey_ | you just need to monitor for SIG_TERM and whatever |
| 16:49.04 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 16:52.23 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 16:53.20 | *** part/#asterisk mechbangirc (~mechbangi@mbl-82-58-183.dsl.net.pk) |
| 17:10.35 | *** join/#asterisk chadkouse (~chadkouse@rrcs-74-218-87-242.central.biz.rr.com) |
| 17:10.40 | chadkouse | kaldemar: sorry my internet connection died -- yes the cable company admin's the modem.... |
| 17:14.35 | chadkouse | so my question is -- what zaptel / asterisk settings can I try to make the cable companies phone lines realize I've answered the call ? |
| 17:15.30 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 17:16.07 | *** join/#asterisk iscario (~quassel@laureades.davout.pck.nerim.net) |
| 17:19.02 | drmessano | chadkouse, can you answer the line with a normal phone? |
| 17:19.11 | chadkouse | i will try |
| 17:21.51 | drmessano | chadkouse, if you answer an analog line and the cable company ATA doesn't show it's being answered.. either their ATA is bad/misconfigured, or your card isn't really pulling up the line. An analog phone will answer half of that |
| 17:22.16 | chadkouse | drmessano: yeah trying to get that test done now... |
| 17:22.23 | chadkouse | drmessano: thanks for the info |
| 17:22.25 | drmessano | You don't send any "signal" that you've answered the line, no more than a $6 walmart phone sends a "signal" down the line |
| 17:22.54 | *** part/#asterisk ihor (~Miranda@194.44.15.90) |
| 17:25.15 | chadkouse | drmessano: ok I was also thinking maybe it's answering too fast -- so I turned on the option to signal RIGNING and added a 1 second wait before answering. |
| 17:25.23 | chadkouse | drmessano: (those didn't seem to help) |
| 17:26.20 | drmessano | What about the analog phone? |
| 17:27.25 | chadkouse | drmessano: unfortunately I am remote so I'm relying on the people in the office to hook it up -- so far they said they hooked it up, and when I called it didn't ring.. so I'm thinking they hooked it up wrong |
| 17:28.54 | *** join/#asterisk gregd (~gregd@188-220-38-34.zone11.bethere.co.uk) |
| 17:29.18 | gregd | hi guys, is it possible to create a conditional include of a context? |
| 17:29.35 | gregd | depending on a variable? |
| 17:32.56 | *** join/#asterisk chasing`Sol (~rc4@smtp.master-zone.net) |
| 17:33.00 | chadkouse | drmessano: hehe they think it is a bad cable :) |
| 17:33.38 | drmessano | chadkouse, nice |
| 17:34.23 | chadkouse | drmessano: weird -- so maybe like 1 of the wires was damaged. |
| 17:34.38 | chadkouse | and the server was receiving data but not able to transmit it :) |
| 17:35.56 | chadkouse | drmessano: muchas gracias |
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| 17:42.53 | *** part/#asterisk mercutioviz (~michaelco@freeswitch/developer/msc) |
| 17:47.01 | *** join/#asterisk phr0zen (phr0zen@blk-224-135-210.eastlink.ca) |
| 17:47.13 | phr0zen | hello, i am wondering if anyone can help me with a chanspy issue |
| 17:49.53 | phr0zen | basically, i am wondering how i can configure it so that i can enter in another extension, then switch to listen in on that one |
| 17:50.11 | phr0zen | right now, i have it so you dial in, it asks for the password, then it asks for extension number |
| 17:50.23 | phr0zen | works fine, but will not let me switch it to another extension |
| 17:50.34 | paulc | phr0zen: I've been playing with that lately.. isn't there an option where you can enter digits whilst listening and it'll flip? |
| 17:50.42 | phr0zen | i thought so |
| 17:50.44 | paulc | Not sure on ChanSpy vs ExtenSpy but I defintely read it somewhere |
| 17:50.46 | phr0zen | doesnt seem to work though |
| 17:50.55 | paulc | what's your spy line of dialplan look like? |
| 17:51.07 | *** part/#asterisk gregd (~gregd@188-220-38-34.zone11.bethere.co.uk) |
| 17:51.54 | phr0zen | exten => 779,1,Macro(user-callerid) |
| 17:51.54 | phr0zen | exten => 779,2,Authenticate(1234) |
| 17:51.54 | phr0zen | exten => 779,3,Read(SPYNUM,extension) |
| 17:51.54 | phr0zen | exten => 779,4,ChanSpy(SIP/${SPYNUM}) |
| 17:51.58 | phr0zen | like so |
| 17:53.52 | paulc | Hmm.. I'm looking at "core show application chanspy" and don't see the option I thought I'd seen.. but.. you could use X to allow escape to a single digit extension, that then goes back to your Read() line - you'd just need a single digit extension in the same context. |
| 17:55.12 | paulc | Ah, no.. it scrolled off the top of the screen :) it says in the docs/help: "Dialing a series of digits followed by '#' builds a channel name to append to 'chanprefix'. For example, executing ChanSpy(Agent) and then dialing the digits '1234#' while spying will begin spying on the channel 'Agent/1234'." |
| 17:55.33 | paulc | I gotta run to a meeting at the day job :-s but let me know how you get on (back in 30) |
| 17:55.36 | phr0zen | so what would i add then to my config? |
| 17:56.00 | phr0zen | i'll be here in 30 lol |
| 17:56.03 | paulc | phrozen: I think nothing.. you just need to enter new digits whilst listening, then press #, and you'll then monitor SIP/xxxxx (whatever you typed in) |
| 17:56.13 | phr0zen | doesnt seem to do that |
| 17:56.16 | phr0zen | might try again |
| 17:58.18 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 17:58.44 | phr0zen | yea, no diec |
| 17:58.46 | phr0zen | dice |
| 17:58.52 | phr0zen | i listen to say 1234 |
| 17:59.09 | phr0zen | press 4321# ... systems plays message "SIP 1234" |
| 18:05.08 | *** join/#asterisk sekil (~Ognjen@78.24.104.73) |
| 18:06.34 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.6.23) |
| 18:08.16 | *** join/#asterisk [o_o]owl (~o_o]owl@unaffiliated/goldroger) |
| 18:13.00 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 18:15.04 | *** join/#asterisk w0ls0n (~w0ls0n@94-120-181-66.dsl.sacoriver.net) |
| 18:15.10 | w0ls0n | good afternoon all. |
| 18:16.23 | w0ls0n | I came in yesterday asking about dtmf not working. I swapped to pfsense (rather than monowall) and I still can't dial an option. I can dial Google Australia but when I hit ANY number nothing happens. |
| 18:16.55 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 18:19.13 | *** join/#asterisk citywok (~kvirc@67-134-194-33.dia.static.qwest.net) |
| 18:21.17 | Tozz_ | w0ls0n: what dtmfmode are u using? |
| 18:23.44 | w0ls0n | rfc2833' |
| 18:23.50 | w0ls0n | without ' |
| 18:28.46 | *** join/#asterisk Lann (ab457d09@gateway/web/freenode/ip.171.69.125.9) |
| 18:28.51 | Lann | heyas |
| 18:28.58 | *** join/#asterisk Da-Geek (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 18:29.05 | Lann | do you guys know if asterisk can play multiple audio files to the same caller simultaneously? |
| 18:29.37 | Qwell | Lann: I wouldn't think so |
| 18:30.37 | JerJer | perhaps via something crazy like AGI piping soxmix |
| 18:30.50 | Qwell | Lann: why would you want to? |
| 18:31.15 | Lann | planning out the feasability of a PBX game |
| 18:32.09 | Qwell | yeah that's gonna be tricky |
| 18:32.42 | Lann | not being able to play multiple audio files simultaneously is a huge problem |
| 18:32.49 | JustinCampbell | have you looked at twilio or tropo? |
| 18:33.40 | russellb | well ... it's possible ... you could hook JACK into the audio path and let it do the mixing |
| 18:33.44 | Lann | no but the amount of backend coding i'm going to have might prevent me from using someone's service like that |
| 18:33.57 | Lann | JACK? |
| 18:33.58 | JustinCampbell | Lann: theyre really flexible |
| 18:34.14 | russellb | JACK == Jack Audio Connection Kit |
| 18:34.19 | russellb | jackaudio.org IIRC |
| 18:34.55 | w0ls0n | I changed dtmfmode = inband and it worked |
| 18:35.38 | w0ls0n | if I only change that in my sip providers info, the dtmfmode in general will overrride it if I setup another sip provider for the US right? |
| 18:36.35 | JustinCampbell | Lann: https://www.tropo.com/docs/webapi/new_tropo_web_api_overview.htm |
| 18:38.46 | phr0zen | i was asking a question regarding chanspy earlier, not sure if you guys were around then |
| 18:38.51 | phr0zen | perhaps you can assist? |
| 18:39.15 | phr0zen | basically, once in chanspy on listening to an extension, i cannot dial say 1234# to get to another extension |
| 18:40.32 | phr0zen | i dial chanspy -> enter password -> enter 1234# |
| 18:40.45 | phr0zen | then try to listen to 4321 by entering in 4321# |
| 18:41.02 | phr0zen | the system plays the message "sip 1234" (basically stuck on that first extension) |
| 18:41.52 | russellb | Lann: Qwell: Another hack to play multiple audio files would be to use ChanSpy()'s whisper mode. You can originate multiple calls that each are doing a playback of a sound file connected to ChanSpy() in whisper mode, all playing audio to the same channel. |
| 18:44.06 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 18:45.34 | Lann | jack looks promising |
| 18:45.49 | Lann | this is going to be a huge pita |
| 18:45.59 | russellb | i'd go the route of my second suggestion |
| 18:46.01 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 18:46.37 | Lann | yeah if i could simply play multiple audio files simultaneously to a single asterisk user it would save tons of headache |
| 18:46.51 | Lann | i'll look at the whisper mode |
| 18:46.58 | russellb | well you can ... it's just there's not a trivial way to do it |
| 18:47.18 | russellb | but you can play audio into a call using Originate and making a call that connects Playback() to ChanSpy() |
| 18:47.24 | russellb | and you can have multiple instances of that going at once |
| 18:47.35 | Lann | oic |
| 18:47.41 | russellb | leifmadsen: *adds that ^^^ to the cookbook ideas list* |
| 18:47.56 | Lann | this is getting pretty complex |
| 18:48.03 | russellb | yup |
| 18:48.08 | Lann | like i was thinking that a player has to say "say" to talk |
| 18:48.17 | Lann | since it'd be lame to parse commands from normal speech |
| 18:49.00 | Lann | when he says say...since i'm normally parsing his commands i have to start doing something else with his recorded speech and somehow stream it to asterisk. this implies that i can play multiple audio files simultaneously to basically set up a buffered chat room |
| 18:49.10 | citywok | lann are you going to provide a phone number, or make people install a sip client and register directly? |
| 18:49.20 | Lann | i was thinking both |
| 18:49.59 | Lann | i'll develop it with sip and then get a phone number going |
| 18:50.13 | Lann | from broadvoice or some such |
| 18:51.10 | Lann | so another big assumption i'm making is that i can pipe a user's speach from asterisk to a script. I thought i'd look at the festival code for an example of this |
| 18:52.12 | Lann | so command mode would be like: you connect to asterisk, asterisk connects to a client app i write and pipes what you say to it. Asterisk is also somehow waiting for audio filenames to play. This would allow commands and responses |
| 18:52.18 | Lann | "chat" will be a whole other issue |
| 18:56.23 | Lann | i think if i manage to get all of that working i'd pipe the commands to a MUD server (for those who know what that is) hehe |
| 18:58.09 | Qwell | Lann: oh dear |
| 18:58.18 | Lann | yeah |
| 18:58.31 | leifmadsen | russellb: I've already build the whisper a file to a channel via ChanSpy() |
| 18:58.35 | leifmadsen | s/build/built |
| 18:58.41 | russellb | leifmadsen: awesome |
| 18:58.47 | leifmadsen | I know it works. It was a bit of a pain to implement, but I made it work |
| 18:58.49 | russellb | leifmadsen: i think it's a powerful use of ChanSpy |
| 18:58.51 | Lann | leifmadsen: mind sharing? |
| 18:59.18 | leifmadsen | it actually works via a features.conf code to check a webservice via CURL() to play back the current call cost |
| 18:59.27 | leifmadsen | Lann: sorry, it's a clients dialplan so I can't share it right now |
| 18:59.42 | leifmadsen | but at least you know it can work. I used Originate, Local channels, and ChanSpy() |
| 18:59.53 | leifmadsen | (on 1.8) |
| 19:00.19 | Lann | i suppose that the legitimate business use of this might be something like the current call cost being spoken above your conversation |
| 19:01.11 | Lann | i really want to do this project after i get a lull in my current project heh |
| 19:01.42 | leifmadsen | ya, it was for an inbound call centre (pay per use) |
| 19:01.47 | leifmadsen | (psychics) |
| 19:01.49 | Lann | the only thing i cant really visualize is how you'd chat with other people. I almost need a way to key in and key out |
| 19:02.18 | leifmadsen | I didn't actually read your scrollback to know exactly what you're trying to do |
| 19:02.20 | citywok | put them in a conference bridge as a chat room |
| 19:02.31 | citywok | leifmadsen: think old fashioned mud, but in audio instead of text |
| 19:02.38 | leifmadsen | neato |
| 19:02.39 | Lann | yeah |
| 19:02.42 | leifmadsen | someone already built that |
| 19:02.46 | Lann | !! |
| 19:02.48 | Lann | where? |
| 19:02.49 | citywok | really? lol |
| 19:02.59 | Lann | if it was done i might not do it |
| 19:03.05 | Lann | i want to play it just as bad as build it |
| 19:03.48 | leifmadsen | http://swik.net/Simon-P.-Ditner |
| 19:03.52 | leifmadsen | 2006 |
| 19:03.57 | leifmadsen | Zork via Asterisk |
| 19:04.10 | citywok | i'm listening to phone calls right now b/c of reports of bad call qualtiy. if this agent says UMM one more time i might go smack the crap out of her. atl east 20 times in the lsat 130 seconds |
| 19:04.26 | russellb | http://unitycode.org/read/ZoIP |
| 19:05.25 | leifmadsen | nice |
| 19:05.38 | Lann | hmm i'll try it |
| 19:06.30 | Lann | from what i read it requires dialpad input though and doesnt seem to be mutiplayer |
| 19:09.03 | Lann | very nice though |
| 19:09.08 | Lann | looking for a phone number of an install |
| 19:09.42 | Lann | ah, bottom of the page |
| 19:12.03 | Lann | he shares source even |
| 19:12.05 | Lann | happy dance |
| 19:13.06 | Tozz_ | I have 2 phones, (same type, same config, asterisk same config) but at 2 locations. From LocA my featuremap (record macro) works. From LocB it does not. I thought ti did not read DTMF properly, but voicemail system works okay |
| 19:13.10 | Tozz_ | what could that be? |
| 19:13.15 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v010-199.mobile.uci.edu) |
| 19:13.16 | *** join/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 19:14.27 | Lann | Anyhow, it's less crazy than my current project: www.minus-reality.com |
| 19:15.05 | *** join/#asterisk alphaz (~alphaz@209-207-94-113.ip.van.radiant.net) |
| 19:15.51 | Docfxit | I'm trying to setup FTP to Ubuntu. It won't connect. I'm getting an error saying "Unable to open connection to server" Could I please get some help in resolving this? |
| 19:16.03 | Qwell | Docfxit: #ubuntu? |
| 19:16.51 | carrar | ubuntu doesn't support FTP |
| 19:16.59 | alphaz | Hey all, i have a question... i have 2 sip servers talking to eachother.. when i receive incoming calls, the re-invite seems to work, as the main server receiving from provider uses 0.3kbps. so clearly rtp has been directly connected to the other server,, but when dialing out.. the slave server seems to always send all rtp through the main server... any idea why this would happen? |
| 19:17.13 | Docfxit | Sorry I'm on the wrong IRC. |
| 19:17.25 | *** part/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 19:17.41 | *** join/#asterisk mayfield (~mayfaild@76-250-152-203.lightspeed.snantx.sbcglobal.net) |
| 19:19.32 | *** join/#asterisk jshriver (~jshriver@72.240.39.37) |
| 19:19.36 | jshriver | greeting |
| 19:19.54 | jshriver | anyone know what would cause the call log to not show origin of call? |
| 19:21.47 | *** join/#asterisk sanman (~aaron@ip-216-93-249-210.twdx.net) |
| 19:21.55 | sanman | hello |
| 19:22.21 | sanman | this may be a little off topic for this channel, but does anyone know of a linux soft phone that supports the cisco sccp protocol? |
| 19:22.41 | Qwell | none |
| 19:22.48 | Qwell | use SIP |
| 19:23.17 | sanman | my company is deploying unified communications, but using sccp |
| 19:23.18 | alphaz | not sure why anyone would ever want to connect a sccp softphone :\ |
| 19:23.25 | sanman | I have a linux desktop |
| 19:23.25 | alphaz | uhh |
| 19:23.30 | alphaz | doesnt cisco have a sccp client |
| 19:23.36 | Tozz_ | buy a phone ;) |
| 19:23.37 | Qwell | for Windows |
| 19:24.03 | sanman | I guess I can run asterisk locally and that can talk sccp to the UM servers |
| 19:24.09 | Qwell | nope |
| 19:24.12 | Qwell | Asterisk is server-only |
| 19:24.25 | sanman | oh |
| 19:24.32 | sanman | well |
| 19:24.39 | sanman | I guess I'm screwed then |
| 19:24.57 | Tozz_ | doesnt Cisco UM support SIP these days? The Cisco 7900 series are offered with SIP |
| 19:25.09 | sanman | I'm sure they probably do |
| 19:25.27 | sanman | but our network team here really gets off on all of the cisco proprietary protocols |
| 19:25.37 | Tozz_ | ah:) |
| 19:26.01 | alphaz | lalala |
| 19:26.06 | sanman | I've had it out with them before over etherchannel (cisco only), instead of 802.3ad |
| 19:26.40 | sanman | but I'm on the SAN team and have no say in the ethernet world :) |
| 19:28.59 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 19:34.35 | *** part/#asterisk sanman (~aaron@ip-216-93-249-210.twdx.net) |
| 19:37.30 | alphaz | haha. sans rock |
| 19:57.46 | *** part/#asterisk cusco (~tralala@49.192.54.77.rev.vodafone.pt) |
| 19:58.46 | *** join/#asterisk atommat (~atommat@atommat.plus.com) |
| 19:59.42 | *** join/#asterisk RypPn (~TuMbL@unaffiliated/ryppn) |
| 20:03.32 | atommat | thought id put this to you guys to see if youd encountered this before... i have been in touch with digium about the card. the card has been proved to be ok with pattern tests etc. but when i dial in to my asterisk server its spread across two spans. each call should be on the a pair of b channels on one span and not on two b channels on different spans. this could be a line problem and i have contact the telco but i thought id investigate it just in c |
| 20:03.32 | atommat | here is the inbound call coming through http://www.pastie.org/1461306 |
| 20:03.42 | atommat | its a b410p |
| 20:04.16 | atommat | can this behaviour be configured out |
| 20:04.19 | atommat | ? |
| 20:04.46 | *** join/#asterisk leroybuckingham (~PC@adsl-99-124-84-62.dsl.milwwi.sbcglobal.net) |
| 20:07.07 | atommat | dahdi conf: http://www.pastie.org/1461335 |
| 20:07.18 | leroybuckingham | Hi everyone, I've noticed the CLI complains when I send a sip notify. I receive the message "parse error: No category context for line 1 of /etc/asterisk/extconfig.conf". That conf file is one line, it reads "meetme => mysql,asterisk,meetme" |
| 20:07.41 | leifmadsen | leroybuckingham: you don't have a [general] header |
| 20:07.51 | leifmadsen | leroybuckingham: look at the extconfig.conf.sample file |
| 20:08.00 | leroybuckingham | Ah, that makes perfect sense I should have noticed |
| 20:08.01 | leroybuckingham | thanks |
| 20:08.03 | leifmadsen | all files require at least one context name |
| 20:08.13 | leifmadsen | which is usually [general] |
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| 20:30.41 | jshriver | anyone know what would cause the call log to not show origin of call? |
| 20:31.40 | jshriver | or are there alternative logs beside cdr-csv? |
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| 20:34.49 | jsidhu | Can someone help me debug why my iax2 trunk please. I keep getting a "Registration of 'B002_B020' rejected: '<unknown>' from: <ip_removed>" i checked for typos and everything looks allright.. iax2 set debug on doesnt give out much information.. any other way to get more verbose output and track down whats broken? |
| 20:40.27 | *** join/#asterisk matiasjrossi (~matias@186.109.85.23) |
| 20:41.25 | atommat | is there a situation where a call can be picked up by two channels? |
| 20:43.41 | matiasjrossi | hi. I'm trying to manual update the status of a sip peer (reachable/unreachable/registered, etc) by issuing a "SIPqualifypeer" through AMI. The peer has qualify enabled, but when the qualify is manually issued, the status is not updated. Does anybody know why my qualifies aren't considered by asterisk? |
| 20:46.09 | Tozz_ | why would u want to do that? |
| 20:47.51 | matiasjrossi | long story :) |
| 20:47.59 | matiasjrossi | basically a buggy mobile sip client |
| 20:48.57 | citywok | set the qualify really low |
| 20:49.32 | matiasjrossi | will try. thank you! |
| 20:49.52 | citywok | look up qualifyfreq i think |
| 20:51.55 | leifmadsen | atommat: no |
| 20:52.19 | atommat | thanks |
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| 21:00.47 | jamko | Good afternoon. So I constantly have issues with post dial delay, different carriers, random numbers. So I am thinking of just sending ring tones to the phones no matter what. It looks like the, r option for the dial command is what does this, however |
| 21:01.03 | jamko | I also see the RINGING command. Any thoughts on the better approach? |
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| 21:20.02 | citywok | jamko: whatever works :P |
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| 21:32.23 | frek818 | hi |
| 21:34.24 | frek818 | I have and issue where I set the session-timers option to refuse I notice that asterisk send adds 'timer' to the 'Support' header on the Invite when I place outbound calls using my SIP trunk. |
| 21:35.14 | frek818 | I would expect that Asterisk would not proclaim that timers are supported if I set the session-timers options to refuse. |
| 21:36.01 | frek818 | I put the 'session-timers=refuse' in the general section of sip.conf |
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| 21:40.25 | pabelanger | frek818: There was a patch on https://reviewboard.asterisk.org that addressed that issue, if I remember right. |
| 21:40.31 | pabelanger | Not sure if it was merged |
| 21:43.36 | pabelanger | https://reviewboard.asterisk.org/r/698/ |
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| 22:07.57 | frek818 | pabelanger: thanks |
| 22:11.00 | carrar | reviewboard.asterisk.org ipv6 ip unreachable |
| 22:12.00 | carrar | ipv4 ip works fine |
| 22:22.03 | pabelanger | carrar: I can confirm |
| 22:24.08 | carrar | That site isn't the only place that has IPv6 miss configured or not working, makes using IPv6 fustrating |
| 22:29.28 | antiwire | hey guys, anyone know what is going on here with dahdi / wanpipe driver? http://pastebin.slackadelic.com/p/yq5BvS16.html |
| 22:31.59 | Qwell | antiwire: its broken |
| 22:32.07 | antiwire | is there a patch to fix that? |
| 22:32.11 | Qwell | ask sangoma |
| 22:32.20 | Qwell | its closed source. nothing we can do |
| 22:33.53 | carrar | ... reasons to use digium |
| 22:39.00 | antiwire | digium makes a US FXO adapter? |
| 22:39.15 | antiwire | USB |
| 22:39.48 | pabelanger | carrar: should be back up |
| 22:44.29 | pabelanger | Anybody know if you can control the CallerID display on a polycom 331? Right now my phone is displaying <6135551234> 6135551234. I only need one, not both fields |
| 22:47.37 | carrar | yup |
| 22:47.39 | carrar | w00t |
| 22:48.07 | russellb | antiwire: no. We did many many years ago, but not anymore |
| 22:48.16 | Qwell | that was FXS wasnt it? |
| 22:48.18 | *** join/#asterisk lamikr (~lamikr@dyn60-31.dsl.spy.dnainternet.fi) |
| 22:48.22 | russellb | oh, right |
| 22:48.25 | WIMPy | pabelanger: Have you tried to set name-pres to restricted? |
| 22:51.58 | pabelanger | WIMPy: no, giving it a shot now |
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| 23:01.36 | lamikr | Hi, I am trying to understand how to get support for SIP messages I send for examply from empathy to work with asterisk...From wireshark dump I can see the format: |
| 23:01.40 | lamikr | MESSAGE sip:202@192.168.3.30 SIP/2.0 |
| 23:01.41 | lamikr | ... |
| 23:01.54 | lamikr | Content-Type: text/plain |
| 23:01.55 | lamikr | Content-Length: 15 |
| 23:01.55 | lamikr | hello 201 calls |
| 23:02.18 | lamikr | But then asterisk responds with error: |
| 23:02.30 | lamikr | IP/2.0 405 Method Not Allowed |
| 23:03.30 | WIMPy | Please use a pastebin next time. |
| 23:03.57 | lamikr | WIMPy ok... sorry for that |
| 23:04.42 | lamikr | Anyway I googled and found only a 5 year old patch for supporting it from http://www.voip-info.org/wiki/index.php?page_id=2419 and http://juraj.bednar.sk/work/software/asterisk/messaging/ |
| 23:05.59 | lamikr | And according to comments that patch was removed from the voip-info org as a obsolete... Is there now some other way howto configure asterisk to support this? |
| 23:09.11 | lamikr | I have seen some suggestions saying that messaging should work with asterisk once you have first created a call, but I feel it's not a good restriction because some other servers allows 2 empathy clients to chat with sip messages also without first starting a phone call. (And even that seemed to fail when I tested). |
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| 23:47.28 | alphaz | hopefully someone can help.. i have 1 mailbox.. and 3 symlinks to that one mailbox for other extensions.. when i get mail, the MWI message indicator light goes on, on all the phones.. but when i read the message on a phone..the light only turns off for that one phone but stays lit on the rest of the phones.. any ideas? |
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| 23:49.23 | WIMPy | You should just use the same mailbox and not create symlinks. |
| 23:49.57 | alphaz | ok i'll try |
| 23:54.09 | alphaz | thx wimpy |
| 23:54.11 | alphaz | that worked |
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| 23:56.46 | alphaz | ah |
| 23:56.51 | alphaz | i remember why i ddint do that now |
| 23:56.51 | alphaz | :\ |
| 23:56.58 | alphaz | ok well a combination of it works i guess |
| 23:57.43 | alphaz | cuz.. asterisk when you click msg button on cisco phone it calls say *97.. and by default it will try to log in with the mailbox with the extention.. |
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