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| 00:21.15 | Aut0Exec | hi guys... I just got msyelf a cisco spa3102.. how do I set the dialplan to still make my local calls? |
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| 00:44.36 | TeknoJuce | Just got inbound calling working with gv and the Nortel i2004 IP Phone w00t!! |
| 00:44.55 | TeknoJuce | my prob was i had sip/1337 |
| 00:45.30 | TeknoJuce | but what you need for unistim is USTM/1337@black |
| 00:45.34 | TeknoJuce | etc |
| 00:45.56 | TeknoJuce | now its working amazing! |
| 00:48.49 | elb | :-) |
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| 00:51.17 | Aut0Exec | how to get calls to go over pstn line? |
| 00:51.27 | Aut0Exec | how do I set this in my dialplan? |
| 00:51.37 | elb | we can't answer that question without a lot more information |
| 00:52.30 | elb | if your cisco device is SIP, you probably just dial it with SIP/number@deviceaddr or the like |
| 00:52.35 | elb | but ... you need to look at its documentation |
| 00:52.43 | Aut0Exec | elb: I just set up my spa3102 and I created entries in sip.conf.. |
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| 00:53.44 | Aut0Exec | and I created a dialplan... I can get calls going out to my voip but I cant call local lines... |
| 00:53.57 | Aut0Exec | where do I direct my "Dial()" to make the local call? |
| 00:54.15 | elb | is the spa3102 your local pstn connection? |
| 00:54.18 | elb | if os, the spa3102 |
| 00:54.20 | elb | if so |
| 00:54.30 | elb | but ... you'll need to read its documentation to see how |
| 00:54.35 | ruben23 | hi guys any idea when i cal on a voip i get this error on my call----> Got SIP response 503 "Max QPS rate exceeded |
| 00:55.22 | Aut0Exec | elb: ok |
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| 00:58.50 | gruvfunk | Aut0Exec -> on the 3102, look for: VoIP-To-PSTN Gateway Enable: "Yes", and then you'll also need to adjust your DialPlan |
| 00:59.37 | Aut0Exec | gruvfunk: ok thanks dude... I have voip-to-pstn gateway enabled... can you give me an examle of a dialplan I would use tho? |
| 01:02.42 | gruvfunk | Aut0Exec: Highly dependent on your configuration, your local area, and how you wish to access the PSTN gateway (dial 9, or auto-recognition based on local area, etc) |
| 01:02.55 | gruvfunk | get started here http://www.cisco.com/en/US/products/ps10024/products_qanda_item09186a0080a359c4.shtml |
| 01:03.44 | Aut0Exec | gruvfunk: hrumph ..yeah I know how to make dialplans... just not sure what to dial it with |
| 01:04.00 | Aut0Exec | meaning... how to dial a local pstline |
| 01:04.07 | Aut0Exec | using a dialplan |
| 01:04.17 | Aut0Exec | like what goes in (Dial.... |
| 01:04.31 | Aut0Exec | based on my configs in my ATA |
| 01:04.42 | Aut0Exec | i have a Sip config called "line1" |
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| 01:06.49 | gruvfunk | Aut0Exec: describe your environment for us, give us details and maybe we can help |
| 01:08.32 | Aut0Exec | gruvfunk: basic 192.168.... network... 1 pstn line... cisco spa3102 with 1 fxo 1 fxs ... I need to make calls going over pstn line and voip line... right now I have voip working... I can call a voip number fine... its just local pstn numbers I cant dial solely because I dont know what to put for the dialplan entry |
| 01:09.06 | Aut0Exec | so lets say I want to dial local numbers by using a 9 |
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| 01:09.13 | msetim | hello guys |
| 01:09.39 | Aut0Exec | gruvfunk: my line1 is called "line1" in sip and fxs port is called "homephone" |
| 01:10.37 | gruvfunk | do you have Asterisk in the mix? do you have any voip phones? are you simply using an analog phone in the FXS, with PSTN line on FXO? |
| 01:15.20 | gruvfunk | Aut0Exec: try <9:><:@gw0> |
| 01:15.47 | Aut0Exec | gruvfunk: is that the settings for my ATA or for the asterisk dialplan? |
| 01:15.57 | gruvfunk | 3102 ATA |
| 01:16.03 | Aut0Exec | k |
| 01:16.18 | gruvfunk | remember that you'll now have to edit 2 dialplans with Asterisk + 3102 |
| 01:16.40 | Aut0Exec | k |
| 01:16.43 | gruvfunk | so you'll likely need to make appropriations for dialing 9 -> out in Asterisk |
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| 01:18.07 | gruvfunk | Aut0Exec -> maybe it's <9:>S0<:@gw0> |
| 01:18.12 | Aut0Exec | ok |
| 01:19.12 | Freeaqingme | Is there any specific reason why one wouldn't put his entire dialplan in a AGI app? |
| 01:19.28 | gruvfunk | Freeaqinme: great point... |
| 01:19.49 | Freeaqingme | that was not in response to any of this conversation per se |
| 01:19.59 | gruvfunk | never have to reload the dialplan or asterisk, when you make changes! |
| 01:20.56 | Freeaqingme | yup, I see a lot of advantages, but wondering if there are also any disadvantages |
| 01:21.25 | gruvfunk | hopefully those smarter than me can weight in |
| 01:30.11 | Maxus2 | hi people, im connecting another asterisk box to my asterisk box with iax2 using regcontext, but the dial plan never seems to be updated once the box is registered? |
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| 02:27.08 | phix | hi |
| 02:27.20 | gruvfunk | what up phix |
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| 02:28.41 | phix | having issues with either asterisk, dahdi, iaxmodem or hylafax |
| 02:30.10 | phix | i finally got fax detection, i had to raisr wait time to 6seconds instead of 4 |
| 02:30.30 | phix | now it is having issues "training the modem" |
| 02:31.10 | phix | i also dont have a timing source now for some reason, it all worked with zaotel :/ |
| 02:32.27 | phix | i will paste some logs and conf filea in a sec. |
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| 03:11.40 | juliocesarlhg | hi |
| 03:11.59 | juliocesarlhg | someone who uses mp202 audiocode? |
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| 03:34.52 | joako | Does anyone know if asterisk14-voicemail-imapstorage |
| 03:34.52 | joako | <PROTECTED> |
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| 06:25.48 | schmidts | good morning |
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| 07:03.37 | wdoekes2 | you're up early, stefan |
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| 07:25.51 | schmidts | and it was a VERY short night, get in bed at 03:15 and had to woke up at 06:00 :( |
| 07:26.14 | schmidts | but now my hosted pbx has a kamailio proxy in front \o/ |
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| 07:47.49 | verywiseman | can each GSM gateway send sms , or must i use sms gateway? |
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| 08:51.58 | m_tadeu | I don't have rtp packets moving around when the call is answered, after the caller waits in a queue. what can be the problem? both are sip, the caller outside nat and the callee inside |
| 08:52.51 | m_tadeu | running asterisk 1.6.2.7 |
| 08:53.21 | m_tadeu | Freeaqingme: sorry, yesterday couldn't reply, 'cos I had a network problem |
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| 09:33.13 | Dovid | anyone know why the cdr sql set up file is not in asterisk source and where can i get it ? |
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| 09:52.10 | joobie | hey guys.. can anyone recommend an appliance type device that can be used to load your own version of linux on it and has a few network ports? Trying to find a device i can use to build a small firewall device with a few extra smarts (hoping to keep it in an appliance type form with a few NIC's on it) |
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| 09:56.46 | kaldemar | joobie: http://wiki.openwrt.org/toh/start |
| 09:58.41 | kaldemar | Dovid: which file? |
| 09:59.01 | kaldemar | Dovid: and by "set up file", do you mean a configuration file? |
| 09:59.30 | joobie | kaldemar, yea been looking at those linksys modems.. but tyring to find something that is more purpose built for this kind of thing |
| 10:00.31 | kaldemar | joobie: most (if not all) of those devices are routers, not modems. |
| 10:01.13 | kaldemar | and built for router and a firewall usage. |
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| 10:13.47 | skrusty | morning all |
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| 10:20.21 | meri | salut |
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| 10:26.25 | m_tadeu | g'morning |
| 10:27.09 | m_tadeu | sometimes I get a stuck call in the queue....the call is already off but it's still there, somehow |
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| 10:31.57 | jg123 | hi |
| 10:33.02 | jg123 | i have a problem with dahdi_monitor, it seems to be losing|dropping samples |
| 10:33.17 | jg123 | is there anything i can do to prevent that ? |
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| 11:18.59 | tzafrir_laptop | jg123, how can you tell it is dropping packets? |
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| 11:35.25 | m_tadeu | how can I remove a call from a queue? |
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| 11:38.19 | Tozz_ | soft hangup |
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| 11:44.35 | kaldemar | or channel hangup in newer versions. |
| 11:45.04 | kaldemar | erm, channel request hangup that is. |
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| 11:51.03 | momo1 | hello |
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| 11:53.50 | momo1 | i want to add security patches for asterisk from http://www.asterisk.org/security |
| 11:54.15 | momo1 | but the file is .diff |
| 11:54.26 | momo1 | not .patch |
| 11:55.53 | kaldemar | yes? |
| 11:56.56 | momo1 | i dont know how to do |
| 11:57.40 | kaldemar | how are you trying to patch your asterisk? |
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| 12:06.39 | jg123 | tzafrir_laptop: sorry i dozed off a bit |
| 12:07.12 | jg123 | i can tell by looking at the data (plot) |
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| 12:07.53 | jg123 | and by comparing it to the date recorded with Monitor() |
| 12:09.11 | jg123 | i should see a continuous sine but i see many phase shifts |
| 12:11.39 | jg123 | and my only explanation to this is that dahdi_monitor seems to be dropping samples |
| 12:14.17 | m_tadeu | what if after 'hangup request some_sip_channel' the call stays on queue? |
| 12:14.24 | IridiumScaffold | does anyone know why using Originate with a grandstream phone would cause all the lines on it to ring? |
| 12:16.54 | tzafrir_laptop | jg123, is the server busy? |
| 12:17.16 | tzafrir_laptop | dahdi_monitor uses a dahdi-level conference to record the call |
| 12:17.51 | jg123 | its not very busy |
| 12:18.36 | jg123 | i forced it so be busy, but then the dropping did not occur |
| 12:18.55 | jg123 | it seems to be random |
| 12:19.31 | jg123 | oh yeah i might have to add the same thing occurs with DAHDIBarge |
| 12:19.37 | *** join/#asterisk drcode (~user1@bzq-84-111-89-77.red.bezeqint.net) |
| 12:19.43 | drcode | hi all |
| 12:20.06 | drcode | can two users on same callerid can talk to each other? |
| 12:20.36 | drcode | or each must have two diffrent callerid ? and If they login with same user and password? |
| 12:20.37 | WIMPy | Noone cares about caller ID. |
| 12:20.55 | jg123 | but i guess thats using the same conference technique |
| 12:21.48 | kaldemar | m_tadeu: the command is "channel request hangup some_sip_channel" |
| 12:21.51 | drcode | and login? |
| 12:22.45 | m_tadeu | kaldemar: in deed |
| 12:23.50 | kaldemar | drcode: 1. a caller id can be whatever you set it to be. 2. there is no login. 3. use a single device (in sip.conf) for a single user, otherwise asterisk can't know who to contact when it tries to dial the device. |
| 12:24.12 | drcode | I see |
| 12:24.21 | drcode | so I must use two diffrent users? |
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| 12:25.25 | kaldemar | drcode: caller id has really nothing whatsoever to do with sip authentication or even extension numbers. |
| 12:25.40 | drcode | ok |
| 12:25.43 | drcode | thanx kaldemar |
| 12:26.26 | IridiumScaffold | what's the difference between "originate"ing a call and using a call file. I'm supplying the same parameters to each, but "originate" causes all the lines on my grandstream phones to light up |
| 12:28.46 | kaldemar | IridiumScaffold: see sip debug to find a difference. there shouldn't be any difference from a phone's point of view if you really use all the same parameters. |
| 12:28.52 | kaldemar | then again... |
| 12:28.54 | kaldemar | ~gs |
| 12:28.54 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 12:29.09 | IridiumScaffold | not my purchasing decision. |
| 12:29.59 | m_tadeu | I have a caller in the queue, which was answered by an agent, hangup after a while, but the caller is still on the queue |
| 12:33.38 | *** join/#asterisk ccesario_ (~ccesario@189-29-59-116-ac.cpe.vivax.com.br) |
| 12:35.15 | *** join/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 12:38.26 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:39.09 | *** join/#asterisk momo1 (~mohamed@41.225.154.175) |
| 12:40.50 | m_tadeu | I'm unable to remove it from the queue |
| 12:48.23 | *** join/#asterisk sourcode (~code@ppp-58-8-122-95.revip2.asianet.co.th) |
| 12:48.57 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 12:48.57 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 12:49.18 | *** join/#asterisk m_tadeu (~quassel@static-b5-252-50.telepac.pt) |
| 12:53.32 | *** join/#asterisk momo1 (~mohamed@41.225.141.126) |
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| 13:01.40 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:08.36 | jg123 | tzafrir_laptop: or do you know another way to record "everything" |
| 13:08.56 | tzafrir_laptop | no |
| 13:14.47 | jg123 | and there is no "easy" way to make monitor record all the frames not just voice |
| 13:15.02 | IridiumScaffold | ok another question: has anyone got sip phones to work with a voicetronix card sending DTMF? |
| 13:30.12 | *** part/#asterisk IridiumScaffold (~Iridescen@124.169.157.159) |
| 13:30.56 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
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| 13:52.27 | *** join/#asterisk tuxx- (~tuxx@nightshade.nl) |
| 13:52.41 | tuxx- | hey guys, is there a similair command like 'sip debug' for dahdi? |
| 13:52.51 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 13:53.21 | WIMPy | pri set debug, if it's ISDN. |
| 13:53.49 | tuxx- | tnx :) |
| 13:58.01 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 14:02.28 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:04.16 | tuxx- | is there any possibility that i can strip the following debug line from a call? Ext: 1 Reason: Forwarded unconditionally (15) '340340' ] |
| 14:04.53 | tuxx- | This debug line is from an incomign line, and it also gets send to the outgiong line, but i dont want that to happen. |
| 14:05.09 | tuxx- | the outgoing line shouldnt know ANY call information except for the destination number |
| 14:06.01 | tuxx- | brb |
| 14:06.20 | *** join/#asterisk tuxx- (~tuxx@nightshade.nl) |
| 14:06.26 | tuxx- | back :_D |
| 14:06.57 | WIMPy | You could try to reset CALLERID(RDNIS). |
| 14:07.21 | WIMPy | But when I checked the other day, I had the impression that it's unused. |
| 14:08.01 | tuxx- | right, i already set the CALLERID(DNIS) to empty, but it still gets send |
| 14:08.06 | tuxx- | ill try the RDNIS also |
| 14:08.11 | WIMPy | So maybe it's only setting the reason, but AFAIK that's not accessible from the dialplan. |
| 14:12.52 | Freeaqingme_ | I'm proposing an Asterisk component for the Zend Framework (one of the most popular php frameworks around). If anybody would be willing to take a look at the proposal it would be highly appreciated: http://framework.zend.com/wiki/display/ZFPROP/Zend_Telephony_Asterisk+-+Dolf+Schimmel+(Freeaqingme) |
| 14:27.01 | Dovid | is it "legal" for a peer name to be a integer from 1=255 ? |
| 14:27.34 | russellb | technically, yes, but it's a bad idea from a security perspective |
| 14:28.56 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:29.17 | Dovid | russellb: 1) why is that ? 2) I am using asterisk 1.8.3.2. When I do say Dial(SIP/123456@20) asterisk shows: |
| 14:29.18 | Dovid | [2011-03-24 18:27:37] WARNING[7794]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x1b66000 (len 784) to 0.0.0.22:5060 returned -1: Invalid argument |
| 14:29.18 | Dovid | Retransmitting #6 (no NAT) to 0.0.0.22:5060: |
| 14:29.18 | Dovid | INVITE sip:18005551212@22 SIP/2.0 |
| 14:29.34 | Dovid | seems like it thinks 22 is a peer |
| 14:30.11 | kaldemar | Dovid: http://svn.digium.com/svn/asterisk/tags/1.8.3.2/README-SERIOUSLY.bestpractices.txt |
| 14:30.13 | russellb | it's 1 of 2 things ... 1) the host= setting for peer 20 is broken. 2) Asterisk is broken and is trying to interpret "20" as an IP address. |
| 14:30.40 | russellb | has to run |
| 14:30.44 | *** join/#asterisk rgenthner (~rgenthner@cpe-74-75-158-14.maine.res.rr.com) |
| 14:30.46 | russellb | good luck. |
| 14:30.55 | Dovid | russellb: I have a host there. seems like an asterisk bug. i will downgrade for now |
| 14:30.57 | rgenthner | can anyone help me with real tiem queues |
| 14:32.13 | Dovid | kaldemar: Thanks. this is a closed internal system. just "translating" PSTN calls to SIP. no cusotmer access |
| 14:32.21 | WIMPy | dovid: Use sip/peer/exten instead of sip/exten@peer. |
| 14:32.25 | kaldemar | Dovid: it's not a bug. see the sample sip conf for dial strings. you'll find "SIP/username@domain (SIP uri)" there which means that with that notation, asterisk expects @ to be followed by a host. |
| 14:32.27 | Dovid | rgenthner: Whats the issue ? |
| 14:32.57 | Dovid | kaldemar & WIMPy: Bad habbits I guess. been doing it this way 6+ years |
| 14:32.59 | rgenthner | Dovid: context for leaving the queue isn't working, i have set context = quotcon fails |
| 14:33.13 | rgenthner | i can press any button until i'm blue in the face |
| 14:33.24 | Dovid | you have a space there |
| 14:33.50 | Dovid | shouldnt it be context=quotconfails OR context=quotcon_fails OR context=quotcon-fails |
| 14:33.54 | rgenthner | yeah but i dont in my mysql table for the queue |
| 14:33.54 | Dovid | also could be a DTMF issue |
| 14:34.04 | Dovid | rgenthner: So add it ;) |
| 14:34.13 | rgenthner | sorry meant i have it |
| 14:34.33 | rgenthner | it seems like no key press will interrupt the hold music |
| 14:35.01 | Dovid | rgenthner: use DTMF=info. do a sip trace and see if the DTMF is coming in at all |
| 14:35.10 | rgenthner | k |
| 14:35.11 | rgenthner | thansk |
| 14:37.01 | kaldemar | no reason to change dtmfmode to see that. you'll see dtmf with rtp debug when using rfc2833. |
| 14:37.24 | Dovid | kaldemar: It's a lot harder to debug with rtp debug :) |
| 14:37.25 | rgenthner | the calls come in over a pri |
| 14:37.30 | rgenthner | would that make a difference |
| 14:37.44 | Dovid | rgenthner: Then you should be having less issues (in general) ;0 |
| 14:38.01 | Dovid | see if you have dtmf in general |
| 14:38.18 | Dovid | if you have dtmf there then post configs |
| 14:38.32 | rgenthner | how do i turn on the dtmf info in the console, i didn't see anything i core set for it |
| 14:39.35 | Dovid | rgenthner: That was just for sip. line an ass i assumed u were using sip |
| 14:39.48 | Dovid | on a pri it comes in inband |
| 14:40.30 | rgenthner | k |
| 14:40.51 | rgenthner | i have a test queue that i just put on my sip trunk |
| 14:43.17 | Dovid | kaldemar: Doing what you said does not work either |
| 14:43.40 | Dovid | Dial(SIP/20/18005551212) asterisk still tries to send the call to @0.0.0.20\ |
| 14:44.19 | WIMPy | That looks like a bug then. |
| 14:44.40 | WIMPy | But it's definitely an unfortunate name as well. |
| 14:45.45 | Dovid | WIMPy: I agree. |
| 14:50.21 | *** join/#asterisk TSM (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
| 14:51.03 | wdoekes2 | in that case my names are unfortunate too.. $ ping 123450001 |
| 14:51.04 | wdoekes2 | PING 123450001 (7.91.178.145) |
| 14:54.40 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 14:55.14 | Aut0ExeC | hi guys... anyone here have a spa3102? |
| 14:55.23 | Aut0ExeC | I need a lil help with dialplan on the device |
| 14:55.41 | Aut0ExeC | I'm not understanding what "SO<..." stands for |
| 14:55.53 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:55.58 | Aut0ExeC | I now understand what ...@gw0 stand for after I read the docs |
| 14:56.49 | schmidts | Aut0ExeC S0 means dial immedialtly if this part of the dialplan match |
| 14:57.08 | Aut0ExeC | I see... weird I didnt see that in the docs |
| 14:57.19 | Aut0ExeC | and @gw0 is just tied to the pstn line right? |
| 14:57.32 | Aut0ExeC | so dialplans will be directed to that |
| 14:57.37 | Aut0ExeC | if need to make pstn calls |
| 14:58.04 | schmidts | Aut0ExeC i dont know what @gw0 means but it sounds logical right ;) |
| 14:58.13 | Aut0ExeC | lol ok |
| 14:58.32 | psilikon | Aut0ExeC, yeah I have a couple of spa3102s |
| 14:58.51 | Aut0ExeC | last question... if I have an asterisk server... would I just pass the whole dialplan to asterisk and let it do the routing? |
| 14:59.06 | Aut0ExeC | or specify in the device dialplan |
| 14:59.20 | Aut0ExeC | psilikon: maybe you can help me with that please |
| 14:59.23 | Aut0ExeC | :) |
| 15:00.25 | Aut0ExeC | psilikon: i've seen some internet docs that suggest using something like <123@yourasteriskserveriphere> |
| 15:00.37 | Aut0ExeC | psilikon: then create a dialplan in extensions.conf for 123 |
| 15:00.44 | Aut0ExeC | that makes sense? |
| 15:00.44 | psilikon | Aut0ExeC, yes I would let * do as much as possible. Those spa's have a ton of features if you are just using them with an ITSP and no pbx. Asterisk trumps everything a spa can do. |
| 15:01.05 | Aut0ExeC | psilikon: ok |
| 15:01.12 | schmidts | psilikon they can T38 :D :D |
| 15:01.38 | Aut0ExeC | whats T38? |
| 15:01.38 | schmidts | dont mind Aut0ExeC ;) |
| 15:01.38 | Aut0ExeC | lol |
| 15:01.42 | Aut0ExeC | ok |
| 15:01.50 | psilikon | I just used a real generic dialplan i the ATA: (*xx|xxx|xxxx|xxxxx|xxxxxx|xxxxxxx|xxxxxxxx|xxxxxxxxxxx) |
| 15:01.54 | schmidts | its an open wound for asterisk |
| 15:02.02 | psilikon | Then I do all the fun stuff inside Asterisk. |
| 15:02.25 | Aut0ExeC | psilikon: what does your dialplan for pstn line look like? |
| 15:02.49 | Aut0ExeC | psilikon: inside Asterisk. |
| 15:02.49 | Aut0ExeC | 14:51 < Aut0ExeC> psilikon: what does your dialplan for pstn line look like? |
| 15:02.55 | Aut0ExeC | oops |
| 15:03.02 | Aut0ExeC | <PROTECTED> |
| 15:03.03 | Aut0ExeC | 14:52 < Aut0ExeC> oops |
| 15:03.03 | Aut0ExeC | >> (14:52) (Aut0ExeC[+i]) (3:freenode/#asterisk[+cnrt]) |
| 15:03.06 | Aut0ExeC | <PROTECTED> |
| 15:03.59 | Aut0ExeC | psilikon: do you have something like exten => _XXXXXXX,1,Dial(Sip/${EXTEN}@pstn) ?? |
| 15:04.11 | psilikon | Aut0ExeC, I have a seperate context for from pstn and to pstn. Which one you want to see? |
| 15:04.24 | Aut0ExeC | psilikon: can I see both please |
| 15:04.59 | psilikon | Aut0ExeC, Well all my ata's are in and "internal" context so anytime 7,10 or 11 digits are dialed it gets routed out to my ITSP |
| 15:05.32 | Aut0ExeC | ok |
| 15:05.41 | psilikon | http://pastie.org/1708444 |
| 15:05.48 | Aut0ExeC | thanks bud |
| 15:05.51 | psilikon | np |
| 15:06.36 | psilikon | It is good to factory defualt a SPA and install the latest firmware. |
| 15:06.48 | Aut0ExeC | ok I didnt do that .. I will do that tho thanks |
| 15:07.57 | Aut0ExeC | you have a sip trunk for voipms? |
| 15:08.10 | Aut0ExeC | i'm a nub at this.. please excuse me |
| 15:08.35 | *** join/#asterisk atom519 (~chatzilla@adsl-216-63-18-203.dsl.kscymo.swbell.net) |
| 15:08.38 | WIMPy | ~siptrunk |
| 15:08.38 | infobot | rumour has it, siptrunk is something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 15:09.41 | psilikon | Aut0ExeC, yeah. I actually really like voip.ms btw |
| 15:09.44 | Aut0ExeC | WIMPy: lol did you not see the "nub" part |
| 15:10.16 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 15:10.21 | atom519 | I'm having trouble setting up Switchvox faxprinter on thin clients, wasn't sure if anyone has had luck with it. |
| 15:10.27 | Aut0ExeC | psilikon: thanks again bro |
| 15:10.38 | WIMPy | did, but thought it might fit. |
| 15:13.13 | Aut0ExeC | is it really beneficial to put "SO" in front of ur dialplan? |
| 15:14.35 | psilikon | http://forums.whirlpool.net.au/archive/332476 |
| 15:14.55 | psilikon | Aut0ExeC, SPA dialplan link. You use the S0 when you need it. |
| 15:15.02 | atom519 | is it possible to access the faxprinter via UNC share as opposed to http:// ? |
| 15:15.15 | Aut0ExeC | awesome thanks |
| 15:15.33 | atom519 | or is SMB just not possible with switchvox |
| 15:18.02 | psilikon | Aut0ExeC, there is a lot of really good information out there on linksys/cisco (formerly Sipura) ATAs. I love em |
| 15:21.12 | Aut0ExeC | yeah I just bought one but I'm a nub at aTAs & asterisk |
| 15:21.14 | Aut0ExeC | :) |
| 15:21.32 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 15:21.34 | Aut0ExeC | you make calls strictly to viop or over ur pstn line as well? |
| 15:21.45 | *** join/#asterisk sahafeez (~sahafeez@99-105-56-250.lightspeed.sntcca.sbcglobal.net) |
| 15:21.57 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 15:23.53 | psilikon | Aut0ExeC, if you are asking if I have a fxo/fxs interface to my asterisk box at home then no. |
| 15:24.41 | psilikon | Aut0ExeC, strictly IP at the house; At work we use SIP and Sangoma T1 cards that interface with the PSTN via PRIs. |
| 15:24.48 | Aut0ExeC | psilikon: ok cuz my next question was gonna be if you had any echo over ur pstn line .. lol... i had this a bit yesterday when I was testin a few things. and I heard this is a little problem withthis model |
| 15:30.38 | *** join/#asterisk outtolunc (~outtolunc@204.sub-75-208-52.myvzw.com) |
| 15:31.41 | Aut0ExeC | with digium cards... is there a web interface to configure the ports? |
| 15:31.55 | Aut0ExeC | how do they get configured? |
| 15:35.16 | Nugget | with vi or emacs and a steady hand. |
| 15:35.36 | jg123 | did someone else ever had any problems with dahdi_monitor recordings ? |
| 15:35.42 | outtolunc | Hold still damnit |
| 15:35.43 | *** join/#asterisk philippel_mac (~p_lindhei@50-46-123-25.evrt.wa.frontiernet.net) |
| 15:35.48 | Aut0ExeC | Nugget: lol wtf? |
| 15:35.58 | Aut0ExeC | I prefer vim actually |
| 15:36.12 | Nugget | The digium cards are configured via /etc |
| 15:36.18 | Aut0ExeC | ahh nice |
| 15:36.22 | philippel_mac | question - if multiple M(somemacro) options are added to a dial command, which one will be run, the first or last? |
| 15:36.29 | Nugget | not really, to be honest, but that's how they work. :) |
| 15:36.31 | Aut0ExeC | Nugget: I need to get one of those |
| 15:36.37 | Aut0ExeC | lol |
| 15:36.43 | philippel_mac | And - has this been consistent between Asterisk releases or ? |
| 15:36.45 | Aut0ExeC | well u prefer web interface? |
| 15:36.58 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 15:37.11 | Aut0ExeC | how else u gonna config a device with no interface I guess... |
| 15:37.16 | Aut0ExeC | has to be done the dirty way |
| 15:37.18 | ollii | hi |
| 15:39.48 | WIMPy | Aut0ExeC: dahdi_genconf |
| 15:40.05 | Aut0ExeC | i guess Nugget doesnt know about that then :) |
| 15:40.10 | Aut0ExeC | I'm a nub so I'm new to all of this |
| 15:40.34 | Aut0ExeC | I'm just messing around wtih spa ATA |
| 15:40.38 | WIMPy | It's not web. |
| 15:40.43 | Aut0ExeC | k |
| 15:40.43 | Nugget | Don't know about what? |
| 15:40.52 | Aut0ExeC | lol what WIMPy said |
| 15:40.56 | Aut0ExeC | dahdi_genconf |
| 15:41.32 | *** join/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 15:41.34 | Nugget | Of course I know about that. I've got several digium cards. |
| 15:42.10 | Aut0ExeC | k |
| 15:42.43 | Aut0ExeC | Nugget: u do telephony as a career? |
| 15:42.48 | Aut0ExeC | or just hobby? |
| 15:42.52 | Nugget | Neither. |
| 15:42.57 | Aut0ExeC | oh |
| 15:42.58 | Aut0ExeC | ok |
| 15:43.49 | Docfxit | When I enter sudo asterisk -vvvvvvr I get an error saying: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) The file is there with zero bytes. What could the problem be? |
| 15:44.57 | psilikon | Docfxit, still with the same thing? Did you reboot the box yet? |
| 15:45.23 | Docfxit | psilikon yes many times. |
| 15:45.51 | psilikon | Docfxit, Did you install asterisk from source? Or via Apt? |
| 15:46.39 | Docfxit | psilikon from the website download. |
| 15:47.37 | psilikon | Docfxit, good. What version of *? |
| 15:47.44 | Docfxit | psilikon It has been running just fine for a while. This cropped up after. |
| 15:48.12 | Docfxit | psilikon 1.4.22, |
| 15:48.17 | psilikon | what about typing "service asterisk restart" |
| 15:49.16 | Docfxit | psilikon Service command not found |
| 15:49.32 | psilikon | Docfxit, so what version of Ubuntu? |
| 15:49.55 | *** join/#asterisk garymc (~chatzilla@host81-148-15-59.in-addr.btopenworld.com) |
| 15:49.58 | Docfxit | psilikon 7.1 |
| 15:50.24 | psilikon | Docfxit, oh, try" /etc/init.d/asterisk restart" |
| 15:50.54 | Docfxit | psilikon I'm planning on updating both of them soon. I just need to get this going now. |
| 15:53.13 | Docfxit | psilikon I ran the restart. It didn't produce any errors. I'm getting the same not found error. |
| 15:55.25 | psilikon | Docfxit, what about '/etc/init.d/asterisk start'? |
| 15:59.08 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 15:59.43 | Aut0ExeC | psilikon: hey man... sorry to bother again... last time.. can I see a copy of your incoming dialplan? |
| 15:59.55 | Aut0ExeC | psilikon: calls that come into ur pstn line |
| 16:00.26 | Aut0ExeC | psilikon: do I just redirect it right back out to line1? |
| 16:00.51 | Docfxit | psilikon I'm getting an error with the start Asterisk died with code 1 |
| 16:01.38 | *** join/#asterisk tstorm (~tstorm@173-164-230-21-SFBA.hfc.comcastbusiness.net) |
| 16:01.49 | Docfxit | psilikon Just before that line I get Asterisk ended with exit status 1 |
| 16:02.06 | psilikon | Aut0ExeC, yes. You direct the incoming call where you want it. For instance maybe a call from an specific itsp that matches a specific callerid gets passed to a specific extension. |
| 16:02.42 | psilikon | Aut0ExeC, You should really check out Asterisk: The future of Telephony's ch. 5. \ |
| 16:02.53 | Aut0ExeC | psilikon: ok |
| 16:02.56 | Aut0ExeC | thanks |
| 16:03.02 | *** join/#asterisk friehmaen (freeman@xers.de) |
| 16:03.35 | psilikon | Docfxit, that means it is in fact already running. |
| 16:03.43 | Docfxit | psilikon It keeps repeating the same errors over and over again. |
| 16:03.44 | psilikon | Docfxit, killall -9 safe_asterisk and killall -9 asterisk |
| 16:04.01 | psilikon | Docfxit, then run /etc/init.d/asterisk start |
| 16:06.26 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 16:09.12 | Docfxit | psilikon You fixed it. That's super. :-) Thank you very very much. |
| 16:10.00 | Docfxit | psilikon I don't understand why rebooting didn't fix it. |
| 16:10.07 | psilikon | Docfxit, remember those killall commands. |
| 16:10.44 | Docfxit | psilikon That's really great. |
| 16:13.47 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 16:15.38 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 16:16.47 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:18.29 | *** join/#asterisk ectospasm (~trey@75.76.170.25) |
| 16:21.34 | *** join/#asterisk koffel (~whome@c-174-57-81-113.hsd1.pa.comcast.net) |
| 16:21.51 | Docfxit | I have a softphone that is getting an error when I dial 9 + area code + phone number. The error is rejected because extension not found. |
| 16:22.22 | psilikon | Docfxit, you need something in your dialplan to accept 91NXXNXXXXXX |
| 16:23.28 | Docfxit | psilikon Could I show you my dialplan privately ?¿ |
| 16:24.21 | m_tadeu | does anyone know what's the asterisk-gui default login/pass in asterisknow? |
| 16:24.23 | Docfxit | psilikon When I dial from a local phone it dials out just fine. |
| 16:26.43 | psilikon | Docfxit, What is the user id for the sip phone? |
| 16:27.31 | Docfxit | psilikon gary |
| 16:27.32 | Docfxit | Gary |
| 16:27.42 | *** join/#asterisk darkskiez (~dz@206.71.84.68) |
| 16:28.03 | Docfxit | psilikon user id = gary |
| 16:28.04 | Docfxit | Gary |
| 16:29.30 | psilikon | In the sip.conf under the gary section, what do you have for context? |
| 16:30.48 | Docfxit | psilikon context = default |
| 16:31.27 | psilikon | Docfxit, you can dial 9+ and get out from another phone? |
| 16:31.41 | Docfxit | psilikon yes. |
| 16:31.57 | psilikon | Docfxit, what type of phone works? |
| 16:32.11 | psilikon | Docfxit, Can you paste your extensions.conf and sip.conf? |
| 16:32.37 | Docfxit | psilikon yes. I'll do that. |
| 16:33.43 | Docfxit | psilikon Can I do that privately ?¿ |
| 16:34.24 | psilikon | Docfxit, ok whatever |
| 16:40.45 | *** join/#asterisk tasca (~tasca@189.34.27.64) |
| 16:42.30 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 16:44.30 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 16:46.12 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 16:49.14 | *** part/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 16:49.19 | *** join/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 16:51.19 | Docfxit | Is this channel alive? |
| 16:51.23 | tuxx- | <PROTECTED> |
| 16:51.53 | *** join/#asterisk r0d3nt (r0d3nt@foster.stonedcoder.org) |
| 16:56.00 | WIMPy | The channel is, but the members are dead. |
| 16:56.55 | *** join/#asterisk \DSAFEW\ (~dsafew@ip70-190-95-86.ph.ph.cox.net) |
| 16:58.06 | Docfxit | I hope it's not a permanent dead. |
| 16:58.41 | WIMPy | Haven't you heard of zombies? |
| 16:59.03 | Docfxit | Sure. In the movies. |
| 17:00.59 | Docfxit | I've never seen this happen. Do they have to reset the channel? Do we have to reconnect? Does anyone know? |
| 17:02.03 | WIMPy | See what happen? |
| 17:02.31 | *** join/#asterisk hozano (~hozano@200.17.112.83) |
| 17:02.51 | Docfxit | This was very active a few min ago. Where did everyone go? |
| 17:03.00 | hozano | can i use asterisk without a phone line? |
| 17:03.05 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 17:03.10 | _Corey_ | It's lunchtime. All the zombies are eating brains... |
| 17:03.29 | WIMPy | hozano: yes |
| 17:03.46 | wdoekes2 | no it's not, it's evening |
| 17:03.48 | Docfxit | Does anyone know where psilikon went? |
| 17:03.51 | wdoekes2 | but yes, we're having brains for dinner |
| 17:05.00 | hozano | WIMPy, I have installed asterisk on ubuntu via apt-get. The service is running. How to test it? |
| 17:05.39 | WIMPy | Connect a phone to it. |
| 17:07.34 | hozano | WIMPy, I am trying to connect to it using XLite, but it says "Authentication error" |
| 17:08.59 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:09.20 | hozano | WIMPy, from ubuntu, i have installed linphone, but it requires a SIP server with a domain name. It does not accept an IP address to set the sip server config |
| 17:09.48 | *** join/#asterisk outtolunc (~outtolunc@204.sub-75-208-52.myvzw.com) |
| 17:10.07 | *** join/#asterisk outtolunc (~outtolunc@204.sub-75-208-52.myvzw.com) |
| 17:10.39 | WIMPy | Well, make sure, user and password match. |
| 17:23.26 | Docfxit | Most of the extensions setup are 3 or 4 digit numbers. I setup a softphone with a user ID of my name. Is the user ID the same as an extension? Am I supposed to create a extension with the user ID? |
| 17:23.44 | WIMPy | no |
| 17:24.05 | WIMPy | But you can, if you like one. |
| 17:24.11 | *** join/#asterisk darkskiez (~dz@206.71.84.68) |
| 17:24.37 | Docfxit | How does asterisk relate my user ID to an extension for a mail box? |
| 17:25.08 | WIMPy | You configure that in sip.conf. |
| 17:25.32 | WIMPy | And from the extension to the user in extensions.conf. |
| 17:26.26 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-215.mobile.uci.edu) |
| 17:26.59 | Docfxit | So in sip.conf the extension would be caller ID ?¿ |
| 17:27.06 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 17:27.29 | WIMPy | There is no extensuion there. |
| 17:27.53 | WIMPy | But it makes sense to set the caller ID to the extension that goes to that user. |
| 17:27.58 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 17:28.05 | WIMPy | Did you try the book? |
| 17:28.10 | WIMPy | ~newbook |
| 17:28.10 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 17:28.53 | bent_screwdriver | is it necessary to get echo cancellation module on a digium TDM card that will only be used to recieve fax on singe channel? |
| 17:29.04 | *** join/#asterisk n0tk (~n0tk@216.160.42.30) |
| 17:29.18 | Docfxit | Where would I find that in the book? |
| 17:29.24 | WIMPy | bent_screwdriver: No. In fact you don't want EC for Fax. |
| 17:29.51 | WIMPy | Docfxit: the parts about sip peers and extensions. |
| 17:30.16 | bent_screwdriver | thx! |
| 17:30.27 | *** join/#asterisk outtolunc (~outtolunc@86.sub-69-99-146.myvzw.com) |
| 17:30.37 | *** join/#asterisk luke-jr (~luke-jr@ishibashi.dashjr.org) |
| 17:31.36 | *** join/#asterisk Freeaqingme_ (~dolf@dsl-083-247-011-232.solcon.nl) |
| 17:31.36 | *** join/#asterisk LedZeplin (jbearer@74.46.1343.static.theplanet.com) |
| 17:32.42 | LedZeplin | I'm adding a T1 PRI to my asterisk server, and the phone company wants to know the what version or protocol to use for the PRI but i'm not having much luck finding info on what to tell them. |
| 17:33.04 | _Corey_ | LedZeplin: Are you in the US? |
| 17:33.08 | LedZeplin | yes |
| 17:33.17 | WIMPy | LedZeplin: That's an interesting direction. |
| 17:33.29 | _Corey_ | ESF/B8ZS/NI-2 is usually the way to go |
| 17:33.31 | WIMPy | Usually you have to ask them that question. |
| 17:33.44 | _Corey_ | yeah, seriously... |
| 17:33.57 | LedZeplin | It's the 'new' verizon I guess ;) |
| 17:36.06 | LedZeplin | and the other question was the number of digits to send on incoming calls |
| 17:36.46 | WIMPy | As in all or less? |
| 17:36.51 | _Corey_ | LedZeplin: Up to you, depends on how many #s and how many areas you're getting DIDs in |
| 17:37.07 | _Corey_ | most small offices get 4, we usually do 10 when we can |
| 17:38.20 | LedZeplin | am I correct in presuming that 4 digits will let you do 10000 DIDs? |
| 17:38.42 | _Corey_ | 4=last 4 digits of the phone number |
| 17:39.05 | _Corey_ | if you're eventually going to get more than a hundred I would definitely ask for all 10 digits |
| 17:39.18 | _Corey_ | it gets confusing when you're covering multiple NPA NXXs |
| 17:39.21 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 17:39.39 | WIMPy | If you plan to set up more than one box or to be active in more than one area code, go for all. |
| 17:39.39 | LedZeplin | I see. |
| 17:41.37 | LedZeplin | they also asked if our main number is what we want displays on outgoing calls from the pri. Do I tell them I don't want that set because I'll handled DID CID inside of asterisk? |
| 17:43.57 | _Corey_ | I like to have the flexibility, regardless of what my customer says they need... |
| 17:44.15 | _Corey_ | Best to have the ability to set it as needed |
| 17:46.08 | *** part/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 17:46.11 | *** join/#asterisk niros (~nir@bzq-218-196-74.red.bezeqint.net) |
| 17:46.31 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 17:49.05 | LedZeplin | _Corey_ WIMPy : thanks for the help. one last question, so I can get smarter, what does the parts of the ESF//B8ZS/NI-2 stand for so I can read up on that? |
| 17:49.27 | niros | Hello, i'v recently noticed tariffs with the names: "Personal Numbers" and: "National Calls" . seems they apper only in UK. someone might have a clue what's their meaning ? |
| 17:50.32 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 17:50.32 | _Corey_ | LedZeplin: sure, so ESF is framing, B8ZS is coding, and NI-2 is National 2 signaling ('national' in chan_dahdi) |
| 17:50.34 | WIMPy | LedZeplin: Dunno, what ESF is. B8ZS is the line coding and NI-2 the signalling protocol. |
| 17:51.01 | _Corey_ | :) |
| 17:52.02 | *** join/#asterisk niros (~nir@bzq-218-196-74.red.bezeqint.net) |
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| 18:05.32 | *** join/#asterisk msetim (~setim@186.214.197.221) |
| 18:05.40 | msetim | hi guys |
| 18:06.51 | msetim | anybody knows the oeboema digium user? |
| 18:07.06 | Freeaqingme_ | oeboema? |
| 18:07.32 | msetim | Freeaqingme, yeah... https://issues.asterisk.org/bug_view_page.php?bug_id=18986 |
| 18:08.18 | Freeaqingme_ | aah |
| 18:11.17 | *** join/#asterisk tuxx- (tuxx@vps460.directvps.nl) |
| 18:18.43 | msetim | Does anybody know who he/she is? |
| 18:21.04 | *** join/#asterisk sahafeez (~sahafeez@99-105-56-250.lightspeed.sntcca.sbcglobal.net) |
| 18:27.33 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 18:28.24 | Aut0ExeC | I'm using spa... if I specify <@gw0>S0 in my dialplan... will that send the call thru to the pstn line without it going to asterisk server? |
| 18:28.36 | Aut0ExeC | essentially bypassing asterisk server? |
| 18:29.08 | Aut0ExeC | or does it just send it to the dialplan that I set up in asterisk that gw0 points to? |
| 18:30.45 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 18:35.39 | Aut0ExeC | sorry i'm a nub at this |
| 18:36.11 | *** join/#asterisk KNERD (~KNERD@99.96.118.71) |
| 18:36.51 | msetim | I need to ask for a direct contact with a report of digium. I would like to pay for a development on specific version. is it permitted to ask there? I google and I don't find out the user contact. |
| 18:37.17 | msetim | There is no rules about it on http://www.asterisk.org/developers/bug-guidelines |
| 18:39.04 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 18:39.27 | *** part/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 18:40.36 | retentiveboy | So, I swear I've been able to "reload" without affecting active call but I just killed on leg in a MeetMe conference inadvertently. A bug? Or, and I just being stupid today. |
| 18:42.39 | retentiveboy | The one station here that was in the conference stayed connected but an offsite SIP station was cut off. The CDR record for that station's call never made it to the database either. Weird. |
| 18:43.33 | *** join/#asterisk AMindMobile (~AMindMobi@95-28-106-82.broadband.corbina.ru) |
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| 18:55.32 | *** join/#asterisk Charrit (~Zairus@226.109.165.83.dynamic.mundo-r.com) |
| 18:55.35 | Charrit | hi |
| 18:56.42 | Charrit | how can I retrieve the SIP username from dialplan? Exists any special variable? |
| 18:59.02 | Aut0ExeC | Charrit: ??? |
| 18:59.05 | Aut0ExeC | explain urself |
| 18:59.10 | *** join/#asterisk nighty^ (~nighty@tin51-1-82-226-147-104.fbx.proxad.net) |
| 18:59.30 | Charrit | [john123] |
| 18:59.35 | Charrit | secret=11111 |
| 18:59.39 | Charrit | canreinvite=no |
| 18:59.40 | Charrit | ... |
| 18:59.42 | Aut0ExeC | ok |
| 19:00.06 | Charrit | I want to extract 'john123' from dialplan |
| 19:00.31 | Charrit | in channel variable or something similar to compare with values that I stored in astdb |
| 19:00.33 | benngard | when od u want to extract the name? |
| 19:00.47 | *** join/#asterisk dev-ast (~dshah@utdpat242007.utdallas.edu) |
| 19:00.55 | benngard | as a caller or callee? |
| 19:00.56 | Charrit | exten => _9.,n,${} |
| 19:01.03 | Charrit | caller |
| 19:01.19 | benngard | $CALLERID(name)} |
| 19:02.18 | benngard | s/$C/${C |
| 19:02.24 | Charrit | but these var can be changed in the sip account, I think |
| 19:03.09 | Charrit | with callerid=John Federer |
| 19:03.30 | Charrit | and then it'll return 'John Federer', or not? |
| 19:05.17 | *** join/#asterisk outtolunc (~outtolunc@c-76-21-79-73.hsd1.ca.comcast.net) |
| 19:05.41 | kaldemar | Charrit: see if funtion SIPPEER has a field. caller id is not really the same thing. |
| 19:07.00 | Charrit | ok, i'll take a look into |
| 19:07.34 | Aut0ExeC | any of u guys have spa3XXX? |
| 19:07.45 | Aut0ExeC | I have a small question |
| 19:07.48 | benngard | no just spa21xx |
| 19:08.20 | kaldemar | Charrit: sorry, SIPCHANINFO(peername) is what you're looking for. |
| 19:09.00 | Charrit | ok, I'll try it |
| 19:09.03 | Charrit | a lot of thanks |
| 19:09.16 | Aut0ExeC | benngard: if I have <@gw0>S0 specified in my dialplan on the device... does that bypass asterisk server completely and place the call to pstn line? |
| 19:10.17 | benngard | Aut0ExeC: sorry, dont understand the question |
| 19:11.25 | Aut0ExeC | ok |
| 19:11.43 | benngard | hmm, shouldn BLINDTRANSFER show up when u type: core show functions? |
| 19:11.48 | Aut0ExeC | benngard: on ur "line" tab, you can specify a dialplan yes? |
| 19:12.11 | *** join/#asterisk raden_work (~jon@66-191-96-74.static.eucl.wi.charter.com) |
| 19:12.11 | Charrit | kaldemar, and in IAX would be with IAXPEER? |
| 19:12.15 | benngard | yes, but i have always used asterisk for that, but not for a long time |
| 19:14.19 | benngard | probably just me that is to tired, the function BLINDTRANSFER works on SIp cahnnels but dont show up as a function |
| 19:15.26 | WIMPy | Function? Application? |
| 19:17.33 | benngard | i use it as a function |
| 19:17.46 | benngard | exten => 0317998975-NOT_INUSE,n,ExecIf($[${LEN(${BLINDTRANSFER})}>0]?Dial(${CUT(BLINDTRANSFER,-,1)})) |
| 19:22.21 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:22.21 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:24.57 | kaldemar | Charrit: you'll get iax peer name with CHANNEL(peername). IAXPEER requires the peer name as an argument.. |
| 19:26.34 | Aut0ExeC | if anyone here uses or used a spa3102 can you please help me out with a small question |
| 19:26.37 | Charrit | I have asterisk 1.4, peername is missing |
| 19:26.42 | *** join/#asterisk outtolunc (~outtolunc@c-76-21-79-73.hsd1.ca.comcast.net) |
| 19:26.44 | Charrit | in CHANNEL |
| 19:29.10 | *** join/#asterisk zkn (~zkn@82.131.54.59.cable.starman.ee) |
| 19:30.10 | Charrit | I'll explain the objetive, perhaps would be a better way to acomplish it |
| 19:30.30 | Nugget | asterisk 1.4? Do you have indoor plumbing there? :) |
| 19:30.34 | kaldemar | Charrit: then you probably have to parse the channel name with func CUT and :5. |
| 19:31.31 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 19:31.32 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 19:32.01 | Charrit | I have one asterisk server, and other clients (IAX) connect to the server for making calls through a SIP provider configured in the server. I want to store in astdb the minutes elapsed in use for each of the users |
| 19:33.29 | Aut0ExeC | if anyone here has or used a cisco spa3XXX... I just need to know.. if I specify ...gw0>S0 in my dial on the device for line1.. does this completly bypass the asterisk server to send the call out to pstn line? |
| 19:34.22 | kaldemar | Charrit: setvar=... in channel configuration files could be one neat way. |
| 19:34.22 | Charrit | Nugget, I read about 1.6 have som problems with lines cards and I didn't upgrade for now |
| 19:35.35 | zkn | does anyone know if error messages "Call rejected by XXX.XXX.XXX.XXX: No authority found" (IAX2) and "Got SIP response 484 "Address Incomplete" back from XXX.XXX.XXX.XXX:5060" (SIP) are some 1.8.3 version "features" that occasionally occur when dialing out ? |
| 19:36.06 | Charrit | kaldemar, Could I use setvar in iax.conf? |
| 19:38.01 | kaldemar | Charrit: yes. |
| 19:39.27 | zkn | Charrit: you can also use setvar=IAXVAR(var)=value |
| 19:40.14 | Aut0ExeC | spa3102 is not a popular item I gather |
| 19:40.16 | Aut0ExeC | :( |
| 19:40.16 | Charrit | Ok, it seems a perfect way to acomplish my objetive |
| 19:40.23 | Charrit | thanks again |
| 19:41.33 | zkn | have there been rumors about a date when 1.8.4 will be released? |
| 19:42.23 | Aut0ExeC | zkn: you do telephony for a living? |
| 19:42.35 | zkn | i wish :) |
| 19:42.41 | Aut0ExeC | oh |
| 19:42.48 | Aut0ExeC | why wish.. when you can |
| 19:42.49 | Aut0ExeC | ? |
| 19:42.58 | Aut0ExeC | :) |
| 19:43.11 | zkn | well, it's part of my job but not a major part |
| 19:43.20 | Aut0ExeC | ok asterisk ? |
| 19:43.23 | zkn | yes |
| 19:43.28 | Aut0ExeC | damn sweet man |
| 19:43.34 | Aut0ExeC | T1? |
| 19:43.53 | Aut0ExeC | with ip phones right? |
| 19:44.05 | Aut0ExeC | all cisco |
| 19:44.49 | zkn | nope, not that hardcore :) using simple interconnected asterisk server in an internatinal company with offices arounf the globe |
| 19:44.58 | zkn | servers |
| 19:45.10 | Aut0ExeC | oh ok |
| 19:45.25 | Aut0ExeC | yeah I work in IT myself |
| 19:45.42 | Aut0ExeC | we have a system here with digital phones |
| 19:45.48 | Aut0ExeC | Avaya |
| 19:46.20 | Aut0ExeC | I want to get into telephony so bad tho man... |
| 19:46.33 | Aut0ExeC | I want to be L33t |
| 19:47.50 | zkn | well, takes some dedications i guess |
| 19:48.06 | _Corey_ | or bad luck, depending on your perspective ;) |
| 19:48.14 | zkn | hehe |
| 19:48.19 | elb | horrible luck |
| 19:48.34 | elb | "oh hey, let me dive into a field that's more or less stuck in 1910" |
| 19:48.41 | elb | ;-) |
| 19:48.42 | WIMPy | Or your location :-) |
| 19:51.02 | *** join/#asterisk outtolunc (~outtolunc@c-76-21-79-73.hsd1.ca.comcast.net) |
| 19:52.37 | zkn | so, no thoughts about these channel errors I'm getting with 1.8.3 ? |
| 19:54.37 | Charrit | zkn, review yout sip.conf and iax.conf in both locations, normally these channel errors are caused by a misconfiguration or password mismatch |
| 19:55.01 | Charrit | (In the case of No authority found) |
| 19:56.08 | Charrit | or iax username not found in iax.conf |
| 19:56.34 | zkn | yesh, normally, weird thing about these is that when I'm monioring the CLI and notice either of these errors occurring, I can dial the same number again, and it will work.. this failure occurs randomly to my untrained eye and has beacome a problem since I started to use 1.8.3 |
| 19:58.40 | zkn | 1.8 versions have really nice features so i don't really want to revert back to earier versions |
| 19:59.07 | zkn | so far it seems that 1.4 is considered the most stable |
| 19:59.19 | elb | I've been having iax2 registration problems with 1.8.3 as well |
| 19:59.25 | Charrit | perhaps a bug, try latest svn |
| 19:59.26 | elb | not that particular problem |
| 20:00.03 | elb | but sometimes it'll stop communicating with a server it knows about, telling the remote server it doesn't accept RSA auth |
| 20:00.09 | elb | iax2 reload fixes it, normally |
| 20:00.36 | zkn | sigh |
| 20:00.44 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 20:04.18 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 20:05.40 | *** join/#asterisk tasca (~tasca@189.34.27.64) |
| 20:05.40 | zkn | would be nice to get everything working properly so I can just forget about all servers for several months |
| 20:06.38 | elb | amen :-) |
| 20:06.57 | WIMPy | ggg |
| 20:08.06 | zkn | dont get me wrong - writing dialplan and thinking of how to solve stuff with Asterisk is fun, but to erradicate issues related to tings i have no idea about is not so much fun anymore |
| 20:08.38 | WIMPy | You'll get used to that as well. |
| 20:08.38 | zkn | of course, there is no bug free software |
| 20:09.20 | Charrit | I think that 1.8 is too young for use in production enviroments, you get the new features but a lot of headaches too |
| 20:09.37 | WIMPy | Not everything that makes things not work is a bug. |
| 20:09.50 | Charrit | sure |
| 20:10.04 | ariel_ | 1.8 is a released version not a rc or beta correct? |
| 20:10.21 | Charrit | yes |
| 20:12.16 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 20:15.30 | Charrit | with Dial(SIP/1231, g)... the dialplan will continue in case of a sucess call? or should I use deadagi as the only way? |
| 20:16.17 | Tozz_ | no it wont |
| 20:16.27 | Tozz_ | oh, g |
| 20:16.57 | Tozz_ | i think deadagi |
| 20:17.13 | Tozz_ | because if voip-info is right, it only continues in the dialplan if the called user hangs up |
| 20:17.31 | Tozz_ | not the caller |
| 20:17.52 | WIMPy | If the caller hangs up, it's always over. |
| 20:18.14 | zkn | oh, yeah, i've been wondering about the same thing if it's possible to do "multitasking" |
| 20:18.14 | Charrit | No, with 'g' parameter dialplan continues |
| 20:18.32 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 20:19.12 | Tozz_ | only if the called person hangs up, not if the caller hangs up |
| 20:19.22 | Charrit | or... it should continue, but as Tozz_ said if the caller hangs up I'm not sure |
| 20:20.39 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 20:25.34 | TeknoJuce | is there a general scale on how much an hour of g711 codec or higher consumes? |
| 20:26.54 | WIMPy | 8000octetts per second and direction. |
| 20:28.32 | TeknoJuce | http://www.voip-info.org/wiki/view/Bandwidth+consumption |
| 20:28.38 | TeknoJuce | guess something like that |
| 20:33.29 | TeknoJuce | so around 40MB of data for an hour of coversation |
| 20:33.41 | TeknoJuce | of g711 |
| 20:33.47 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
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| 20:41.20 | TeknoJuce | is this something that I set accendently? keeps firing off digit 5 during the conversations every couple mins: |
| 20:41.21 | TeknoJuce | Send Digit 5 |
| 20:41.21 | TeknoJuce | Sending Stream Based Tone Frequency Component List Download 770 1336 |
| 20:41.21 | TeknoJuce | Sending Stream Based Tone On |
| 20:41.21 | TeknoJuce | Sending Stream Based Tone Off |
| 20:41.21 | TeknoJuce | Send Digit off 5 |
| 20:43.06 | *** part/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 20:43.58 | TeknoJuce | though if that chart is just one direction then I guess it would be around 80MB/h |
| 20:44.38 | WIMPy | And RTP is evil, if that's involved. |
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| 21:45.54 | *** join/#asterisk guifort (~guifort@20.113.98-84.rev.gaoland.net) |
| 21:45.57 | guifort | Hello |
| 21:46.36 | guifort | can you help me for a configuration of extention.conf a small problem with country prefix ? |
| 21:49.26 | WIMPy | ~ask |
| 21:49.26 | 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. |
| 21:50.58 | guifort | I have a problem with registered number on my phone I am in france , When I dial a number 06XXXXXXXXX for exemple it's work my SIP provider accept the call |
| 21:51.22 | guifort | With : exten => _0[6-7]XXXXXXXX,1,Dial(SIP/XXXX/${EXTEN}) |
| 21:52.21 | guifort | When I call a number who start with 00336XXXX it's doesnt work, my sip provider say the number is false. I use this rule exten => _0033[6-7]XXXXXXXX,1,Dial(SIP/ |
| 21:52.46 | guifort | How can I remove the 0033 prefix and replace it with 0 for remake the standart number 06 XX |
| 21:53.07 | WIMPy | 0{EXTEN:4} |
| 21:53.21 | WIMPy | 0${EXTEN:4} |
| 21:54.35 | guifort | I try , thx |
| 21:55.22 | jdoe | what do the RTCP stats actually measure? If I call my cell from a sip phone, leave it silent for a few seconds, then hang up I get a huge number in lost packets... but there are no call quality problems. Surely there aren't that many being dropped. |
| 21:55.51 | guifort | It's working, thx very much =) |
| 21:56.03 | elb | perhaps silence suppression isn't being correctly communicated |
| 21:56.20 | elb | do you get no dropped packets if you whistle into the mic for the duration of the call? |
| 21:56.50 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 22:02.55 | SunTsu | Er, what the ...? I just solved my week old problem that I was not able to do outgoing calls on one of my SIP providers... I needed to set remotesecret AND secret for the outgoing extension |
| 22:03.15 | *** join/#asterisk outtolunc (~outtolunc@c-76-21-79-73.hsd1.ca.comcast.net) |
| 22:06.18 | WIMPy | That does not seem to fit to the documentation. |
| 22:07.26 | SunTsu | yeah, I don't understand why. without secret set I get AUTH failure on the call |
| 22:08.46 | SunTsu | but, riddle solved - on the day my new sip hardware was delivered |
| 22:13.28 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 22:29.41 | *** join/#asterisk cyphorious (~cyphoriou@chello062178189196.2.15.tuwien.teleweb.at) |
| 22:30.27 | *** join/#asterisk hairyraven (~nobody@95.73.138.192) |
| 22:30.36 | jdoe | elb: dunno. Asterisk doesn't support it afaik, I don't see any warnings about it in my logs like I used to in any case. |
| 22:40.25 | *** join/#asterisk DanFromUK (~DanFromUK@2.26.252.165) |
| 22:40.58 | DanFromUK | hi, is it possible to run a macro when a queue member picks up a call? (asterisk 1.4) |
| 22:48.47 | *** join/#asterisk brightidea1980 (~root@77.30.197.124) |
| 22:56.19 | KNERD | DanFromUK: run |
| 22:56.28 | KNERD | DanFromUK: yes |
| 22:57.32 | KNERD | What does this mean? <--- Reliably Transmitting (NAT) to 192.168.5.108:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.5.108:5060;branch=z9hG4bK-48ee511f;received=192.168.5.108;rport=5060 |
| 22:58.16 | *** join/#asterisk wonderworld (~ww@port-92-201-2-51.dynamic.qsc.de) |
| 23:01.02 | *** join/#asterisk kchehab (~kchehab@89.249.210.40) |
| 23:01.06 | kchehab | hi ppl |
| 23:01.31 | kchehab | i am trying to monitor all outgoing faxes from my pbx extensions |
| 23:01.58 | kchehab | i tired to recieve and resend them ,but that will not give the user the real status for the call |
| 23:02.01 | kchehab | any ideas |
| 23:02.06 | p3nguin | knerd: It means that authentication is required and was not already performed. |
| 23:02.38 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 23:03.17 | DanFromUK | KNERD: any pointers? I think ive found something about using an AGI instead. |
| 23:03.44 | KNERD | p3nguin: thanks...so would that mean it would try athenticating? |
| 23:03.59 | KNERD | DanFromUK: AGI could be a good place to start |
| 23:06.13 | DanFromUK | KNERD: doesnt look like AGI does what i want. Can AGI start mixmonitor? |
| 23:06.42 | KNERD | DanFromUK: AGI can tell asterisk to do whatever you want it to do |
| 23:06.54 | hardwire | LIES! |
| 23:06.58 | hardwire | It cannot make pie. |
| 23:07.10 | KNERD | wanna bet? |
| 23:07.14 | kchehab | ? |
| 23:07.19 | hardwire | show me pie. |
| 23:07.21 | KNERD | i can wire it up to the stove to turn it on or off |
| 23:07.32 | hardwire | pshaw! |
| 23:07.34 | DanFromUK | KNERD: found it. $agi->exec('MixMonitor', '/usr/share/asterisk/sounds/sounds/conversation.wav'); |
| 23:07.38 | DanFromUK | thanks for the pointers |
| 23:07.46 | KNERD | sure |
| 23:08.24 | DanFromUK | there should be a site dedicated to weird and wonderful projects using asterisk |
| 23:08.58 | DanFromUK | night all. |
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| 23:14.49 | igorww | does anyone know if "findslot=next" option for parking slots is broken? I can not seem to get it to work. |
| 23:17.27 | p3nguin | knerd: Normally, one side sends a call, the other side rejects with a 401 (meaning it needs to authenticate), then the sending side will authenticate. The 401 is normal operation in the call process. |
| 23:17.53 | KNERD | p3nguin: thanks for the clarification |
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| 23:45.06 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.247.38) |
| 23:45.08 | DelphiWorld | hi guys |
| 23:45.14 | DelphiWorld | anyone used mgcp with asterisk? |
| 23:46.17 | ariel_ | why? |
| 23:47.08 | elb | looking for solidarity is common among software users |
| 23:47.13 | elb | it makes them feel warm and fuzzy inside |
| 23:52.16 | *** join/#asterisk Dobah (~chatzilla@cpe-98-150-177-51.hawaii.res.rr.com) |
| 23:52.35 | DelphiWorld | i have a mgcp endpoint |
| 23:52.41 | DelphiWorld | and i want to serv it using asterisk |
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