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| 00:15.35 | Traderz | pretty quiet |
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| 00:23.32 | Cesare | hello |
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| 01:57.49 | Cesare | i have some problems in debugging the registration to a sip provider, can anyone give me a little help ? i'm sure is trivial (because i'm bit noob ) |
| 01:58.00 | kaldemar | koffel: see if you have a SIP ALG or anything similar on your router. if so, disable it. |
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| 02:07.29 | koffel | nothing else |
| 02:09.05 | koffel | i have asterisk and freepbx on my box |
| 02:09.20 | koffel | and modem directly to my asterisk box |
| 02:09.43 | koffel | i get that error |
| 02:09.50 | kaldemar | koffel: on your modem/router, NOT you asterisk box. |
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| 02:10.00 | koffel | only when it goes to voice mail |
| 02:10.35 | koffel | my modem aka gateway goes direct |
| 02:10.50 | koffel | nothing on the modem expect 2 pc |
| 02:13.35 | kaldemar | in the gateway software there may be an application level gateway software that causes problems. |
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| 02:19.58 | koffel | i guess i gota call comcast then |
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| 02:23.19 | ruben23 | hi guys when i reboot asterisk and run asterisk -rvvv i always have this ---> Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 02:24.56 | ruben23 | any idea guys..? |
| 02:25.22 | koffel | try asterisk start |
| 02:26.02 | jaytee | service asterisk start if you're running Red Hat or CentOS, /etc/init.d/asterisk start if running ubuntu |
| 02:26.10 | kaldemar | ruben23: it's not running or it's running as a different user as the one you're trying to connect with. |
| 02:26.36 | ruben23 | i have this when i run asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvgc ---------------------->http://pastebin.com/SYcf2zes is this migth cause the problem |
| 02:27.10 | ruben23 | [Apr 4 10:25:11] WARNING[1827]: chan_iax2.c:12772 load_module: Unable to open IAX timing interface: No such file or directory <--------------------is thisw a major error..? |
| 02:28.59 | kaldemar | ruben23: no, it is a warning like it says. does asterisk start with -c or is the startup interrupted? |
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| 02:48.06 | TTT_Travis | Trying to transfer my home phone to VOIP somehow but none say number is portable. I can only port to cell phone through Verizon Wireless. Are those my only options? |
| 02:51.09 | jaytee | what ITSP are you using? |
| 02:51.55 | jaytee | you should be able to port most landline numbers |
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| 03:34.58 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 2.0-beta1 (2011/04/01), 1.8.3.2 (2011/03/17), 1.6.2.17.2 (2011/03/17), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
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| 05:11.27 | daxt | guys do u know any DID provider who uses AMR-NB codec ? |
| 05:14.13 | shapr | Is there some way to detect a hook flash on an FXS port? |
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| 06:45.41 | schmidts | good morning |
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| 07:12.37 | rlfx | hi, how to change music on parking? |
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| 07:30.49 | m_tadeu | hi...I have set rtp ports 20000-21000 in the router and in rtp.conf. but when I do a netstat, doesn't look like asterisk is listerning to those ports |
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| 07:40.12 | kaldemar | m_tadeu: don't let that bother you. is something not working? |
| 07:40.43 | p3nguin | m_tadeu: netstat -lnpu won't show rtp ports as listening because rtp is not a listening daemon. |
| 07:40.51 | m_tadeu | kaldemar: I'm unable to see rtp packages moving around when rtp debug on |
| 07:41.09 | kaldemar | m_tadeu: do you see them with tcpdump? |
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| 07:41.39 | p3nguin | or -lnu, or -lu |
| 07:47.25 | m_tadeu | thi seems to be a problem with the sip client...I tried with another client and I get rtp packets :( |
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| 07:50.06 | jg1234 | hi |
| 07:51.27 | p3nguin | I guess you know how to fix it, then. |
| 07:53.05 | jg1234 | i am trying to record a complete "analog telephony session" with dahdi_monitor, but it looks like the dialtone was cut off. Is there a way to prevent this ? |
| 07:53.13 | m_tadeu | in rtp debug, now I get the packets moving around....something I find strange is that the packets are from/to port 1026 |
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| 07:54.30 | E-bola | Do anybody have an example of a dialplan that lets users chose which outbound provider to dialout via by pressing 0 or 1 before the number to dial (or whatever they press) |
| 07:54.42 | kaldemar | m_tadeu: a client can choose its ports. nothing strange there. |
| 07:57.37 | m_tadeu | kaldemar: I thought it would use the ports defined in asterisk for rtp |
| 07:59.16 | kaldemar | m_tadeu: in rtp.conf you define what ports asterisk uses. the client has its own configuration. |
| 08:00.25 | p3nguin | Check the RTP ports that Asterisk is using in that call. |
| 08:00.41 | p3nguin | It is likely to be within the range you set in rtp.conf. |
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| 08:04.27 | Cesare | is there a better way to trigger a call from a script different from asterisk -rx "originate SIP/clickncall/123123123 extension 34" |
| 08:05.29 | kaldemar | Cesare: AMI originate or a callfile are your other choices. |
| 08:05.58 | Cesare | (i'm noob) |
| 08:06.00 | Cesare | :) |
| 08:07.02 | bratner | Hi! What would be the rational behind enabling/disabling rtpchecksums in rtp.conf? When i used wireshark to snoop on rtp traffic it did say that the checksums are bad. |
| 08:07.25 | p3nguin | cesare: That way should be fine. Make sure you include the context in the extension. |
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| 08:25.13 | upp | hello, can any one help me with a free german trunk, i want only to test if my asterisk can dial out |
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| 08:29.09 | Iiiak | plop |
| 08:30.32 | m_tadeu | I have rtp packets moving around but no sound on the sip client |
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| 08:31.31 | kaldemar | m_tadeu: is the client behind a NAT? |
| 08:32.03 | m_tadeu | kaldemar: it is...so is asterisk...different nat, I mean |
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| 08:33.35 | p3nguin | You've got to configure Asterisk's general section for nat as well as the peer which is behind NAT. |
| 08:34.01 | p3nguin | What the heck would be the point to using ext4 with the journal *disabled* as opposed to simply using ext2? |
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| 08:35.51 | m_tadeu | p3nguin: it's set up for nat=yes....weird thing is, on the client side, wireshark reports all rtp packets with invalid checksum...is this normal? |
| 08:35.58 | Cesare | kaldemar: thanks, works really good :) |
| 08:36.14 | Cesare | kaldemar: and looks also faster |
| 08:36.34 | kaldemar | p3nguin: ext4 is more efficient etc. |
| 08:37.02 | p3nguin | So it makes sense to use ext4 with no journal instead of ext2 which doesn't have a journal? |
| 08:37.14 | p3nguin | I guess I have to reformat. |
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| 08:37.32 | p3nguin | pewp. |
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| 08:43.36 | m_tadeu | every packet that comes from asterisk has a checksum error...sip and rtp packets |
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| 08:58.41 | longword | m_t: Turn off TCP/IP checksum offloading on your NIC |
| 08:59.57 | upp | i want to test my Asterisk with a free sip trunk, can any one help me |
| 09:01.44 | kaldemar | upp: what do you need help for? |
| 09:03.27 | upp | kaldmar: do you know any free sip trunk on europ? |
| 09:04.25 | p3nguin | ~trunk |
| 09:04.26 | infobot | extra, extra, read all about it, trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 09:05.05 | kaldemar | however, for example "the definitive guide" uses the term "SIP trunk"... |
| 09:05.14 | p3nguin | As long as people keep saying "SIP trunk," other people are going to keep saying "SIP trunk." Just stop it. |
| 09:05.49 | kaldemar | upp: no, but you can look for some for example here: http://www.voip-info.org/wiki/view/VOIP+Service+Providers |
| 09:05.51 | upp | ahh but you know what i mean when i say " free sip trunk" provider |
| 09:06.01 | p3nguin | Why can't you say ITSP? |
| 09:06.06 | p3nguin | It's even less typing? |
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| 09:06.11 | Smirker | Hi hi |
| 09:06.14 | p3nguin | s/?/./ |
| 09:06.15 | upp | yes i have see all this things |
| 09:07.25 | kaldemar | p3nguin: i'm afraid the term has come to stay. |
| 09:07.36 | p3nguin | I have similar fears. |
| 09:07.51 | Smirker | I've got a SIP service, ip auth based, got 100 DIDs pointing to it. 200 concurrent channels. it all works, for the most part. however, 50% of the time when I dial my indials, the phone i call from doesn't start ringing for 5-10 seconds. on the rare occasion it times out after 30 seconds and gives me a busy tone. |
| 09:07.53 | kaldemar | not that i'm afraid of it. it doesn't bother me. |
| 09:08.22 | Smirker | i've monitored using asterisk -rvvvvvvvv, and i don't see anything during that 5-10 second wait time. only when it starts ringing. |
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| 09:08.37 | Smirker | so i am wondering, is it possible that it is my end? or is it likely that it is my wholesalers end? |
| 09:09.10 | Smirker | i'm connected via private network to my wholesaler. |
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| 09:09.53 | kaldemar | Smirker: see sip debug and tcpdump to find out whether you get any SIP packets. |
| 09:10.11 | Smirker | does tcpdump do udp? |
| 09:10.22 | shapr | yes |
| 09:10.28 | Smirker | aight. |
| 09:10.39 | Smirker | maybe they should call it pdump. |
| 09:10.58 | kaldemar | Smirker: depending on your setup, delay in DNS requests may also cause delay. |
| 09:11.34 | upp | kaldemar: i have set astersik and they work on my LAN perfect, now i want to dial to the outside, so i need trunk? |
| 09:11.38 | bratner | I have several peers configured and i want my calls to go out of the one with the lowest "qualify" number. is there an existing macro for it? |
| 09:14.29 | p3nguin | Do you know what the qualify number indicates? |
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| 09:17.12 | Smirker | with sip set debug on, nothing happens while i wait that 5-10 seconds. then as soon as my phone starts ringing I see an INVITE and away it goes |
| 09:17.26 | Smirker | does that imply my provider might have something up? |
| 09:17.48 | Smirker | (i'm waiting for them to call me back) |
| 09:17.59 | kaldemar | upp: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt599.html |
| 09:18.26 | upp | ok thanks |
| 09:19.34 | Smirker | kaldemar: i only have IPs set up in my sip config, no hostnames. |
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| 09:32.08 | m_tadeu | shouldn't sip show peers xxxx display something about the nat config for that peer? |
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| 09:33.07 | kaldemar | m_tadeu: it does, "Force rport" equals to the nat parameter in sip.conf. |
| 09:33.45 | m_tadeu | kaldemar: ah ok...thanx |
| 09:36.11 | Smirker | any other ideas why there would be a 5-10 second wait time? not using hostnames anywhere. i don't see the INVITE message while waiting those 5-10 seconds. |
| 09:39.31 | p3nguin | 10 seconds is a long time to wait for a call to start, if you're sure the number has been dialed and sent. You're sure that you're not experiencing a delay in the phone's digit map, where you dial the numbers but the phone doesn't actually dial and start the call? |
| 09:40.41 | Smirker | urg netsplit |
| 09:41.21 | Smirker | p3nguin: i agree. i've tried from landline and mobile. i've also had several other people try it. it's not that, because sometimes it happens pretty much instantly, sometimes 5-10 seconds, rarely 30 seconds/busy tone (due to, i assume, timeout). |
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| 09:45.04 | m_tadeu | I can't find out the reson I can't hear any sound from the ivr in the sip client... |
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| 09:48.24 | kaldemar | bratner: no existing macro in asterisk itself. someone may have cooked up something. it will without doubt be quite hairy. either use manager interface or shell to get the qualify values and then do a comparison. |
| 09:49.14 | jg1234 | is there a way to get monitor NOT to filter out dtmf ? |
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| 09:50.36 | kaldemar | bratner: something as awful as Set(RTT=${SHELL(asterisk -rx "sip show peer peername" | awk '/Status/ {print $4}'):1}) would get you the time in dialplan though. |
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| 09:55.41 | bratner | kaldemar, i hoped somebody already did the dirty work. i got a perl script that does basically the same though it works over the manager tcp socket. and returns the name of the peer with the lowest value. |
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| 09:56.30 | bratner | another question, is there a way to group peers so a Dial() command will load-balance between the group members? |
| 09:57.21 | kaldemar | the Dial application doesn't do any balancing by itself. you must write the logic yourself. |
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| 10:49.09 | nickfennell | Anyone played with Elastix ? |
| 10:51.44 | jg1234 | does anyone have a clue why dahdi_monitor is always dropping 1.5s of the dialtone ? |
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| 11:38.12 | m_tadeu | ok...on the internal network I have ivr sound, but if the client is on another nat, I get rtp packages, but no sound...what can this be? |
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| 11:42.36 | kaldemar | m_tadeu: misconfigured nat settings is the first guess. |
| 11:45.11 | m_tadeu | kaldemar: my sip.conf seems good, but maybe you can give it a look(http://pastebin.com/ybYn3YXH)...the sip peers are in the database |
| 11:47.24 | kaldemar | you have both externhost and externip defined. one is enough. the externhost must translate to an address in your name service. |
| 11:48.08 | kaldemar | now, tell what "in another NAT" means. the client is outside your LAN, but does it have a public address or it is behind a NAT too? |
| 11:49.10 | m_tadeu | asterisk is behind a nat and the client is behind another nat(not the asterisk nat) |
| 11:49.18 | m_tadeu | none of them have a public ip |
| 11:49.43 | kaldemar | does the client device have nat=yes in its configuration? |
| 11:50.50 | m_tadeu | I'm using zoiper....it doesn't have that config |
| 11:50.59 | kaldemar | in asterisk. |
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| 11:51.53 | m_tadeu | in the database, is set to null, which I figure it should use the default value described in sip.conf |
| 11:53.13 | kaldemar | check it. |
| 11:53.56 | m_tadeu | sip show peer xxx says force fport : yes |
| 11:54.02 | kaldemar | enable sip debug and pastebin a call. |
| 11:54.04 | m_tadeu | *rport |
| 11:54.18 | m_tadeu | ok |
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| 11:56.16 | m_tadeu | kaldemar: here it is http://pastebin.com/D7HeEfkS |
| 12:03.30 | m_tadeu | kaldemar: I see nothing strange there...the weird thing is, I have rtp packages moving around |
| 12:03.41 | kaldemar | m_tadeu: looks quite normal when it comes to addresses. |
| 12:04.09 | m_tadeu | when I try to analyse with wireshark, I get a wave that looks like a small noise |
| 12:04.12 | kaldemar | is the client machine with the zoiper receiving any RTP packages? |
| 12:04.23 | m_tadeu | it is |
| 12:05.26 | kaldemar | maybe the machine has something borked like sound drivers. |
| 12:07.37 | m_tadeu | well I have 2 netwoek connections here(same pc, different internet connection), so I can test in and out the lan....when I'm in the lan I get sound |
| 12:08.26 | kaldemar | does the NAT router have some SIP functionality? |
| 12:09.34 | m_tadeu | it has....but looks unactive |
| 12:11.11 | *** join/#asterisk coppice (~chatzilla@9.160.232.220.dyn.pacific.net.hk) |
| 12:11.59 | kaldemar | make sure it is completely disabled. |
| 12:15.46 | Dovid | Anyone here use Tinet or HE ? |
| 12:21.43 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 12:47.23 | Scott-Mc | wondering if anyone has any ideas for a minor problem, I have a system which places an automated call to my local extension when something is needing done, (basically uses text-to-speech and drops a file in /var/spool/asterisk which the extension uses Playback()), the problem is it ends up leaving lots of voicemails so I am wondering if I can make this specific action not leave a voicemail when it's not answered. |
| 12:47.55 | m_tadeu | kaldemar: the router says the voip service is disabled....but I can see several asterisk processes running...do you think they might interfeer, dispite the ports forwarded? |
| 12:48.40 | m_tadeu | I'm unable to kill those processes |
| 12:49.04 | kaldemar | you're most likely looking at asterisk threads, having many of those is normal. |
| 12:49.09 | *** join/#asterisk longword (~paul@eventhorizon.sungardas.ie) |
| 12:51.54 | m_tadeu | I see....you look like you found such a problem....have any tip to solve it? |
| 12:52.46 | *** join/#asterisk Pan3D (~Pan3D@63.208.160.190) |
| 12:52.47 | Dovid | hi. looking at a sip trace. the following means that asterisk supports sip session timers ? "Supported: replaces, timer" |
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| 12:55.57 | kaldemar | Dovid: if asterisk sends it, yes. |
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| 12:59.53 | |Physis| | how to make calls using chan_agent?? |
| 13:01.11 | Dovid | kaldemar: Strange because I have in sip.conf "session-timers=refuse" |
| 13:01.44 | *** part/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 13:02.12 | |Physis| | how change the source channel in the CDR record for this call to agent/agent_id so that we know which agent generates the call?? |
| 13:08.29 | kaldemar | Dovid: strange it is. mine does not have timer in supported header when there is session-timers=refuse in [general]. |
| 13:09.57 | kaldemar | |Physis|: the source channel in CDR is read-only. you need to use some other method if the source channel is not useful. |
| 13:11.11 | |Physis| | in agents.conf updatecdr thus this |
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| 13:13.35 | |Physis| | I do not know how to make outbound calls per agent in agentlogin or another way |
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| 13:47.28 | bobg | our asterisk pbx is plagued with rashes of dropped calls. I don't get anything interested in the asterisk logs. I am trying to figure out a strategy to capture more information on how each call is terminated so that I can narrow the problem down. (It a pretty busy pbx so I don't think that I can enable full SIP logging). Any ideas on how I can do this? |
| 13:47.50 | bobg | s/interested/interesting/ |
| 13:49.33 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:49.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:49.42 | *** join/#asterisk c4rg (crg@lagoon.freebsd.lublin.pl) |
| 13:49.55 | c4rg | anyone happened to have problems with strftime in 1.6? |
| 13:52.25 | c4rg | ${STRFTIME().. used in dialplan works OK, but when used as part of a file name when a call is being recorded doesn't work |
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| 14:00.03 | Dovid | kaldemar: can it be an issue with my * ? I am using 1.6.1.X |
| 14:00.33 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 14:04.22 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 14:04.33 | Aut0ExeC | hi guys... why is disa such a "security" risk?> |
| 14:04.44 | Aut0ExeC | i'm a nub here |
| 14:04.54 | Aut0ExeC | I dont get why if there is authentication |
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| 14:09.20 | Dovid | kaldemar: still here ? |
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| 14:11.42 | Aut0ExeC | hey is my Master.csv file is not created and asterisk complains that it is not there.... is it ok to create the file? |
| 14:13.30 | Dovid | Aut0ExeC: why not ? see if it works |
| 14:13.58 | |Physis| | como realizar chamadas com o chan_agent? ou como fazer funcionar o updatecdr no queues.conf? |
| 14:14.05 | |Physis| | how to make calls with chan_agent? or how to operate in the updatecdr queues.conf? |
| 14:15.10 | *** join/#asterisk n3hxs (~ed@63.68.135.4) |
| 14:15.32 | Aut0ExeC | k |
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| 14:19.13 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 14:29.03 | Dovid | kaldemar: My issue seems to be realted to bug 17005 |
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| 14:50.09 | asterisk-learner | Hi, can I make use of AgentLogin() application using a zoiper account ? |
| 14:55.42 | *** join/#asterisk Goshen (~Goshen@c-174-52-7-122.hsd1.ut.comcast.net) |
| 14:55.43 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:57.05 | malcolmd | zoiper will need to register as a sip peer to asterisk. from there, you could use agentlogin to create an agent capable of receiving calls from a queue that you create |
| 14:58.59 | *** join/#asterisk txwikinger (~quassel@sblug/member/txwikinger) |
| 14:59.45 | asterisk-learner | i am using iax2, created a context agent in extensions.conf where i have smthg like that : exten => _XXXX,n,AgentLogin(${EXTEN:2},s) |
| 14:59.52 | asterisk-learner | first priority is a NoOp().... |
| 15:00.11 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 15:00.22 | asterisk-learner | cant i make zoiper login using AgentLogin() without defining a queue ? |
| 15:01.13 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 15:03.05 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 15:05.17 | malcolmd | agentlogin is used as a frontend application to register agents who are members of queues, as defined in queues.conf. |
| 15:07.19 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
| 15:07.29 | tompaw | Hi. |
| 15:08.06 | tompaw | I'm having problems with Action: Originate. Even though I use Async: 1, it still seems to be queuing the calls instead of starting all of them at once. |
| 15:08.22 | tompaw | The only difference with Async is that I receive a 'successfully queued' message right away. |
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| 15:25.23 | benngard | shouldnt I to Dial prevent COLP changes? or am i wrong? |
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| 15:35.57 | JasonL | is "pri show span" still a command in 1.6.2.17 ? |
| 15:36.48 | jdoe | topic joke is stale now ;) |
| 15:41.19 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- 1.8.3.2 (2011/03/17), 1.6.2.17.2 (2011/03/17), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 15:41.24 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3.2 (2011/03/17), 1.6.2.17.2 (2011/03/17), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 15:41.28 | Qwell | *ahem* |
| 15:42.46 | chazzam | sputters |
| 15:43.47 | chazzam | did dahdi 2.4.1 get recalled then? |
| 15:47.09 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 15:49.17 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 15:50.22 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3.2 (2011/03/17), 1.6.2.17.2 (2011/03/17), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.1 (2011/04/01), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 15:58.09 | *** join/#asterisk Trengo (~roar@mail.pt.clara.net) |
| 15:58.41 | Trengo | hi, is it possible to block outgoing calls per destination country? |
| 16:00.34 | tzafrir_laptop | ~ping |
| 16:00.35 | infobot | ~pong |
| 16:00.42 | tzafrir_laptop | ~book |
| 16:00.42 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 16:01.46 | *** join/#asterisk garymc (~chatzilla@host81-139-133-67.in-addr.btopenworld.com) |
| 16:01.59 | tzafrir_laptop | the bot no longer answers private messagees? |
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| 16:07.04 | benngard | can any CONNECTEDLINE guru take a look at: http://pastebin.com/rN4ZUzJx i am dialing 959 and wathching Display of caller phone |
| 16:08.11 | benngard | is it correct behavior? |
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| 16:17.31 | Aut0ExeC | weird... I specify a password in my sip.conf but I can use the sip account without putting in the password |
| 16:17.45 | Aut0ExeC | anyone know why? |
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| 16:23.23 | ironm | Hello Ihor .. give me a second please |
| 16:25.45 | Aut0ExeC | is DISA a built in feature or do I need to install something else? I'm getting "no application DISA" :( |
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| 16:27.58 | ChannelZ | It's in app_disa.so |
| 16:28.11 | Aut0ExeC | oh i see |
| 16:28.17 | Aut0ExeC | i'll check to see if I have that module |
| 16:28.18 | ChannelZ | module load app_disa |
| 16:28.20 | Aut0ExeC | i'm running on openwrt |
| 16:28.22 | Aut0ExeC | ok |
| 16:28.22 | Aut0ExeC | thanks |
| 16:28.31 | ChannelZ | and/or check your asterisk/modules.conf |
| 16:28.54 | Aut0ExeC | no app_disa :( |
| 16:29.43 | ChannelZ | hmmm. well honestly it doesn't do anything you can't do yourself with your dialplan and a separate context |
| 16:29.53 | ChannelZ | you just don't get a config file |
| 16:30.08 | Aut0ExeC | ChannelZ: what do you mean...? |
| 16:30.19 | Aut0ExeC | and btw.. I dont have that module |
| 16:30.47 | Aut0ExeC | ChannelZ: sorry i'm a nub here |
| 16:30.55 | ChannelZ | All DISA does is let someone dial another extension 'protected' by a password. You can program a similar setup yourself just in the dialplan, it just takes a little bit of extra work |
| 16:31.20 | Aut0ExeC | ChannelZ: i see.... i'll have to research that... thanks |
| 16:31.32 | ChannelZ | and did you build this * yourself or is it a package or something that comes built for openwrt? |
| 16:31.48 | Aut0ExeC | yeah came built in openwrt |
| 16:33.16 | Aut0ExeC | i'm looking for a package that might add that module |
| 16:33.31 | *** join/#asterisk topriddy (~Seamfix@41.58.99.4) |
| 16:33.44 | topriddy | hello people... |
| 16:33.56 | ChannelZ | odd that it's left out. I guess for space assuming it's running on a hardware device with almost no storage |
| 16:34.02 | Aut0ExeC | yeah |
| 16:34.06 | topriddy | wrote a ussd app, need suggestions for a simulator |
| 16:34.07 | Aut0ExeC | i know |
| 16:34.25 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 16:34.37 | Aut0ExeC | ChannelZ: i'll look up your suggestion tho... thanx |
| 16:34.44 | topriddy | actually i have simulated it in LeibICT gateway simulator and it works fine, but the phone visual simulator wont run it and keeps crashing so |
| 16:34.56 | topriddy | was thinking of maybe someone knows another i can test |
| 16:35.04 | *** join/#asterisk onixx (1000@bas1-stetherese38-2925261214.dsl.bell.ca) |
| 16:35.47 | onixx | hello; I am having issues with jitterbuffer on asterisk 1.4 and app alarmreceiver |
| 16:36.20 | topriddy | nobody to help? |
| 16:36.36 | onixx | I have a local ATA dialin to asterisk app_alarmreceiver, tones sent from alarmreceiver are interrupted |
| 16:36.54 | onixx | how do I disable the jitterbuffer ? |
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| 16:51.38 | paulc | onixx: which ATA? and what format? |
| 16:55.06 | onixx | grandstream ht502 |
| 16:55.09 | onixx | pcmu |
| 16:55.28 | onixx | the same configuration currently works with an old ata496 |
| 16:56.39 | onixx | when I manually dial app_alarmreceived using a phone, from ata496, I can report an alarm by dialing the tones and I can hear the kissoff tone returned after I enter the digits. uninterrupted |
| 16:56.58 | onixx | doing the same with ht502, the kissoff tone is interrupted |
| 16:57.21 | onixx | I suspect the ht502 supports a feature the the 496 did not |
| 16:59.13 | onixx | paulc: I have tried different setting on jitterbuffer on the ht502 with no difference. the 496 does not even have these settings available |
| 17:00.37 | paulc | onixx: Hmm.. it's been a while since I played with this stuff.. is kissoff any way related to a DTMF tone? Maybe configure the ATA for inband DTMF using G711 so that it passes all audio through untouched (rather than clamping DTMF and sending RFC2833 packets) |
| 17:02.24 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:03.04 | onixx | paulc: kissoff is a DTMF... however, I do have the same chopping issue if I listen to my music on hold from the ht502 |
| 17:03.33 | onixx | no issues when calling out another extension |
| 17:03.40 | paulc | onixx: music getting interpreted at DTMFs that you then hear? or... I'm confused |
| 17:06.16 | onixx | paulc: no no... I am just saying that the same type of interruptions in audio are heard when I dial app_musiconhold from the grandstream ht502 |
| 17:07.06 | paulc | onixx: would it be cruel to suggest not using Grandstream? (personal bias: I've played with them, didn't like them, generally use Linksys/Cisco ATAs (like the PAP2T-NA) and never had any problems with them) |
| 17:10.03 | onixx | paulc: looks like this is the issue I am having http://forums.digium.com/viewtopic.php?t=15577 |
| 17:11.22 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 17:11.49 | onixx | exit |
| 17:12.17 | onixx | pailc: maybe I should upgrade from 1.4.29 to 1.4.40... |
| 17:12.39 | onixx | paulc: thanks |
| 17:12.41 | paulc | onixx: Or go to a 1.6 or 1.8 release? ;-) |
| 17:13.08 | onixx | paulc: ;-) not sure about this... It was a nightmare when I went from 1.2 to 1.4 |
| 17:14.31 | paulc | onixx: haha yeah, I hear you.. but at 1.4 you're slipping behind a bit.. I guess the thing to do really is build a lab box, with a new version, and have a play with all your stuff to see what the changes/differences are |
| 17:16.06 | onixx | paulc: true... gotta leave now... going to the applestore to pickup ipad2 |
| 17:16.16 | paulc | onixx: enjoy! :-) |
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| 17:23.22 | Schreiber1337 | Hello... All... does DEVICE_STATE still work in 1.8.x? |
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| 17:54.04 | *** join/#asterisk voila (IceChat77@117.212.76.243) |
| 17:54.58 | voila | hi |
| 17:55.22 | voila | well , how can i receive digits (alphanumeric) from user over ZAP channel ?? i mean which command of asterisk will work for me ? |
| 17:58.00 | malcolmd | the read application? |
| 17:59.33 | voila | malcolmd: r u asking from me or telling me ?? :) |
| 17:59.48 | malcolmd | have you considered it already and found it lacking? |
| 18:00.32 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 18:00.58 | malcolmd | https://wiki.asterisk.org/wiki/display/AST/Application_Read |
| 18:05.27 | Schreiber1337 | Anyone having problems with DEVICE_STATE always returning UNKNOWN on 1.8? |
| 18:18.31 | *** part/#asterisk sbszulu (~dundubala@41.15.157.155) |
| 18:20.06 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 18:20.26 | Aut0ExeC | hey anyone have a nice disa repacement example? like perhaps using authentication? |
| 18:20.30 | Aut0ExeC | etc |
| 18:20.32 | Aut0ExeC | etc |
| 18:20.41 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
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| 18:32.28 | Kobaz | do sip session timers help clean up calls that weren't properly ended? |
| 18:32.54 | Kobaz | like if a channel is still up in asterisk but the far end actually hung up |
| 18:37.32 | Aut0ExeC | Kobaz: u using a card? |
| 18:37.54 | Aut0ExeC | I had a similiar problem with my cisco ATA spa3102... |
| 18:38.34 | Aut0ExeC | had to mess aroudn with the disconnect tones |
| 18:39.17 | Kobaz | sip |
| 18:39.31 | Kobaz | polycom to a non polycom sip endpoint |
| 18:39.33 | Aut0ExeC | oh ok |
| 18:40.34 | Aut0ExeC | oh ok that case no sorry bro |
| 18:41.35 | anonymouz666 | Kobaz: set the rtptimeout in sip.conf |
| 18:43.03 | Kobaz | is that global or per-peer |
| 18:43.40 | Aut0ExeC | anyone here have a nice alternative to DISA? like a dialplan with authentication? |
| 18:44.20 | benngard | any1 here, more than me, who runs a h323 trunk from avaya cm to asterisk? |
| 18:44.21 | Kobaz | anonymouz666: i'm having a problem with calls in queue keeping waiting forever because people hang up but the channels are still around, so the agents are in use |
| 18:46.08 | Kobaz | i had to soft hangup channels that were going for like 60 hours |
| 19:01.02 | wdoekes2 | Kobaz: session timers should help.. but I have issues too, with sip calls that fail to end |
| 19:01.18 | wdoekes2 | (on 1.6.2) |
| 19:02.18 | wdoekes2 | I haven't pinpointed if there is a particular cause (e.g. that session timers are disabled by one or both ends) |
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| 19:06.01 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:06.03 | Kobaz | i don't even know if polycom supports session timers |
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| 19:06.45 | leifmadsen | anyone do much with pickup and callgroups? Just trying to make sure I understand: callgroup you assign to channels that you want to pickup if ringing, and a pickupgroup is assigned to a phone to give permission to pick up phones in a callgroup? |
| 19:07.01 | leifmadsen | (I found a deadlock with this setup somehow though, but want to make sure I'm doing it right as well) |
| 19:07.20 | Kobaz | i would think it shouldn't deadlock no matter what setup you configure |
| 19:07.45 | leifmadsen | ya well you'd think so :)( |
| 19:07.58 | leifmadsen | I can reliably reproduce it on this customers machine |
| 19:08.03 | leifmadsen | regardless, I can |
| 19:08.05 | leifmadsen | can |
| 19:08.16 | leifmadsen | can't get it to work, and just want to verify I'm setting it up right |
| 19:08.22 | Kobaz | i dont use pickup and callgroups since it's technology dependent |
| 19:08.24 | Kobaz | so i wrote my own |
| 19:08.24 | leifmadsen | (not used to this keyboard :)) |
| 19:08.38 | leifmadsen | that is less than helpful for me at the current time |
| 19:08.43 | Kobaz | yeah, heh |
| 19:08.43 | leifmadsen | but it's nice to know you did that |
| 19:09.18 | Kobaz | yeah i really don't know pickupgroups, used them once |
| 19:09.42 | _Corey_ | leifmadsen: I can't find an example where our values for callgroup and pickup group are not the same |
| 19:10.06 | leifmadsen | :) |
| 19:10.17 | _Corey_ | we do use them in several sites |
| 19:10.19 | leifmadsen | right but I want to verify which one controls what functionality |
| 19:10.27 | russellb | holy crap it's leifmadsen |
| 19:10.34 | russellb | leifmadsen: i don't think I've talked to you in a month |
| 19:10.36 | Kobaz | ;callgroup=1,3-4 ; We are in caller groups 1,3,4 |
| 19:10.37 | Kobaz | ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5 |
| 19:10.37 | leifmadsen | one is to give permissions to pickup, the other is for what to pickup |
| 19:10.40 | Kobaz | i'm sure you've read that |
| 19:10.43 | leifmadsen | yes |
| 19:10.50 | leifmadsen | sometimes documentation is incorrect :) |
| 19:10.52 | leifmadsen | russellb: omghai! |
| 19:10.53 | _Corey_ | my understanding is that the callgroup is what you're in and pickup is what you CAN pick up |
| 19:10.54 | Kobaz | hah, yeah |
| 19:11.07 | leifmadsen | ya ok so I got it setup right then... too bad it doesn't work :) |
| 19:11.13 | leifmadsen | (at least not with SIP channels it seems) |
| 19:11.25 | Kobaz | leifmadsen: could be a bug in that version |
| 19:11.40 | leifmadsen | I just get something like: [Apr 4 15:03:37] NOTICE[19811]: chan_sip.c:21650 handle_request_invite: Nothing to pick up for 8df57d3c-5754e65f-46ec9f92@192.168.23.133 |
| 19:11.46 | leifmadsen | ya, 1.8.4-rc2 |
| 19:11.57 | Kobaz | a lot of stuff is broken for me in 1.8 |
| 19:11.59 | Kobaz | try 1.6.x |
| 19:12.03 | leifmadsen | nope |
| 19:12.03 | _Corey_ | all of my production servers are running something <1.8 |
| 19:12.05 | Kobaz | see if it works the way you expect it |
| 19:12.11 | leifmadsen | can't do that here |
| 19:12.17 | leifmadsen | will have to try some other day |
| 19:12.19 | Kobaz | but like, try it in the lab |
| 19:12.30 | leifmadsen | yep, some other day when I have a lab setup |
| 19:12.32 | Kobaz | and see if it's just set up wrong, or it actually doesn't work in 1.8 |
| 19:12.38 | _Corey_ | :) |
| 19:12.40 | leifmadsen | I'll just have to tell them they can't have that functionality right now |
| 19:12.49 | Kobaz | i was playing with 1.8.3 the other day, allowtransfer doesn't work |
| 19:12.54 | leifmadsen | unless I come up with a different way of grabbing the channel like with Bridge() or something |
| 19:12.57 | Kobaz | but it works in 1.6.0 |
| 19:13.05 | Kobaz | i should do a bug report |
| 19:13.12 | _Corey_ | leifmadsen: Maybe directed picku? |
| 19:13.14 | leifmadsen | Kobaz: yes I know the steps to determine if something works or not :) |
| 19:13.18 | _Corey_ | er pickup |
| 19:13.32 | _Corey_ | we use that a lot and it seems to work well in 1.8 |
| 19:13.33 | Kobaz | leifmadsen: haha, yeah... just throwing it out because i have nothing else to contribute |
| 19:13.53 | _Corey_ | presumes you know which extension is ringing, etc |
| 19:14.32 | leifmadsen | _Corey_: ya I'll have a group of 3 phones that could be ringing, so I'll know what to pickup |
| 19:14.34 | leifmadsen | I'll look into that |
| 19:14.39 | leifmadsen | may be a work around for now |
| 19:32.55 | benngard | any that can explain for me how CONNECTEDLINE should work? differnt phones gives me different result |
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| 19:40.10 | Schreiber1337 | Anyone using DEVICE_STATE successfully in 1.8 |
| 19:42.43 | benngard | yes |
| 19:43.08 | benngard | i use it the whole time |
| 19:44.06 | Schreiber1337 | benngard: I have callcounter=yes call-limit=20 in sip.conf and I still get "UNKNOWN" returned no matter what I do... is there anything else I need to look at? |
| 19:44.16 | benngard | hint :) |
| 19:44.25 | benngard | sec |
| 19:44.50 | benngard | exten => 0317998975,hint,SIP/0317998975 <- somthing like that |
| 19:45.21 | benngard | and then try core show hints |
| 19:45.47 | Schreiber1337 | Yep... I have a "subscribecontext=SIPhints" that adds that in for every extension... |
| 19:46.08 | benngard | what does core show hints say? |
| 19:46.34 | Schreiber1337 | And I get " 4490@SIPhints : SIP/4490 State:Idle Watchers 0" for each extension |
| 19:47.48 | benngard | and when u try DEVICE_STATE u are in contecxt SIPhints? |
| 19:48.12 | Schreiber1337 | Hmmm... no... |
| 19:48.20 | Schreiber1337 | Let me try including that... |
| 19:48.28 | benngard | sec, lemme check syntax |
| 19:49.04 | Schreiber1337 | Not it... I am already including it... |
| 19:49.39 | leifmadsen | I just used DEVICE_STATE() and I don't have any hints setup and it works fine |
| 19:49.59 | leifmadsen | just enabled callcounter=yes and then did NoOp(${DEVICE_STATE(SIP/0004f2040001)}) |
| 19:50.03 | leifmadsen | returns the state just fine |
| 19:50.04 | Schreiber1337 | SayDigits(${DEVICE_STATE(SIP/${EXTEN})}) returns UNKNOWN |
| 19:50.29 | leifmadsen | I'm using 1.8.3.2 |
| 19:50.37 | leifmadsen | beyond that not sure what to tell you |
| 19:51.22 | benngard | plz do use noop and paste result, dont know what saydigits will do |
| 19:52.13 | leifmadsen | russellb: hey thanks for the call completion recipe! It just came in very handy :) |
| 19:53.41 | benngard | lol, just realized some has hacked my asterisk, SVN-may-ooh323_ipv6_direct_rtp-r311741MS :) |
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| 19:56.22 | Schreiber1337 | benngard: Executing [s@macro-InternalSPA942woVoice:1] NoOp("SIP/4328-00000038", "UNKNOWN") in new stack |
| 19:56.55 | benngard | show me dialplan row plz |
| 19:57.29 | Schreiber1337 | exten => s,1,noop(${DEVICE_STATE(SIP/${EXTEN})}) |
| 19:58.46 | Schreiber1337 | is it formatted wrong? |
| 20:00.25 | benngard | try: exten => 4490,1,NoOp$({DEVICE_STATE(SIP/4490)}) |
| 20:00.44 | benngard | an dial 4490 ofc |
| 20:03.20 | benngard | exten => 4490,1,NoOp(${DEVICE_STATE(SIP/4490)}) |
| 20:05.51 | Schreiber1337 | benngard: That works... I must have something wrong in my syntax in my macro... thanks.... |
| 20:07.16 | benngard | np, i did a spelling error myself at first attempt |
| 20:08.41 | benngard | P-Asserted-Identity, thats COLP update correct? |
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| 20:21.37 | Schreiber1337 | benngard: Yep... working great... thanks agin! |
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| 20:29.11 | benngard | Schreiber1337: u are welcome |
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| 20:35.15 | Aut0ExeC | can someone help me out with a DISA alternative to call in to my pbx and get a line to dial out locally? |
| 20:35.52 | Aut0ExeC | I was thinking something like Authentication, then Read, then Dial |
| 20:36.02 | Aut0ExeC | but i'm a nub with the dialplans.. |
| 20:39.54 | drmessano^ | Why wouldnt you use DISA? |
| 20:40.03 | Schreiber1337 | Aut0ExeC: What exactly are you trying to achieve... |
| 20:40.03 | drmessano^ | Thats what you're describing |
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| 20:46.21 | *** join/#asterisk mboylan (~mboylan@66.206.176.224) |
| 20:46.40 | Aut0ExeC | Well unfortunately my openwrt build doesnt have DISA |
| 20:46.51 | Aut0ExeC | I would like to call in to asterisk to get a line to call out locally |
| 20:47.29 | Aut0ExeC | but since I dont have DISA , I understand that I can make my own dialplan that will perform something that DISA can do |
| 20:47.40 | Aut0ExeC | is this correct? |
| 20:48.05 | mboylan | Hi guys... question for you. I've been using a kickstart file for CentOS to install asterisk on quite a few new servers here. Has been working well for weeks/months. As of today, though, there seems to be an issue with the Dahdi installation. It's almost like the yum repos are missing some dependencies after the latest update. None of the modules are installed and /dev/dahdi is never created. Anyone else seen this? |
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| 20:49.17 | tzafrir_laptop | mboylan, you miss the dahdi modules package |
| 20:49.26 | tzafrir_laptop | for the specific kernel |
| 20:49.31 | tzafrir_laptop | (newer kernel?) |
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| 20:50.18 | mboylan | it's all going through yum though -- shouldn't it resolve those sort of dependencies on its own? It worked on the systems that were already built and I yum update'd... but building new seems to be missing some |
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| 20:52.40 | dancarlson | Hello Asterisk Gurus. I have an asterisk box that will handle calls for branded sip clients using different domains. I'd like to set the domain info for outgoing calls in the SIP Header fields of From, Call-ID, and Contact dynamically per call in my AGI. I'm using Asterisk 1.8.2.3. Is this possible without modifying Asterisk? Any suggestions? |
| 20:52.48 | mboylan | in the post script section of the kickstart file, it's installing these: # Install third-party packages |
| 20:52.48 | mboylan | echo "Installing Asterisk..." /usr/bin/yum -y install asterisk18 asterisk18-core asterisk18-configs asterisk18-voicemail asterisk18-addons dahdi-linux dahdi-tools libpri asterisk-sounds-moh-opsound-wav |
| 20:54.05 | mboylan | dahdi-linux used to be enough to grab all that was needed. Seems to not be the case now? I tried installing the dahdi-linux-kmod too but that didn't help matters. Something broke repo side w/ the latest update, it seems... |
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| 21:32.16 | Goshen | Anyone have a sip.conf(polycom config file) for 3.3.1? I updated and it doesn't like my sip.conf anymore |
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| 21:51.17 | [hC] | Anyone know if there has been any activity in 1.6/1.8/trunk that provides a variable option to app_voicemail which currently checks to see if your voice mailbox password is set to your voicemailbox number that it prompts you to set it up? I'm looking for an override such that i can say 'if it equals $VAR' - willing to write a patch if it hasnt been added yet |
| 21:53.48 | [hC] | hm. crap, guess not |
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| 23:05.21 | root52 | Hey all. What is this telling me? "WARNING[20157] pbx.c: Don't know what to do with 'Local...." I have read all about Local channels and best I can tell is that this is there because. At some point we move to a SIP channel and this Local channel (that is used originate the call) get orphoned. Any Idea on how best to deal with this message. |
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| 23:19.13 | Greek-B0y | nickfennell: You were asking about elastix? |
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