IRC log for #asterisk on 20150218

00:00.49ChannelZ-WkThere was a security thing in October with SRTP so who knows what their package is (is lenny LTS of some sort?)
00:01.54*** join/#asterisk RobertLaptop (~rmiddle@74.112.203.154)
00:02.17jamicqueall debian version are LTS, however lenny is outdated :(
00:03.35jamicquelenny was released in 2009, support edned in 2012. I have one system where I would like to lunch webrtc and don;t like to update os
00:03.36jamicque:)
00:05.58ChannelZ-Wkresistence is futile
00:08.16jamicqueI'll first try installing openssl from source... maybe this would help...
00:13.11MaliutaLapjamicque: so don't re-install the OS, migrate the services off to something newer
00:14.43jamicqueMaliutaLap: I don't want to reinstall OS, the problem is that on this server there is notonly asterisk and I want to neable webrtc there. All my newer instances are running on 14.04 - on Docker! :)
00:15.32jamicqueHowever, I have one old lenny in my backpack... And I do not want to touch it harder than I should :)
00:34.49RobertLaptopI am looking to control asterisk using a web app.  Asterisk gui would have fit the bill but it delevelopment was dropped.  Is there anything that I can use instead?  And I don't think freepbx is an option as I need something that stays out of the way.  Freepbx does anything but that.
00:35.25ChannelZ-WkAsterisk + GUI + Stays Out Of The Way = Does not compute
00:40.18robmalRobertLaptop: There's XiVu and SAIL.
00:40.31*** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com)
00:41.07robmalWe have multiple sites running SAIL, some quite large so it might be worth taking a look for you.
00:42.38RobertLaptopChannelZ-Wk, The Asterisk GUI project was designed to stay out of your way.  All says were made direct to the dialplan files no DB or other middle layers in between.
00:42.51RobertLaptoprobmal, I will have to look at those.
00:43.57robmalThey both have their downsides, if you need any help /msg me
00:45.43RobertLaptoprobmal, OK. SAIL looks like a PBX project not a library that allows the processing of dialplans using AJAX type calls.
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00:50.05robmalYup, but they're starting to catch up with web2.0 as we speak ;-) If you need any features prepared, aside custom apps, they can put them in place quite fast.
00:55.26mbowieRobertLaptop: Let me know if you find anything promising... I took a quick glance last week (briefly) without success. (vi education for admins FTW!)
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01:32.58RobertLaptopmbowie, I understand.  The Asterisk QUI filled a nich that doesn't seem to be covered anymore.
01:35.00mbowieRobertLaptop: Agreed.
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02:06.56Borganyone about? I need help with 'No SMDI interfaces are available to listen on, not starting SMDI listener.'
02:10.18jab416171hmm... I'm trying to set it up so if I call from my windows PC (extension 130) it uses a different route than if I call from anything else.
02:10.29jab416171I thought it was just a matter of setting "route CID" but that didn't do it
02:10.42jab416171oh duh, I'm looking at outbound, should be looking at inbound
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02:19.01jab416171dang, it's still not working
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02:22.33*** mode/#asterisk [+o mjordan] by ChanServ
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02:27.23RobertLaptopmbowie, https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Asterisk+REST+API
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02:39.34Borganyone?
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02:48.04jab416171doesn't look like anyone is here Borg
02:50.07RobertLaptopI am sure there are people around but without knowning the question kinda hard to answer.
02:51.47WIMPyYes. might be hard to find anyone using SMDI. I can't remember it ever being mentioned here before.
02:51.48BorgRobertLaptop, this is the problem :  I need help with 'No SMDI interfaces are available to listen on, not starting SMDI listener.'
02:52.28RobertLaptopYes not even sure what SMDI even is.
02:52.33Borghmm... well the problem is I'm trying to get Asterisk to start on FreeBSD, but i'm not sure it's actually running, hung or what.
02:53.02jab416171how do I configure asterisk so, if I'm calling from extension 130 to any number, use Trunk X, but for all other calls, use Trunk Y?
02:53.09BorgSimplified Message Desk Interface (SMDI) is a protocol that defines the interface between a voice mail system and a phone system such as a PBX or public telephone switch. It was developed by Bell Labs. It is used to provide the voice mail system the information it needs to process the call.
02:53.33WIMPyBorg: So so where do you get so far?
02:54.18WIMPyjab416171: Use contexts. And Don't call devices "extensions". Extensions don't make calls.
02:55.05BorgWIMPy, well I compiled it from ports, did an initial configuration of /usr/local/etc/asterisk/asterisk.conf and tried to fire up the daemon. I've also been looking for a handy GUI to manage it via a web interface, but no success on that yet either.
02:55.40jab416171I guess a better way to put it: I want to use Trunk X for outgoing calls that originate from me (I call asterisk, and then call out to someone, or make a call from a softphone), and Trunk Y for all incoming calls from other people that then call my phones.
02:55.46jab416171WIMPy, what are contexts?
02:56.26WIMPyBorg: But you already got some output. So it at least starts (to start).
02:56.45WIMPy~book
02:56.45infobotAsterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
02:56.49WIMPy~primer
02:56.50infobotNew to asterisk configuration? Check out this primer to get started.  http://burner.com/asterisk-primer
02:57.09jab416171if it makes a difference, I'm using asterisk 11 and PIAF
02:57.11WIMPyjab416171: there you have two starting points on how to use Asterisk.
02:58.16WIMPyYes. We can't support any configuration utilities here.
02:58.59WIMPyThey tend to use very different terminology from Asterisk and are hard to understand from teh Asterisk point of view.
02:59.07BorgWIMPy, yeah. I'm talking with the Webmin devs to see if they might consider such a module. would be quite handy.
02:59.09jab416171understood
03:00.09WIMPyjab416171: So PIAF uses FreePBX, which is the topic of #freepbx.
03:00.24jab416171WIMPy, I tried asking in #freepbx but nobody's around
03:01.49WIMPyThat happens frequently, but asking here about FreePBX configuration won't work.
03:03.01jab416171got it
03:06.45WIMPyBorg: About that SMDI message: You can safely ignore that. But you still haven't told us how far you got otherwise.
03:07.02jab416171ah, I figured it out. I just need to change my dial pattern so it matches on caller id for the outbound route
03:07.27BorgWIMPy, I haven't gotten much further. the daemon just crashes with no other error. :-s
03:07.52WIMPyBorg: Then start it with -cvvvddd.
03:10.26BorgWIMPy, aha! now that was useful. '/usr/local/lib/asterisk/modules/pbx_lua.so: Undefined symbol "luaL_loadbuffer"
03:10.26Borg'
03:10.40Borgso LUA is a problem I see. interesting.
03:11.06WIMPyPretty interesting if you compiled yourself.
03:11.45WIMPyBut that should probably only keep the module from loading.
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03:27.13linociscowhat is asterisk SCF?
03:29.07jab416171anyone have a recommendation for a decent windows SIP softphone?
03:29.28jab416171I have zoiper right now and it keeps having problems with routing audio to the right source
03:30.04linociscojab416171, what about x-lite?
03:30.20jab416171haven't tried it
03:37.54jab416171thanks linocisco, it works a lot better than zoiper
03:38.09jab416171zoiper wouldn't even ring when I called it, and for some reason, it was registering itself with my public IP
03:38.41linociscojab416171, i was using 3CX before, but now it is only compatible with 3CX PBX system
03:40.21jab416171well that's silly
03:44.16jab416171yeah this client's a lot better, thanks again
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04:03.58jab416171can asterisk do SMS?
04:04.12volga629Hello Everyone, how to troubleshoot this issue
04:04.43volga629[2015-02-17 23:02:04] WARNING[1951]: chan_sip.c:3751 __sip_xmit: sip_xmit of 0x7ff014028960 (len 593) to clienpubip:1025 returned -2: No such file or directory
04:04.43volga629[2015-02-17 23:03:14] WARNING[1951]: chan_sip.c:3751 __sip_xmit: sip_xmit of 0x7ff014028960 (len 593) to cline pub ip:1025 returned -2: Success
04:04.47WIMPyjab416171: If you're talking real SMS, yes.
04:05.07jab416171WIMPy, "real" SMS?
04:05.17volga629this I only see with TLS clients
04:05.23WIMPyLike what's used in the PSTN.
04:08.39jab416171so, what clients support receiving SMS in that way?
04:08.44jab416171can I send an SMS to a cell phone?
04:08.59[TK]D-FenderIt has nothing to do with where it goes
04:09.03[TK]D-Fenderit's abot how it gets there
04:09.07[TK]D-Fenderabout*
04:09.16WIMPyIf you have access to an SMSc that supports it, yes.
04:09.41jab416171SMSc?
04:09.53jab416171does google voice count?
04:10.19WIMPyThe Short Message Service Center.
04:10.40WIMPyThe switch for SMs.
04:10.58[TK]D-Fenderjab416171, Forget GV
04:11.30jab416171why do you say that?
04:11.42[TK]D-FenderBecause they are pulling XMPP and are effectively dead
04:11.45[TK]D-FenderForget them
04:12.08jab416171I can send an SMS without relying on XMPP
04:12.19[TK]D-Fenderusing what to get to GV?
04:12.28jab416171http?
04:12.29[TK]D-FenderWhat would Asterisk have to speak to do this?
04:12.35WIMPyReal SMS involves a modem connection.
04:13.09WIMPyOff course there are http services and the like. But that has nothing to do with anything telephony related any more.
04:13.23jab416171that's true
04:13.35jab416171okay, how would I send an SMS with the PSTN?
04:14.08RobertLaptopWIMPy, smpp is much more used method for sending SMS.  I have no idea if Asterisk supports SMPP.
04:14.09[TK]D-Fender<jab416171> can asterisk do SMS?
04:14.09[TK]D-Fender<[TK]D-Fender> Sortof
04:14.09[TK]D-Fender<[TK]D-Fender> It can do SMS over E1, or via SIP MESSAGE if you have an ITSP that supports it
04:14.09[TK]D-Fender<[TK]D-Fender> Or via chan_dongle, etc
04:14.26jab416171right
04:14.35WIMPyWhat is SMPP?
04:14.37jab416171I didn't understand most of what you said
04:14.45WIMPycore show application SMS.
04:15.10RobertLaptopWIMPy, http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer
04:15.17jab416171compatible with BT SMS PSTN service in UK and Telecom Italia in
04:15.17jab416171Italy.
04:15.40WIMPyRobertLaptop: Is that a new thing?
04:15.42jab416171is it saying that's the only PSTN it's compatible with?
04:15.59WIMPyjab416171: ETSI
04:16.29jab416171ah
04:16.38WIMPyUgh. That looks horrible.
04:17.58RobertLaptopWIMPy, Not sure if Asterisk supports it.  That is the way all the carriers pass messages around.
04:18.28WIMPyNot something I came across so far.
04:20.31coppiceSMPP is one of several protocols uses to exchange SMS between carriers, and between large customers and carriers. its pretty much unmaintained now
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04:38.24mattslAnyone have any suggestions on what cause stuck calls?
04:42.26WIMPyThat's not enough information to make a comment.
04:42.41WIMPyAnd define "stuck".
04:45.58mattslSpecifically, I've got a server that becomes unusable after 1-3 days. I can't find anything consistent other than I have noticed a relatively high occurrence of SIP channels remaining open after the remote end has release the call. I'm trying to determine if that's a symptom or a cause.
04:47.31WIMPyAsterisk channels or sip channels?
04:48.19mattslSIP
04:48.36WIMPyHow many?
04:48.47mattslnot enough that I would think they would matter, 3 or 4...
04:49.02mattslAnd I see no significant change in CPU usage
04:49.10WIMPyThat sounds perfectely normal.
04:50.57mattslYeah. I figured that they weren't really an issue. But the server becomes so unresponsive that I can't even get it to let me ssh to it without several tries.
04:51.15mattslThat was the only thing different on it than I could see from any other server I have running smoothly
04:51.36WIMPySo that might be completely unrelated to Asterisk.
04:52.01mattslquite possibly
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04:53.17mattslSeeing as how it's the only thing running on the box and I saw at least one thing that was out of the ordinary with it, I started there. I'm thinking the stuck channels though are just a symptom of it dropping packets
04:53.40mattslBut I'm left wondering what else I could possibly troubleshoot...
04:53.51WIMPyLikely, yes.
04:54.12mattsl(It does have Zabbix, but that was added after the problem to try to get some sort of info about machine state when it occurred)
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04:54.48WIMPyLogin in and stay connected until it happens.
04:55.08mattsllike I said, that can be days...
04:55.30mattslAnd I don't know what I'd be looking for
04:56.17WIMPyYou can try to dump to process list to files every few minutes in the hope to find something.
04:58.23mattslYeah. I've been able to get into it when it was in the degraded state a couple times and saw nothing in top
04:58.34mattslI mean the CPU usage never exceeded 10%
04:58.50mattslIt's clearly something blocking, but I have no idea what
04:59.22WIMPyMaybe the network interface itself?
05:00.25mattslThat's what I was thinking. Could there be a problem with the interface on the software side? It seems like I'm able to hit it over the VPN tunnel for several minutes longer than through it's actual IP
05:00.47mattslBut that tunnel is on the same physical interface to which I can't connect
05:01.29WIMPyThat definitely looks like the way to investigate.
05:03.04mattslI've got it in production currently, so I'm about to just pull it, replace it entirely, and then let it run for days until it dies again. I'm just concerned that it might be something crazy unrelated to the box itself.
05:04.32mattslOr that without active calls going through it it won't ever actually lock up.
05:04.48WIMPyTry to do more tests on the network connection, like e.g. different packet sizes.
05:06.15WIMPyBut it's not a case of shooting yourself in the foot with something like fail2ban or so?
05:07.23mattslhow so?
05:07.34mattslYou mean with it actually blocking?
05:07.43mattslOr with it just screwing up?
05:07.53WIMPyBlocking
05:07.57mattslNope
05:08.32mattslIt's currently configured to look backwards pretty far, so a reboot wouldn't help if that were it.
05:08.41mattsl(like a week)
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05:39.16hebberUsing negative eventfilters in manager (AMI) doesn't seem to work - does someone know how to do the trick?
05:40.13hebber[TK]D-Fender: thanks for the tip using an Answer in Originate to avoid the unwanted CDR
05:42.55[TK]D-Fenderhebber, You're welcome
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06:45.42RaptorJesuscan anyone tell me why my asterisk is using 1.5gb of virtual mem?
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06:49.17hebberthats a lot
06:51.21hebberdo you refer to VIRT or virtual memory in virtualization?
06:54.10RaptorJesusuhm
06:54.11RaptorJesusvirt
06:54.28RaptorJesusand it keeps freezing
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07:07.45RaptorJesusok it freezes when i turn off a phone
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07:32.04RaptorJesusok yea wtf
07:32.09RaptorJesusi unplug a pjsip phone
07:32.11RaptorJesusserver crashes
07:32.40ChainsawIt isn't "surprise removal" compliant. It's not a USB port...
07:33.34RaptorJesusolder versions don't do this
07:33.55ChainsawIf you can make it occur at will, perhaps it's time to make a debug version explode so you can get a backtrace.
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07:34.46ChainsawBacktrace turns into bug report, and then either you fix it or someone else does. And all will be well with the world.
07:35.04RaptorJesushow does I do such?
07:35.30ChainsawDid you build your Asterisk from source or is it a distro package?
07:35.57RaptorJesusfrom source
07:36.10RaptorJesuswith freepbx
07:36.51ChainsawIn menuselect, you need these options: DONT_OPTIMIZE DEBUG_THREADS BETTER_BACKTRACES
07:37.26ChainsawRun ulimit -c unlimited before you start the Asterisk binary. You can process the resulting .core file with gdb.
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08:55.23Tujunope, nope - fromdomain issue remains.
08:56.26Tujuhttp://www.google.fi/search?hl=fi&source=hp&q=asterisk+outbound+invite+fromdomain
08:57.00Tujuhttps://issues.asterisk.org/jira/browse/ASTERISK-20841
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09:41.19stefan27Getting  19823 errors last day   res_rtp_asterisk.c: PJ ICE Rx error status code: 370401 'Unauthorized'.
09:41.58stefan27as well as 3 segmentations faults, many clients are using webrtc client with dtls encryption
09:42.28stefan27Im trying to start the debugging process, I'll post a core-dump:
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09:43.53stefan27Crash 1: http://pastebin.com/Z96fpCNJ looks dtls related
09:45.58stefan27using chan_sip, not pjsip
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10:00.40h8hi everyone
10:01.16h8what problema am I looking at if I'm trying to connect to an asterisk server behind nat, through a public ip but the registration doesn't get to asterisk yet I can make calls
10:01.31h8sip show peers doesn't show the peer as being logged in, yet I See the outgoing call
10:02.41roxh8: you can make calls TO it, or FROM it
10:02.52*** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain)
10:03.00h8FROM it
10:03.19h8so it's like the peer is connected, yet it doesn't show up and I can't call the peer
10:07.54roxh8: if you are behind a NAT, caling FROM it is not a problem, you need to configure your NAT to route SIP packets TO it
10:08.59roxh8: if this is the only asterisk behin this particular NAT, then you can probably just route all SIP traffic to it
10:09.25roxh8: is your router SIP aware?
10:10.26h8the firewall is a pfsense, firewall of the asterisk server that is
10:10.34h8it is the only asterisk, yes
10:14.13h8I'm connected through a vpn to that network, through there, the registration works, it just doesn't work through the public ip
10:21.15Tujuh8: you might need to figure out how your fw handles port 5060
10:21.27Tujuand how your phones behaves.
10:21.48Tujusome cisco's listen return packets in 5060 too, some models in high ports.
10:22.07Tujuformer are very hard to get to work.
10:22.51Tujusome phones allow you to set in config that it's behind nat.
10:23.00Tujusome allow setting public ip too.
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10:56.49Tujui'm trying to rewrite To header in dialplan but have not been able to.
10:57.27Tujuexten => 2226,2,SipAddHeader(To: "${EXTEN}@example.com")    should work, but it doesn't.
10:57.47Tujuthat 'Add' part smells suspicious to me.
10:58.03Tujui need to overwrite, not add anything.
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12:49.34tparcinaToday users are having small periods of time (second, maybe two) when they don't hear the other person (or the other person doesn't hear them).
12:50.10tparcinaWe record all phone calls, and when I listen for the phone call, I can hear both speakers fine.
12:50.57tparcinaI'm checking the network usage, and it's under 100 Mbit (tipicly 20-30 Mbit) on 1 Gbit network.
12:51.10tparcinaWhat could be the problem?
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13:01.21Tujuthey hesitate to talk knowing that everything is recorded? :)
13:01.50Tujui least it would make me stutter a bit.
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13:09.26roramirezHello, exist any way to compiling a specific module in Asterisk source?, i have source 1.8.25
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13:45.02RobertLaptopAnyone know when RHEL 7 RPM's are planed for Asterisk 13?
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14:17.51Tuju[TK]D-Fender: i had to hack today to make fromdomain to work.
14:18.45Tujui found quite many blogs like this http://mikepultz.com/2009/04/handling-sip-uri-dialing-in-asterisk/ someone else have fought the same issue.
14:20.16Tujunah, it broke again. it doesn't work.
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14:23.12Tujuif i do sip reload, it works for a while and after half an hour, it fails again.
14:23.37crised[TK]D-Fender: Hello there, I'm  having problems with the SIP phone. Outgoing calls goes ok, but DID incoming not. I changed a firewall setting (I'm using freebsd, similar to pfsense): (not to change the UDP source port), and it could receive DID inbound calls, but problem is that started to ring in the middle of the night, just because the telephone wanted to.
14:23.46crisedno one was ringing it.
14:24.12[TK]D-FenderYou're probably being scanned
14:24.40crised[TK]D-Fender: how's that
14:25.04[TK]D-Fendersome other hacker is looking for server's to dial out and your phone is accepting their call
14:25.24[TK]D-FenderYou're just reg'ing a Polycom to some outside provier right?
14:25.30[TK]D-Fenderprovider*
14:25.34crised[TK]D-Fender: to voip.ms only
14:26.03[TK]D-Fendercrised: There is a provisioning option to lock out calls from sourcees you have not registered to.
14:26.29[TK]D-Fendercrised: But IIRC you stopped doing it the way I told you and you started doing it directly on the phone itself
14:26.40crised[TK]D-Fender: so to only accept phone calls coming from voip.ms?
14:26.48[TK]D-Fenderyes
14:26.49crised[TK]D-Fender: yes, lol
14:27.03crised[TK]D-Fender: that would make lot of sense
14:27.11[TK]D-FenderOr you could try to just lock down SIP at your firewall itself
14:27.35crised[TK]D-Fender: what if I change the rule of static nat port?
14:27.59[TK]D-Fenderthat's what I said.  restrict the forwarding range
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14:28.52crisedI modified this rule:    nat on $ext_if from $localnet to any -> ($ext_if) static-port  and added these two:      rdr pass on $ext_if proto udp from any to any port 5060 -> 192.168.1.69
14:28.52crisedrdr pass on $ext_if proto udp from any to any port 5080 -> 192.168.1.69
14:28.52crised<PROTECTED>
14:29.05crisedI think the first rule, made all the difference
14:29.17crisedI added static-port at the end
14:29.47Tujucrised: you need to make another context where your phones are.
14:29.48crised"static-port    Withnat rules, the static-port option prevents pf(4) from modify-    ing the source port on TCP and UDP packets."
14:30.27crisedTuju: I only have one phone...
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14:30.47Tujuokay, well that's probably then okay to listen it ringing over night.
14:31.10crisedTuju: how do I do that?
14:31.40Tujuyou need to have more than one [name]   sections in extensions and context=name   in peer settings.
14:31.56crisedTuju: I don't follow at all
14:32.02crisedTuju: are you talking on the phone?
14:32.12Tuju?
14:32.29crisedTuju: are you talking about firewall config or phone config?
14:32.45Tujuthis is asterisk channel, i'm talking about asterisk configuration.
14:32.57crisedTuju: I don't have asterisk
14:33.08Tujuwell, my bad then.
14:33.08crisedTuju: asterisk is inside my voip provider
14:33.12crisedTuju: :)
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14:33.49Tujuone moment, we hack into their systems and fix it.
14:34.07crised[TK]D-Fender: so what do you suggest?
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14:40.13[TK]D-Fendercrised: I already gave you 2 ways.  take your pick.
14:41.05crised[TK]D-Fender: Can I do the config of lock out calls directly from the phone?
14:41.51[TK]D-Fender[09:26][TK]D-Fendercrised: There is a provisioning option to lock out calls from sourcees you have not registered to. <-
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14:42.04[TK]D-FenderThis is not the sort of thing I think they give you direct on the interface
14:42.35crised[TK]D-Fender: ok, so you advice to restric this rule?    Konsole output     rdr pass on $ext_if proto udp from any to any port 5060 -> 192.168.1.69
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14:43.54[TK]D-FenderFirewall them out or tell your phone to ignore them
14:43.57[TK]D-FenderTake your pick.
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14:45.10crised[TK]D-Fender: I prefer firewall, how would the rule look like? I'm onnecting to server tampa.voip.ms, this means only THIS server can access my phone?
14:46.07robscowI have transfers enabled in features.conf, when people press *2.  however, when we make calls to outside numbers, like a conference, and they ask for a pin, the moment we enter the code and press #, I heard "Transfer... sorry, that's not a valid extension" from our own Asterisk system.  is there something common/stupid Ive done to the configs?
14:46.32robscowfor external calls I simply have this...
14:46.47dan_jHi. As anyone seen a cheap SIP based loud ringer which is compatible with asterisk? Maybe a raspberry PI or arduino project if that works out cheaper.
14:46.54dan_jHas*
14:46.55robscowexten => _0[12378]XXXXXXXXX,1,Set(CALLERID(num)=20310)        same => n,Dial(SIP/${EXTEN}@spitfire,0,Tg)        same => n,Hangup
14:49.14[TK]D-Fendercrised: That's what I said.
14:49.39[TK]D-Fenderrobscow: What kind of phones are you using?
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14:51.04robscowpanasonic sip phones, KX-TGP500 to be precise
14:53.07[TK]D-FenderThose should support native transfers on their own without *-based DTMF features for this
14:55.29robscowI'm not seeing anything in the configs for the phones
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14:58.07crised[TK]D-Fender: How does this looks? rdr pass on $ext_if proto udp from tampa.voip.ms to any port 5060 -> 192.168.1.69
14:59.27[TK]D-Fendercrised: I have no idea how your firewall works.  You'll have to figure that out for yourself
14:59.57[TK]D-FenderRobWon't be in configs.  Typically it's fixed functionaly.  For ATA like devices it's usually hook-flash based
15:00.59robscowahh ok, so maybe i should remove the asterisk capability? and allow the handsets to do it themselves?
15:02.30[TK]D-Fenderyes
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15:11.34crised[TK]D-Fender: how can I make an anonymous call to see if my firewall settings are OK, now?
15:13.16crised[TK]D-Fender: how these hackers scanned my ip?
15:13.29[TK]D-Fendercrised: They threw a call at you.  Your mphone accepted
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15:14.24crised[TK]D-Fender: Why me?
15:14.34crisedThey are sniffing my ISP traffic or something?
15:15.02crised[TK]D-Fender: How can I make an anonymous call to see if my firewall blocks these?
15:15.08[TK]D-Fenderthey are scanning THE WHOLE INTERNET
15:15.23[TK]D-FenderGet some machine to throw a call at you.
15:16.05crisedwow
15:16.10Tuju[TK]D-Fender: what was the logic that is used to map incoming calls to defined trunks?
15:16.24Tujuyou mentioned something about the order how they're defined.
15:16.31[TK]D-FenderTuju: type= <---
15:16.55[TK]D-Fenderpeer = by IP, user = username, friend = both
15:16.55Tuju[TK]D-Fender: and if i've got more than one 'friend' ?
15:17.11crisedHow can I make anonymos call?
15:17.12[TK]D-Fenderfirst one it finds that matches gets it
15:17.23Tujuhmmm....maybe i should put user then.
15:17.28[TK]D-Fendercrised: Pick a SIP client and have it call out to your IP
15:19.08crised[TK]D-Fender: sighs... how can I do this online easily?
15:19.37[TK]D-Fendercrised: Go find some service or somebody to do the test.  Or remote control some other system on the outside
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15:22.25crised[TK]D-Fender: can you do this test?
15:22.54[TK]D-FenderNot currently
15:23.09crised[TK]D-Fender: ok, I'll give it sometime
15:23.11crisedthanks
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15:25.41Tuju[TK]D-Fender: I've two asterisk hosts, there are two trunks in between. So i cannot use ip, right?
15:26.04Tuju[TK]D-Fender: so either friend or user?
15:26.05[TK]D-FenderWhy 2?
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15:26.15Tuju[TK]D-Fender: two different context
15:26.23Tujutwo organizations.
15:26.37[TK]D-Fenderthen make user sections to match
15:26.59Tujuif i put type=user, sip show peers doesn't list them anymore and registration fails.
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15:27.42[TK]D-Fendersip show users <--------
15:27.44Tujuif i have type=firend, those trunks get mixed.
15:28.07[TK]D-FenderAnd if you know the IP of the opther box you shouldn't be registering anyway
15:28.37Tujuokay, they are listed in users, right.
15:29.14Tujucan i still have two trunks if i don't register?
15:29.49yun1989hello all
15:30.12[TK]D-FenderTuju: You can have as many as you want
15:30.13Tujuhi yun1989
15:30.25Tuju[TK]D-Fender: ack, good.
15:30.44yun1989someone know if is possible to connect CP 352 to asterisk ?
15:30.47Tujuso i should drop register => lines ?
15:30.52yun1989http://en.intelbras.com.br/business/intercoms/condominiums/stations/cp-352
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15:32.03[TK]D-Fenderyun1989: What does it speak?
15:33.44yun1989@[TK]D-Fender hello
15:34.16yun1989it's possible to connect the condominums analog system to asterisk ?
15:34.24yun1989in this case cp352
15:35.08[TK]D-Fenderyun1989: I just asked you a very direct question.  What does it SPEAK?
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15:35.50Tujuhmmm... it appears to work, this is scary.
15:36.15Tuju[TK]D-Fender: maybe you two don't speak common protocol :)
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15:38.03[TK]D-Fenderyun1989: That thing appears to be analog FXS/FXO so you could get an interface for * to speak to it
15:38.10yun1989http://en.wikipedia.org/wiki/Analog_telephone_adapter
15:38.41Tujuyun1989: cisco has ATA in SPA series.
15:38.42yun1989yes i need the gateway beteween the IP network to the condominuims network
15:38.55[TK]D-FenderSo go get an ATA
15:39.29yun1989but i think the problem is in cp352 don't support ATA
15:39.40[TK]D-Fenderif you need to use a PSTN port on the CP for connectivity, or a SIP>FXO gateway if you need * to act like a phone on it
15:39.50yun1989i need understand more this case
15:39.56[TK]D-FenderGo read its manual
15:40.00[TK]D-FenderWe don't use that here
15:40.28Tujuyun1989: ATA has plain analong telephone port in it, it does the ip.network <-----> analog   conversion.
15:40.50Tujuthose cost few tens dollars.
15:41.03Tujui've got one here.
15:41.43Tujuhttp://www.cisco.com/c/en/us/products/unified-communications/spa122-ata-router/index.html
15:42.17Tuju"2 voice channel ATA with router"
15:43.28[TK]D-FenderThat is if you need to take a PSTN port on the CP352 for this
15:43.40[TK]D-FenderYou'll need a different device if you need * to act like a phone to it
15:46.18yun1989thanks
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15:52.14Tuju[TK]D-Fender: you're probably only here who understands * in deep guts.
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15:53.34[TK]D-FenderNo, there are many here who understand far more than me.
15:53.41[TK]D-FenderI don't get into the actual code.
15:53.59[TK]D-FenderI do get the premise it's built around and how pieces interact in order to determine a path
15:54.05[TK]D-FenderThat's just linear thinking.
16:00.40coppiceThat's better that thinking like a log
16:01.16[TK]D-Fendermmmm log
16:01.42coppiceor thinking like a square
16:10.53filemoo
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16:16.11pjensen00thinking^2
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16:37.55human0x27Hello all, any jabber users?  We use jabber to send call information to call center employees.  I recently upgraded from 1.8.8.8 to 1.8.9.3.  I'm getting the following errors around every 2 minutes
16:37.59human0x27http://pastebin.com/raw.php?i=Fzng48Lg
16:38.36human0x27(debugging was enabled when I captured that output)
16:39.27QwellIs there a newline after the <?  That's crazy.  It should totally still work, but wat?
16:39.55human0x27yeah.. that confused me as well.. but it was happening elsewhere without causing an error.
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16:47.43mjordanthe fact that it is complaining about invalid XML is a sign that something isn't right with the world
16:49.10Qwellmjordan: libiksemel is pretty simple.  I could easily see a random newline breaking things.
16:58.49mjordanyup, but it isn't wrong at that point. That is invalid XML :-)
17:03.01Qwellhuh, xmlvalidation.com agrees.  I assumed spacing didn't matter at all.
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17:06.19Qwell'<' name (whitespace attribute)* whitespace? '>'
17:07.41human0x27so, the newline is the issue?
17:08.50Qwellassuming it's actually there, and something else isn't happening to make it bail early.
17:09.00human0x27hm.. interesting.
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17:26.35RPerreHello there. I'm using ARI and I'm trying to record a bridge. I make a POST /bridges/{bridgeId}/record and get a positive result. But when i try to stop and save (using POST /recordings/live/{recordingName}/stop) I get 404: Recording not found. Any help? thanks!
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17:28.57jamicqueHi, does anyone know if asterisk java supports AMI v2 ins Asterisk 12? Or is there any other java lib that works with AMIv2?
17:36.10mjordanRPerre: what are you using for the recordingName? And if you list all of the live recordings, does it list the one that you started on the bridge?
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17:38.24RPerremjordan, I'm using random strings for the name. I think there is no method to list recordings taking place, only stored ones.
17:42.33RPerremjordan, results: http://pastebin.com/JFhWFdWT
17:42.58RPerrei'm using curl for debug
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17:44.16RPerreon asterisk CLI:
17:44.17RPerre<PROTECTED>
17:44.17RPerre<PROTECTED>
17:44.17RPerre<PROTECTED>
17:44.17RPerre<PROTECTED>
17:44.17RPerre<PROTECTED>
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17:59.12RPerreI'm stuck on this recording stuff :/
18:19.43pabelangermight be the first person to do something with it
18:24.05RPerrepabelanger really? :o
18:24.25RPerreARI seems so elegant
18:24.32pabelangernew hotness
18:24.33RPerrewell...it IS elegant
18:24.46pabelangeronly know of a handful of people using it ATM
18:24.48pabelangermight you
18:24.57pabelangermore could be using, but not saying anything
18:25.08RPerrepabelanger, so you think this might me a bug?
18:25.15pabelanger*shurgs*
18:25.28pabelangerI'd check the testsuite to see if there is coverage for what you are doing
18:25.44pabelangerthen compare
18:25.50RPerrepabelanger, testsuite? i dont know what ur talking about :/
18:27.00pabelangerhttp://svn.asterisk.org/svn/testsuite/asterisk/trunk/tests/rest_api/
18:27.15pabelangerthere are some recording tests
18:27.21pabelangerso, you could see what they are doing
18:30.36RPerrethanks pabelanger!
18:36.15RPerrepabelanger, well it seems i'm doing things the right way. but getting same problem *facepalm*
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19:12.02matthew-moretalkanyone any ideas how I can set the Content-Type header for CURL Function Post Requests?
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19:17.28mbowieRobertLaptop: My scroll-back was too short (now fixed) but I see I had a mention... did you have any luck on the UI front?
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19:28.29KNERDis there a way to activate a feature code from the * CLI?
19:39.53ndbhi, is it possible to launch a thread with a socket binded without blocking from an asterisk module?
19:40.13ndbmy plan is to connect on this socket and send audio
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20:05.49*** join/#asterisk killfill (~killfill@190.107.178.50)
20:05.54killfillhi guys
20:06.24killfillim trying to connect a fanvil phone to my asterisk , using IAX. this is what i see in the logs: http://pastebin.com/raw.php?i=W27mdYFF
20:06.35killfillthere is no movment after that.
20:07.47killfillthis is the config: http://pastebin.com/raw.php?i=YCSwzx9w
20:07.59killfillis there any obvious thing im missing?..
20:12.37doopwhy are you denying then permitting the same thing
20:13.48killfillyeah, i copy pasted it from an already created extension.. the permitting seems to win, as i can connect just fine with zoiper tho
20:14.15doopit's silly.  take out the deny.
20:14.21killfillok
20:15.46killfilldoop: doesnt resolve the problem tho
20:16.37doophave you tried some other piece of hardware or softphone with the same credentials
20:17.08killfillyup, with zoiper works great
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20:17.19doopthen the problem is your fanvil phone
20:17.55killfillyah, i would thing so too, but this is the third one i test, and this product says it supports iax..
20:18.16killfilli know this is a chinees phone, so i dont expect the bet quality product.. but at least it should connect
20:18.18doopit may be a problem with the design of the phone
20:18.38doopexpect it to not work until it proves otherwise
20:18.40killfilltested 2 diff models
20:18.54doopfwiw i've never even heard of fanvil
20:19.00doopuntil now
20:19.10mbowieIt's only one "f" away from being an anvil
20:19.13killfillme neither.. its the only iax phone i could buy in here.. :P
20:19.18killfillheh
20:19.26doopwhy did you need it to be iax
20:20.31killfillbecause i got multiple vpns connecting to the same asterisk
20:20.46killfilland want to bypass the horror nat story of sip.. :)
20:21.05doopexplain
20:21.52Qwellkillfill: IAX, or IAX2?
20:21.52killfillas far as i know, to enable SIP on different networks (or behind a nat) i would need to setup a sip-proxy, right?
20:22.08killfillQwell: IAX2
20:22.21doopkillfill which part of the system is behind NAT exactly
20:22.28QwellDon't most of those phones also support SIP?
20:23.02killfilldoop: actually none. but the traffic flows thourhgt different networks. (different vpns)
20:23.38killfilldoop: when i connect them throught sip, only 1 party hears, not the other one. i think its the same problem as with nat, when your not passing thorght a sipproxy (?)
20:23.45killfillQwell: i wanted to avoid SIP
20:24.00QwellWhy?
20:26.03killfillQwell: could not make sip work. when establishing a call, one end doesnt hear the other one.
20:26.12QwellSo configure things properly.
20:26.51killfilltrying to. i thought iax would be another way to do it too.
20:28.20killfillyou guys think iax2 is no way to go?
20:30.22doopi think designing your network appropriately is the way to go
20:32.06killfilloh i would not expect that answear
20:34.56doopis any of your traffic running through a firewall?
20:35.03doopthat could also be the problem
20:36.34doopwhich party can hear and which party cannot?
20:36.36killfilldoop: yup, its running throught several firewalls, but the udp port is open , i have connectivity
20:36.52QwellThe UDP port?
20:36.52doopmore than one udp port
20:38.06killfillyup the udp port
20:38.16doopwhich udp port did you open
20:40.44killfill4569
20:40.52doopno, for SIP
20:42.20killfillaah
20:42.32killfill5060 tcp+udp, and 10.000 - 20.000
20:42.55doopi hope you didn't specify those with the decimal point
20:43.43killfillnope.. :P
20:43.50doopwhich party can hear and which party cannot?
20:44.00killfilli added them for knowing how much 0 i needed to write in here.. :P
20:45.49killfilli just re-test the setup, none of the parties hear the other one
20:46.12doopwell that's not exactly the problem you described earlier now is it
20:48.44killfillyeah i know. im sorry. i probably got confused, did much tests
20:51.07doopwhat happens when the box and the phone are on the same network without any firewall between them
20:53.23killfilli didnt test it, but it should work.. i mean, i got cisco phones working here just fine.
20:53.37dooplet's test it to make sure your phone works
20:54.06doopbut i'm seriously thinking your danfil phones are crap
20:54.13doopor whatever name
20:54.20doopfanvil
20:57.58killfillyeah, damn phone.. i would need to get it tomorrow at the other office
20:58.17doopyou're asking for help but you don't even have the phone available for testing?
20:58.44killfillyes i have it for testing, just remotely
20:58.52doopblinking
20:58.55doopok
20:58.56doopso
20:59.22doopwith all your VPNs and firewalls, presumably you can put the phone and the box on the same virtual network
20:59.51killfilli guess i was naive to think iax would just work from a different network as the asterisk is in...
21:00.02doopiax works fine
21:00.11doopit's the phone that appears to be cocked up
21:01.17doopyour naïveté perhaps comes more from expecting a crap phone to implement the protocol correctly
21:01.18killfilldamn phone.. i dont find similar problems on the internet either
21:01.34killfillheh
21:03.06killfilliax2 is just 1 port right?
21:03.13doopcorrect
21:03.34doopand your iax2 softphone works fine from the same network, yes?
21:03.49doop(the same network as the fanvil phone)
21:04.08killfillyup, your comming aobut my naive-ness is correct. its not phone not the protocol. sorry.
21:04.15killfillzoiper works fine
21:04.28killfills/comming/comment
21:04.54killfillaah.. 'its the phone not the protocol' sorry for the typos.
21:04.59doopso if zoiper works fine from that network then in all likelihood your hardphone is not working properly
21:05.30killfillyeah, i thought maybe there were additional iax config trick i could set in asterisk to make this work
21:05.32dooptest it on the same network and if you still have the same trouble contact the manufacturer or send it back
21:06.42killfillyeah.
21:15.20killfillso when peoapl talks aobut "iax2" there is only 1 version of the protocol right?
21:15.25killfillno iax2.1 or such
21:15.42doopafaik
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21:40.46WIMPyOh. You have an IAX hardware phone?
21:46.43*** join/#asterisk igeek_88 (~iweinberg@26-139-89-200.fibertel.com.ar)
21:47.23igeek_88Hello everyone. I need a program that will allow me to convert a wma file to WAV PCM Uncompressed' 8000hz 16bit mono in order to use it as an announcement on Asterisk freepbx.
21:48.10WIMPysox
21:48.32WIMPyand asking the same question to google will give you a lot of examples on how to use it.
21:49.06malcolmdi like how that same question was asked in #freepbx also
21:49.38igeek_88Yup. I've asked it myself hehe. Sorry.
21:49.53igeek_88Thanks.
21:49.57malcolmdnp
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22:27.24[[thufir]]when you "include" for a dialplan:  https://wiki.asterisk.org/wiki/display/AST/Include+Statements+Basics   what's getting included?  is it a file which gets appended during run time?
22:28.49[TK]D-FenderNo, CONTEXTS do
22:41.49[TK]D-Fender[[thufir]], Calls landing in [users] can also dial the things contained in that included context
22:42.41[[thufir]][TK]D-Fender: what I mean, pardon, is what's a context?  physically.  it's a script file?
22:42.59[TK]D-Fender[users] <- THAT is a context
22:43.07[TK]D-Fenderist is a logical separation in your dialplan
22:43.12[[thufir]]ok.
22:43.33[TK]D-FenderI want Joh to be able to dial certain things, but I don't want Mark to be able to dial those
22:43.37[TK]D-FenderJohn*
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23:09.48RobertLaptop<mbowie> https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Asterisk+REST+API
23:10.03RobertLaptopmbowie ^^^^
23:12.04mbowieRobertLaptop: Thanks for the replay... I did see that. We're on v11 for now, but we'll factor this into our upgrade motivation. ;-)
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23:25.12rolerI have an Elastix NLX4000 box with a Sangoma T1 card. It also has 4 fxo and 1 fxs port. Right now the system is working great with the pstn lines. We just had a T1 PRI  installed and the cutover date is Monday. I can't get the system to receive or make calls, and our ISP says there are errors on the T1. How can I debug these errors on the command line to see what is going on?
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23:32.29moy@roler you might want to do 'wanpipemon -i w1g1 -c Ta', assuming w1g1 is your T1 interface. You can check what other interfaces you have using 'ifconfig'
23:32.33moythen read this: http://wiki.sangoma.com/wanpipemon-T1-E1-line-alarms-0
23:33.22TrevizoliHi friends! Could someone explain me what means a message channel.c "Prodding Channel 'SIP/xxxxxx-00000000' failed". I searched in source code and find only that is something to make thinks moving...but that have failed...
23:34.04rolermoy; thanks. I'll read that URL. I have 2215 line code  violations :)
23:34.40moy@roler I think some are normal on startup, as long as they are not constantly increasing
23:36.47rolerok.. going up stairs to try once more.
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23:40.53rolermoy; all of my alarms are off. Rx level is >-2.5
23:42.16WIMPyroler: 'dahdi show status'
23:42.46rolerWIMPy; that looks okay too
23:43.04WIMPyAnd what doesn't ?
23:43.35rolerWIMPy; alarms OK, IRQ 0, bpviol 0, CRC 0, Fra ESF, Codi B8ZS, LBO 0 db (CSU)/0-133 feet (DSX-1)
23:43.45rolerthats the wanpipe1 card
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23:45.37rolercan I check anything else? :)
23:45.55WIMPyMake a call
23:46.24rolerOk. that sounds like a plan
23:46.27rolerthanks guys
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23:50.32[[thufir]]does "telnet 192.168.1.2 5060" show me whether the asterisk server at 192.168.1.2 is accepting connections on port 5060?  or that's done with netcat?
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23:51.00WIMPySIP usually uses UDP
23:51.07moy[[thufir]]: I guess that depends on whether you have configured your sip driver for tcp
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23:51.29[[thufir]]ok.
23:51.30moy[[thufir]]: you can use sipsak for a quick 'sip' ping over udp
23:51.41moyerr 'ping'
23:52.09[[thufir]]ahhhh, thank you.  wow, I've been trying to do that for a bit!

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