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| 00:02.14 | [TK]D-Fender | Brokedcomputer: that is still for basic DEVICE setup |
| 00:02.32 | Brokedcomputer | hahaha, as I said, like me giving you the bible and telling you to read it... |
| 00:02.38 | [TK]D-Fender | pointing a device to * is pretty minimal. PROCESSING a call is where the work is |
| 00:02.44 | [TK]D-Fender | extensions.con <- |
| 00:03.03 | [TK]D-Fender | You need JUST enough on basic device setup to get it so it CAN talk at all. |
| 00:03.03 | Brokedcomputer | Is that the header of the page for the table of contents? |
| 00:03.13 | [TK]D-Fender | then making it DO anything.... that is the dialplan |
| 00:03.15 | Brokedcomputer | I have absolutly no config yet |
| 00:05.11 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
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| 00:11.42 | Brokedcomputer | [TK]D-Fender - If you gave me a link to a page I should start at in that book online, I may just be able to get more out of your help, for now, your suggestions seem very distracting to me. |
| 00:15.09 | [TK]D-Fender | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceConfig_id216341.html |
| 00:15.18 | [TK]D-Fender | that does cover a chunk of the start |
| 00:15.30 | Brokedcomputer | Thats what I am working on now, |
| 00:16.25 | [TK]D-Fender | Then chapter 6 <--- |
| 00:19.06 | Brokedcomputer | Out of question, when I make a password for the phones Secret in Sip.conf, do I ever have to type that into the phone? Or how is that used? |
| 00:20.38 | [TK]D-Fender | that is a parameter in configuring it |
| 00:20.45 | [TK]D-Fender | not "as you dial" |
| 00:21.48 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 00:22.09 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 00:22.30 | [TK]D-Fender | Now just before you did any further, maybe you could give us an idea of what you want to have in the end so we can make sure there aren't any "gotchas" waiting rather than find out after spending a lot of time on it.... |
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| 01:07.39 | Brokedcomputer | That was very strange. Sorry there [TK]D-Fender |
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| 01:08.08 | Brokedcomputer | I lost all semblance of a DNS, it was very strange, and all I did was plug in the ethernet of my Cisco 7960 phone. |
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| 01:32.46 | K-Man`` | hi all, i am trying to use the extenspy application on Asterisk but getting unexpected results. Can someone help? many thanks in advance |
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| 01:57.29 | chamm75 | Good evening. Does anyone have experience setting up SIP trunking with flowroute and a pfSense firewall? |
| 01:57.41 | chamm75 | I'd be greatly appreciative for a little assistance. |
| 02:01.22 | chamm75 | I've got to get this trunk up and running by tomorrow morning. I've been at it all day, but I'm clearly missing something. |
| 02:01.43 | chamm75 | Or, I've got it set up exactly right, and it's not working for some other reason. |
| 02:02.07 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
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| 02:09.22 | chamm75 | Using Asterisk 1.8.11 behind a pfSense firewall. SIP trunk registers without error. I can dial the DID, and hear the IVR message, but it hangs up after 6.4 seconds every time. Asterisk gives the error: WARNING[1634]: chan_sip.c:3695 retrans_pkt: Hanging up call XXXXXXXXX_XXXXXXXX@4.55.17.35 - no reply to our critical packet |
| 02:09.48 | *** part/#asterisk K-Man`` (~dcr@syscom.lnk.telstra.net) |
| 02:10.01 | chamm75 | I sense that it has something to do with the media handoff, but I don't know why it's failing. |
| 02:10.21 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 02:10.21 | WIMPy | Looks like you aren't allowed to send a reply out to the internet. |
| 02:10.23 | chamm75 | I've got RTP ports defined in the rtp.conf, and those ports are forwarded to the PBX |
| 02:10.45 | chamm75 | I've got firewall rules allowing all LAN traffic outbound. |
| 02:10.48 | WIMPy | Nope. That pure SIP, no media. |
| 02:11.36 | chamm75 | Hmmm. Ok that's helpful. |
| 02:12.02 | chamm75 | But that IP address isn't the flowroute SIP server. |
| 02:12.09 | WIMPy | Maybe you have nf_nat_sip loaded, mangling your packets. |
| 02:12.11 | chamm75 | Does it have something to do with canreinvite settings? |
| 02:12.41 | WIMPy | That doen;t do anything if you do IVR. |
| 02:13.14 | chamm75 | I've googled that IP address, and several other web pages have people asking a similar question. I think it relates specifically to how flowroute handles SIP. |
| 02:13.27 | chamm75 | The trunk is set up to sip.flowroute.com |
| 02:13.49 | chamm75 | Which resolves to 216.115.69.144 |
| 02:14.38 | chamm75 | Would you mind taking a look at a SIP debug file? |
| 02:17.01 | chamm75 | http://pastebin.com/fitpxwKJ |
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| 02:20.02 | WIMPy | Wow. 4 vias. Never seen such thing before. But I see you have NAT enabled. That might be an issue. |
| 02:20.19 | chamm75 | I've tried disabling it. Same result. |
| 02:20.30 | chamm75 | Times out, no response 6400ms |
| 02:22.33 | chamm75 | flowroute has a really neat configurator that's supposed to write the PEER details. I've copied & pasted directly, as well as tried several modifications, including nat=no and canreinvite=yes. |
| 02:22.46 | chamm75 | Outbound calls seem to work fine. |
| 02:23.05 | chamm75 | Inbound call rings to the IVR, then immediately disconnects |
| 02:23.36 | chamm75 | I'm confused about line 43. |
| 02:24.17 | chamm75 | That RTP port is different every time, and it's not within the range that I've forwarded to my PBX |
| 02:24.42 | chamm75 | But I believe you when you say it's not a media issue. |
| 02:24.49 | ossip | do you know why Festival TTS stop reading text afeter a comma ',' |
| 02:24.52 | ossip | ? |
| 02:25.09 | chamm75 | I'm sure it's related in some way to my firewall, but I'm struggling to figure out how. |
| 02:25.29 | WIMPy | yout RTP ports are defined in rtp. conf. The remote ports are random. |
| 02:25.30 | chamm75 | I'm allowing all outbound traffic, automatically creating NAT rules. |
| 02:25.48 | WIMPy | But lack of rtp activity gives you a different message. |
| 02:26.06 | chamm75 | I've got 5060-5061 plus 16700-16899 forwarded to the PBX. |
| 02:26.25 | chamm75 | (When the source IP address matches flowroute) |
| 02:26.29 | WIMPy | You are not receiving a response to your SIP messages. It might be because they don't get to the peer. |
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| 02:28.00 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 02:28.19 | WIMPy | When you use nf_conntrack_sip you shouldn't need to forward RTP ports. |
| 02:28.33 | chamm75 | Is it possible that this is entirely the fault of the trunk provider? I swear it worked fine for, like, a day. |
| 02:28.57 | chamm75 | My firewall doesn't do any special SIP handling. (That's always been bad for me in the past) |
| 02:29.22 | chamm75 | I've even tried specifying a different SIP port. |
| 02:29.26 | WIMPy | As said above, nf_nat_sip will cause trouble. |
| 02:29.28 | chamm75 | port=5160 |
| 02:29.32 | chamm75 | Same result. |
| 02:30.27 | chamm75 | I do not have a nf_nat_sip in the config directory. Is there someplace I should be looking? |
| 02:30.53 | WIMPy | lsmod |
| 02:31.43 | chamm75 | Nope, not in there. |
| 02:31.53 | chamm75 | lsmod | grep sip returns nothing. |
| 02:32.14 | chamm75 | It's a pretty clean install of AsteriskNOW |
| 02:32.38 | WIMPy | On the firewall. |
| 02:33.06 | chamm75 | It's running BSD. |
| 02:33.17 | chamm75 | pfSense, which is based on FreeBSD |
| 02:33.37 | chamm75 | Fork of m0n0wall. I've used it for years, but haven't done SIP trunking through it. |
| 02:33.39 | Brokedcomputer | Does asterisk have a TFTP server, or would we need to install one? |
| 02:33.55 | WIMPy | Ok. Don't know what those features are called there. |
| 02:34.02 | chamm75 | Many distributions of Asterisk have TFTP server |
| 02:34.27 | Brokedcomputer | Cool thanks |
| 02:35.46 | chamm75 | AFAIK, there is no SIP ALG mod included in pfSense. |
| 02:36.53 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 02:37.01 | chamm75 | There is a SIP proxy module, sipproxd |
| 02:37.19 | chamm75 | Maybe I should look at installing that. Maybe it'll handle something that the firewall isn't handling. |
| 02:38.36 | chamm75 | http://doc.pfsense.org/index.php/Siproxd_package |
| 02:39.03 | chamm75 | Looks like maybe I should invest some time looking into that. |
| 02:39.23 | chamm75 | Like I said, I swear this worked for a day. |
| 02:42.55 | WIMPy | That might be overkill. |
| 02:44.45 | chamm75 | I believe it's the BSD equivalent of nf_nat_sip |
| 02:45.02 | chamm75 | In my previous experience, SIP algorithms have never been necessary. |
| 02:45.17 | chamm75 | But I've never had this much trouble setting up a trunk, either. |
| 02:45.19 | WIMPy | A proxy is another application. |
| 02:46.02 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 02:47.00 | chamm75 | Well, this goes back to, could this be a problem with the trunk provider? I can't imagine I'm the only person on Flowroute trying to use this configuration. |
| 02:47.05 | chamm75 | But maybe I am... |
| 02:48.01 | chamm75 | Also, siproxd is listed as BETA software. That's never a good sign. |
| 02:48.38 | WIMPy | Does tat really mean anything? |
| 02:49.04 | chamm75 | Seeing as how SNORT is listed as stable, and it's buggy as hell, not really. |
| 02:49.51 | chamm75 | So... I need to be looking at SIP. I'm not sure what all the VIAs mean. |
| 02:50.14 | ChannelZ | meh. My Asterisk 11 install seems to have lost its console color. |
| 02:50.26 | WIMPy | I think that means lots of proxies. |
| 02:50.36 | chamm75 | But you say that whatever is at 4.5.17.2 isn't responding to my SIP packets. |
| 02:50.42 | chamm75 | 4.55.17.2 |
| 02:51.50 | WIMPy | The PB has alreay expired. But That's not the IP I remember. |
| 02:52.07 | carrar | chamm75, why not move interface on the pbx to the DMZ so you don't have those issues? |
| 02:52.15 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-xsgerhdgnaqgafug) |
| 02:52.18 | carrar | move a interface that is |
| 02:52.29 | *** join/#asterisk camerin (hoax@newelite.bshellz.net) |
| 02:53.17 | chamm75 | Because the customer has only one public IP address. |
| 02:53.24 | chamm75 | So I have to forward ports. |
| 02:53.26 | *** part/#asterisk doulos1 (~bcalhoun@71-14-6-250.static.gwnt.ga.charter.com) |
| 02:54.49 | chamm75 | http://pastebin.com/WjF6BQSe |
| 02:56.18 | chamm75 | I just did a TCPdump on the Asterisk box itself, and I see two-way communication with that server. |
| 02:56.38 | WIMPy | You're replying to 216.115.69.144 |
| 02:57.04 | chamm75 | Which is sip.flowroute.com |
| 02:57.19 | chamm75 | Honestly, I have no idea where the 4.55.17.2 comes from. |
| 02:57.35 | chamm75 | It's specified in the INVITE request |
| 02:57.54 | chamm75 | I assume it's flowroute's equipment. |
| 02:58.02 | carrar | that means flowroute is using level3 for sip |
| 02:58.16 | chamm75 | But I do have a firewall rule allowing traffic from it. |
| 02:58.16 | carrar | we'll for the RTP part |
| 02:58.28 | WIMPy | Looks like that IP is used for media. |
| 02:58.31 | carrar | make sure your RTP is wide open |
| 02:58.56 | chamm75 | The RTP ports specified in rtp.conf are wide open. |
| 02:59.04 | chamm75 | 16700-16899 |
| 02:59.20 | chamm75 | Eventually, I'm going to have phones outside this firewall. |
| 02:59.29 | carrar | and wide open ipwise on the fw? |
| 02:59.34 | chamm75 | If I can get the SIP trunk working. :-/ |
| 02:59.41 | chamm75 | Yeah. |
| 02:59.49 | chamm75 | the SIP port is only open to a whitelist |
| 03:00.15 | carrar | make sure you are responding with the correct ip |
| 03:00.19 | WIMPy | That might be a hint. |
| 03:00.25 | chamm75 | which includes 216.115.69.144 |
| 03:00.36 | WIMPy | Maybe theyuse different IPs. Have you checked for misses on the FW? |
| 03:01.18 | Brokedcomputer | Hey Guys, I have configured the defaults of the extensions.conf and iax, and sip files, what do I need to do to get the Cisco 7960 IP phone to attach to the server? |
| 03:01.29 | carrar | bl, lunch |
| 03:02.02 | WIMPy | Brokedcomputer: defaults? And what firmware does your 7960 have? SIP or SCCP? |
| 03:02.26 | Brokedcomputer | I have previously had it set to SIP, but I am starting from scratch, and have the files needed for sip. |
| 03:03.46 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 03:03.53 | chamm75 | Am I looking for blocked inbound connections on port 5060? |
| 03:04.38 | WIMPy | yes |
| 03:05.06 | chamm75 | Nope, nothing blocked. |
| 03:05.47 | Brokedcomputer | Sorry, the defaults from this book, chapter 5&6 http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ |
| 03:06.53 | *** join/#asterisk lanning (~lanning@50-193-22-25-static.hfc.comcastbusiness.net) |
| 03:10.18 | chamm75 | Does UDP traffic from a source port of 5300 mean anything? |
| 03:10.35 | chamm75 | Some of that is getting blocked, but it's sourced from a 10.x.x.x network. |
| 03:10.52 | WIMPy | Possible. You don;t know what source port the peer uses. |
| 03:11.18 | chamm75 | This port 5300 keeps showing up, though. To a different "random" port. |
| 03:11.26 | chamm75 | I'll try allowing it temporarily. |
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| 03:12.26 | WIMPy | To a random port does not sound like somethig you're looking for. |
| 03:14.16 | chamm75 | That didnt' help. Still timing out. |
| 03:14.25 | chamm75 | The firewall is passing all traffic. |
| 03:16.18 | chamm75 | Does this offer any insight? |
| 03:16.20 | chamm75 | http://pastebin.com/hCgJBsm8 |
| 03:17.07 | chamm75 | This is killing me. :-/ |
| 03:17.33 | chamm75 | And flowroute support doesn't open until noon tomorrow, at which time I'll have a very pissed off customer. |
| 03:21.03 | ossip | Is it possible to use Festival TTS in early-media state? |
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| 03:27.59 | chamm75 | Ah, well, thanks for the help WIMPy. It's much appreciated. I'm going to hack away for a bit longer, then go to bed. Sounds like a support call to the provider might be the only way to resolve this. |
| 03:28.32 | chamm75 | I'm not convinced I'm doing anything wrong. :-) I've been at this for a little while, and while I'm not an expert, I'm not a novice, either. |
| 03:29.07 | WIMPy | Might help if you had an account with another ITSP to try. |
| 03:30.07 | chamm75 | Yeah. I might be switching to another provider. |
| 03:30.27 | chamm75 | Which is a shame, because the pricing model from flowroute is perfect for my customer. |
| 03:30.41 | chamm75 | I don't know of any other providers that price that way. |
| 03:30.52 | chamm75 | Flat rate, unlimited channels, pre-pay. |
| 03:31.42 | chamm75 | All the other providers I checked out limited concurrent connections. Or had a monthly rate per channel. |
| 03:32.02 | WIMPy | Makes sense. |
| 03:32.10 | chamm75 | My client has 11 months of complete downtime, where they might use 1-2 channels, then a month of chaos where they'll use 15 channels at once. |
| 03:32.27 | chamm75 | Doesn't make sense for them to pay monthly for 15 channels for the other 11 months. |
| 03:32.42 | chamm75 | So... I'll hope that flowroute support can do it. |
| 03:33.02 | WIMPy | If they know when they need the extra channels, I don;t see the issue. |
| 03:38.02 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 03:40.38 | chamm75 | I may look at alternate solutions if I can't get flowroute working. |
| 03:41.16 | chamm75 | They do know when their busy time is, but if I'm not around, I don't know that they'll want to turn up/down 13 additional channels. |
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| 04:10.33 | fireman_biff | My dundi/iax2 calls are occasionally failing with iax cause code 50 (no authority found). The secret used by the calling PBX matches dundi/secret in the database of the called PBX, and the username used by the calling PBX exists in the iax config on the called PBX. What else should I check? |
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| 07:22.24 | lkthomas | guys, after enable snmp on res_snmp.conf, do I need to restart asterisk ? |
| 07:27.08 | kaldemar | "module reload res_snmp.so" will probably do. |
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| 07:44.33 | lkthomas | kaldemar: it will not affect asterisk system right ? |
| 07:49.37 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:50.14 | bulkorok | hi |
| 07:51.54 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 07:53.13 | kaldemar | lkthomas: it will affect the snmp resource module and anything that may depend on it. |
| 08:06.21 | lkthomas | ok |
| 08:06.22 | lkthomas | thanks |
| 08:06.42 | lkthomas | http://www.voip-info.org/wiki/page_history.php?page_id=6405&source=current |
| 08:06.43 | lkthomas | this is stupid |
| 08:06.48 | lkthomas | how could I copy and paste like that |
| 08:09.09 | lkthomas | nevermind |
| 08:09.11 | lkthomas | found a workaround |
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| 08:22.10 | mirela666 | Does AsteriskGUI have own channel? |
| 08:23.03 | mirela666 | aha found it - #asterisk-gui |
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| 09:53.21 | sacoetzee | hey everyone |
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| 10:02.51 | sacoetzee | having an issue with a snom 370 and asterisk 1.8.15.1, where transfers are immediately bouncing back to the phone. I've reported on the snom forums too. Link to post info in pastebin http://pastebin.com/06b4gdJL |
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| 10:29.07 | bulkorok | when I use ReceivingFax, di I can force asterisk to send the T38 INVITE and not the sending side!? |
| 10:29.39 | bulkorok | it seems like fax receiving fails if the sender issues the T38 re-INVITE |
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| 12:00.56 | mathi | hi |
| 12:01.19 | mathi | before installing asterisk 11, shouldn't I uninstall my previous version? (10) |
| 12:02.08 | bulkorok | check configure --help for directory options... then you don't have to uninstall |
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| 12:38.15 | Rumbles | it took me 2 days but I finally got fax working over SIP lines!!! :D go fudge yourself 1840's technology |
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| 12:46.40 | ghost75 | (13:38:28) Rumbles: it took me 2 days but I finally got fax working over SIP lines <- that is so complicated? |
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| 12:52.02 | bulkorok | I would carry the printed paper by feet than faxing them |
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| 12:55.33 | ghost75 | still so many people use fax :< |
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| 13:15.17 | kannan | hello, I want to spawn an external script with System , but need to execute next priority immediately without waiting for the script to complete. the script is meant to interrupt the call after x seconds and will sleep until then |
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| 13:15.42 | kannan | with System, it goes to next priority only after the script is completed |
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| 13:16.48 | [TK]D-Fender | call it as a daemon then |
| 13:16.55 | [TK]D-Fender | & <- |
| 13:17.28 | kannan | oh ok thanks , will do |
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| 13:58.14 | Katty | ATTENTION |
| 13:58.19 | Katty | do you have espresso?! |
| 13:58.20 | latv | Hi! is there any updated tutorial how to succeed with WebRTC call? most peoplewho try it, get "no protocols out of 'sip' supported" error or get silence |
| 13:58.21 | Katty | NO?! |
| 13:58.25 | Katty | go get some. NOA |
| 13:58.27 | Katty | NAO |
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| 14:01.31 | tonikasch | Hi, do you know whcich package is asterisk's cdr_csv.so file included in on openwrt? |
| 14:01.44 | tonikasch | (asterisk 1.8) |
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| 14:24.38 | ghost75 | i have now included monitoring trunk delay time in nagios |
| 14:25.10 | ghost75 | 100ms is still ok time? |
| 14:26.23 | ghost75 | is sip btw |
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| 14:29.29 | [TK]D-Fender | That time is not a really valid metric. that it's up at all or constant is the key |
| 14:29.48 | ghost75 | not for me |
| 14:32.38 | ghost75 | i get like 70ms average from isp |
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| 14:35.15 | bombev | any idea Directed Call Pickup ** doesnt work when I try to pick up call with **<ext) it give me busy signal |
| 14:35.53 | file | latv, the No protocols out of 'sip' supported has been fixed in SVN and I have seen no reports of silence |
| 14:38.51 | latv | file, thanks, perhaps silence appears after patching asterisk with sipml5 patch. I personally didn't do it, only got No protocols out of 'sip' supported error with release tar.gz |
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| 14:39.12 | file | I have never used the sipml5 patch for development, and it works fine with the latest Chrome Canary without it |
| 14:39.22 | latv | fine |
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| 14:48.10 | gimp__ | hi all! |
| 14:48.48 | [TK]D-Fender | ghost75, Depends what you were loking at. |
| 14:49.02 | [TK]D-Fender | SIP QUALIFY (sip show peers) is rarely meaningful. |
| 14:49.10 | [TK]D-Fender | ACTUAL ping may be more indicative. |
| 14:49.18 | [TK]D-Fender | Then again jitter/PL is far worse |
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| 14:49.51 | [TK]D-Fender | bombev, Go look if your phone is even talking to your server for that request. "sip set debug on" <--- |
| 14:50.16 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 14:50.44 | gimp__ | i have a curious question regarding the Bridge application. i have successfully bridged 2 channels from a script which is executed from the dialplan. when user A does the bridge, the user B extension exits " exited non-zero on 'Bridge/SIP/ml-win-00000007<ZOMBIE>'".. and when either one press * to end the bridge, user B gets hangup while user A is still in the dialplan script. how can i keep the users online after the Bridge has finished? |
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| 14:57.23 | *** join/#asterisk Nikez (~Nikez@h87-96-186-134.dynamic.se.alltele.net) |
| 14:57.28 | Nikez | Hey guys |
| 14:57.32 | Nikez | Im kinda stuck here. |
| 14:57.42 | Nikez | It probably is a really noobish question.. |
| 14:58.20 | Nikez | How do I enable calling from any number to any number? :P Let's say I have username 1000, friend has 1001. |
| 14:58.30 | Nikez | I want to be able to call him |
| 14:58.37 | kannan | an external system script called with System in daemon mode works well. the script sleeps and then inteurrpts a conference (meetme); in order to stream a file thru AMI , is there a way to stream it into a meetme room ? or on the channels? is it enough to stream a file to one channel, will all the participants hear tyhe audio? |
| 14:58.53 | Nikez | or anyone for that matter, so it could be 1923 or whatever random number. |
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| 15:00.03 | [TK]D-Fender | mikthen make an extension that matches what they dail. |
| 15:00.05 | [TK]D-Fender | dial* |
| 15:00.15 | [TK]D-Fender | Nikez, Chapter 6 : Dialplan Basics |
| 15:00.16 | [TK]D-Fender | ~book |
| 15:00.16 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:00.18 | [TK]D-Fender | ^^ |
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| 15:04.58 | gimp__ | Nikez, you use the dialplan wildcard digits to specify this, sich as . _ X Z N. its called extension patterns |
| 15:08.52 | Nikez | Thanks guys |
| 15:08.55 | Nikez | I finally made it |
| 15:09.04 | gimp__ | nice |
| 15:10.31 | Brokedcomputer | Good Morning guys, |
| 15:11.49 | Brokedcomputer | So, I have the sip and sep files for my cisco 7960 and 7940, I was wondering if I could get some help making them work with the Asterisk server, I have an ftp server setup, and I am sure that I have the files in the right folder *I put them in both the ftp's root and sub folders* and now I am not sure how to configure them. |
| 15:14.04 | gimp__ | Brokedcomputer, you probably need to define the sip phones in some profile in sip.conf, either by their static ip or by username+password and put them in a dialplan context |
| 15:14.30 | Brokedcomputer | Alright, that makes sense, |
| 15:14.38 | Qwell | It would help if the phones themselves were configured first. |
| 15:14.47 | Brokedcomputer | Hehe, |
| 15:14.55 | Brokedcomputer | that is also very sensible, :P |
| 15:14.58 | gimp__ | ^^ |
| 15:15.23 | Brokedcomputer | I believe that is where I am at, needing to configure the phones, though the username and password in Sip.conf is also needed. |
| 15:15.38 | Brokedcomputer | Not sure exactly how to set that up, but I feel I may know where it goes. |
| 15:15.43 | gimp__ | you should check ouit the documentation for the cisco phones |
| 15:16.05 | gimp__ | or google found this for you http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx |
| 15:16.13 | Brokedcomputer | I have been trying to read up on that, though, it has been very confusing... |
| 15:16.36 | Brokedcomputer | Already had this open on my browser, http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx |
| 15:16.51 | Brokedcomputer | Google didn't find it for me, I knew about it, |
| 15:17.21 | gimp__ | i googled "cisco 7960 asterisk" there are several discussions on the topic |
| 15:17.42 | Brokedcomputer | Yes, but I am not sure how to change concept into action on my setup. |
| 15:19.37 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
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| 15:24.10 | Brokedcomputer | Well let me ask this, where do I set the phones to be extensions for example 100 and 101? In extensions.conf or somewhere else? Otherwise I think I have the conf files ready... atleast I think so... |
| 15:24.43 | gimp__ | user identities goes in sip.conf, dialplan extensions goes in extensions.conf |
| 15:25.00 | Brokedcomputer | Alright, I think I have that setup then |
| 15:25.32 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v025-135.mobile.uci.edu) |
| 15:25.33 | Brokedcomputer | So now I am trying to get the phone to pull its setup from the ftp, not sure that I have that setup right, the phone doesn't seem to want to pull? |
| 15:25.40 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:25.49 | [TK]D-Fender | Did you test with your own FTP client? |
| 15:25.57 | Brokedcomputer | Good idea |
| 15:26.12 | [TK]D-Fender | You sure the phone is configured for FTP? By default is seems to be TFTP which is completely different |
| 15:26.34 | Brokedcomputer | Hmmm |
| 15:27.09 | Brokedcomputer | Well, it did have ftp but another page to do with asterisk and Cisco told me to setup vsftpd |
| 15:27.20 | Brokedcomputer | tftp I mean on the first part. |
| 15:29.30 | [TK]D-Fender | As long as the phone is configured to match... |
| 15:29.52 | [TK]D-Fender | Go prove that it is indeed using FTP, and then prove that your server is in fact serving up the files |
| 15:29.57 | Brokedcomputer | hmm not sure if I need to add a port or something then, but ... I would rather do it the way the phone wants. |
| 15:30.45 | Brokedcomputer | It says TFPT server address in the settings |
| 15:31.26 | [TK]D-Fender | And do you see an option to change that? |
| 15:31.34 | Brokedcomputer | Only the address |
| 15:32.56 | Brokedcomputer | Ok, I think the TFTP server is active now, the phone is looking for a file. |
| 15:34.11 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 15:34.19 | kannan | can we send an audio file into a meetme room, where the participants can all hear? |
| 15:34.45 | kannan | can an AMI script do this? |
| 15:37.08 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 15:37.36 | WIMPy | kannan: Not directely, but you might be able to originate a call in to the conference that does a playback. |
| 15:37.57 | Brokedcomputer | Why do I get a -bash: sip: command not found ? Did I configure something wrong or miss a step? |
| 15:37.59 | kannan | WIMPy , like a LOCAL channel with originate? |
| 15:38.24 | WIMPy | Exactely |
| 15:38.27 | gimp__ | kannan: maybe have a look at ConfBridge, it allows to specify sounds for events such as user joined,left,kicked,muted and so on: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 |
| 15:38.43 | *** join/#asterisk fireman_biff (~biff@65.48.133.101) |
| 15:38.52 | gimp__ | Brokedcomputer, because you should give astersik the commands, not the command prompt |
| 15:38.53 | kannan | gimp__ ty, i will see it |
| 15:38.58 | gimp__ | start asterisk with "sudo asterisk -r" |
| 15:39.04 | Brokedcomputer | OH hehe |
| 15:39.43 | fireman_biff | How can I troubleshoot a non-PRI T1? With a PRI I would normally use "pri set debug on"... is there an equivalent for a non-PRI T1? |
| 15:39.47 | Brokedcomputer | And that tells me no such command |
| 15:40.10 | WIMPy | While speaking of ConfBridge... I noticed that the enter and leave sounds are played after the corresponding announcement. That doesn;t seem to make sense. Does anyone know if that is intended |
| 15:40.17 | WIMPy | behaviour or a bug? |
| 15:40.58 | *** join/#asterisk wotanskrieger (c8c9a427@gateway/web/freenode/ip.200.201.164.39) |
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| 15:42.59 | *** join/#asterisk evilman_home (kvirc@2.92.147.165) |
| 15:46.24 | fireman_biff | Or is there a way to get debug info from dahdi? |
| 15:49.00 | ghost75 | (15:49:16) [TK]D-Fender: SIP QUALIFY (sip show peers) is rarely meaningful. <- for me kind of is |
| 15:49.19 | ghost75 | but i could also deploy a ping probe |
| 15:49.58 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 15:49.58 | [TK]D-Fender | ghost75, that is Layer 7 and is often prioritized remarkably low. Polycom's are over 70ms on LAN <--- |
| 15:50.19 | [TK]D-Fender | ghost75, Hence whatever you get .... could mean precisely nothing as far as actual voice is concerned. |
| 15:52.08 | ghost75 | great stuff, they block icmp response |
| 15:54.25 | Katty | runs through |
| 15:54.33 | Katty | AHHH!!!! THE KNITS ARE COMING!! AHHH |
| 15:54.38 | Katty | runs back out |
| 15:54.53 | Brokedcomputer | I don't seem to have the default configs of asterisk, or any files for asterisk it seems? |
| 15:55.21 | Brokedcomputer | I installed AsteriskNOW without GUI, does that not have the example confs? |
| 15:55.55 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:55.56 | ghost75 | maybe freepbx is easier for you |
| 15:56.31 | Brokedcomputer | It probably would be but I don't have any basis, since I am ok with Ubuntu I tried that, but there was no one that wanted to help me. |
| 15:56.46 | latv | file, can you suggest a method to setup webrtc server and client inside natted network? i have a trouble that rtp going outside, most probably because of stun discovery |
| 15:56.47 | Katty | dances with [TK]D-Fender |
| 15:56.51 | [TK]D-Fender | Brokedcomputer, Most of the defaults are just conceptual samples anyway and not suggested for any kind of use. |
| 15:56.54 | *** join/#asterisk wonderworld (~w@dsdf-4db552d9.pool.mediaWays.net) |
| 15:56.54 | Brokedcomputer | I am now trying to do this with CLI, but its difficult to get help/ |
| 15:57.10 | [TK]D-Fender | Brokedcomputer, Start your own from scratch. Yuo need VERY little to be up and placing calls. |
| 15:57.14 | Brokedcomputer | Fair enough, thought I can't at all find a asterisk.conf |
| 15:57.30 | *** join/#asterisk YoMomma (~YoMomma@cpe-142-129-35-163.socal.res.rr.com) |
| 15:57.34 | ghost75 | i only know german tutorials |
| 15:57.34 | [TK]D-Fender | /etc/asterisk <--- |
| 15:57.35 | file | latv, the ICE candidates should have host as prioritized over outside if possible - nothing you can do to change that... it's part of the ICE negotiation |
| 15:57.37 | [TK]D-Fender | Nothing there? |
| 15:57.39 | Brokedcomputer | Thats all I am trying to do is setup ONE phone to make a test call to voice mail. |
| 15:57.48 | Brokedcomputer | Nothing in there about asterisk.conf. |
| 15:57.52 | gimp__ | Brokedcomputer, if you use ubuntu, install asterisk-config |
| 15:58.02 | *** join/#asterisk felimwhiteley_ (~quassel@089-101-203026.ntlworld.ie) |
| 15:58.04 | [TK]D-Fender | PASTEBIN "ls -la /etc/asterisk" |
| 15:58.15 | Brokedcomputer | I am using Centos as that is what came with AsteriskNOW. |
| 15:58.17 | [TK]D-Fender | ~pb |
| 15:58.17 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:58.25 | *** join/#asterisk Galen (~Galen@rrcs-24-43-17-235.west.biz.rr.com) |
| 15:58.57 | wonderworld | i am having a strange problem. asterisk suddenly stopped to accept REGISTERs. port 5060 seems to be blocked. even if i try to access it from localhost (nc 127.0.0.1 5060) i am unable to connect. iptables is not running. i didn't change the servers config and i don't know where to start fixing it... |
| 15:59.28 | gimp__ | wonderworld, is asterisk running? |
| 15:59.32 | wonderworld | yes it is |
| 15:59.38 | wonderworld | i restarted it several times as well |
| 15:59.47 | gimp__ | do /var/log/asterisk/messages have a clue? |
| 15:59.48 | Brokedcomputer | http://pastebin.com/WHnTGMWE |
| 15:59.54 | wonderworld | wait, let me check |
| 16:00.10 | Katty | [TK]D-Fender: why u no dance wif me?!!?! |
| 16:00.21 | Katty | [TK]D-Fender: no snobs allowed! |
| 16:00.27 | ghost75 | nc 127.0.0.1 5060 <- that will show refused anyway |
| 16:01.15 | wonderworld | ohh, ok. so this might not be the best way to test it |
| 16:01.33 | ghost75 | maybe its a problem on the router |
| 16:01.40 | [TK]D-Fender | Brokedcomputer, I suggest you install the sample configs for now substituting the ones you are alrady doing on your own, but leaving the rest for now. |
| 16:01.41 | ghost75 | i had that on openwrt |
| 16:01.56 | wonderworld | messages shows nothing |
| 16:01.56 | [TK]D-Fender | Brokedcomputer, they are all in the "config" folder in the * tarball. |
| 16:02.09 | wonderworld | cli with high verbosity shows nothing |
| 16:02.14 | ghost75 | reboot router wonderworld |
| 16:02.15 | gimp__ | wonderworld, if you run asterisk on localhost, it will occupy port 5060. the same port is used by the sip client so you cant run both on the same machine & ip |
| 16:02.24 | Brokedcomputer | I will check that out thanks [TK]D-Fender |
| 16:02.27 | [TK]D-Fender | Katty, Sure I'm be right over... I'll start walknig now. See you in 2013! |
| 16:02.45 | wonderworld | ghost75: it's a hosted server. no router i could access |
| 16:02.47 | gimp__ | wonderworld, also if you behind a NAT you will need a stun server or something. try to get it working on the same LAN first |
| 16:02.54 | wonderworld | how could i test the connection from local host |
| 16:03.09 | wonderworld | it used to work fine for years...it just suddenly started |
| 16:04.02 | Katty | [TK]D-Fender: maybe you should take a plane. |
| 16:04.20 | [TK]D-Fender | Katty, Actually ... I have no passport... good thing I didn't make it to the border yet! |
| 16:04.30 | tonikasch | Do you know if cdr_csv.so is included by default in Asteisk 1.8.10? Or you have to install some other packages? |
| 16:04.55 | [TK]D-Fender | tonikasch, yes |
| 16:05.22 | ghost75 | i found also its hard to troubleshoot sip connections if everything is blocked |
| 16:05.37 | tonikasch | [TK]D-Fender, Ok, so it's a problem with openwrt, they don't include this file in the main package |
| 16:05.48 | Katty | [TK]D-Fender: that's ok. you shouldn't enter this country anyway |
| 16:05.51 | wonderworld | hmm. i guess i will reboot the server |
| 16:06.04 | wonderworld | "windows method" ;) |
| 16:06.22 | [TK]D-Fender | Katty, I'll let you know on say ...Nov 7th ;) |
| 16:06.48 | Katty | [TK]D-Fender: <3 |
| 16:07.13 | *** join/#asterisk IceMan82 (~chatzilla@67.48.122.234) |
| 16:07.16 | [TK]D-Fender | Katty, Or however long it takes the Supreme Court to assign a victor ;) |
| 16:08.26 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-ulrdjcpiesxykpsf) |
| 16:08.43 | IceMan82 | Hello, does anyone have any experience with UniMRCP? I've been trying to integrate the bridge connector with asterisk for 2 days with no success. |
| 16:12.24 | *** join/#asterisk mobile_gordita (~Robert@66-87-111-34.pools.spcsdns.net) |
| 16:12.40 | Katty | carrar: ESPRESSO |
| 16:12.43 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:13.19 | *** join/#asterisk Gugge (gugge@kriminel.dk) |
| 16:14.07 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 16:15.45 | gimp__ | i have a curious question regarding the Bridge application. i have successfully bridged 2 channels from a script which is executed from the dialplan. when user A does the bridge, the user B extension exits " exited non-zero on 'Bridge/SIP/ml-win-00000007<ZOMBIE>'".. and when either one press * to end the bridge, user B gets hangup while user A is still in the dialplan script. how can i keep the users online after the Bridge has finished? |
| 16:19.52 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 16:27.22 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 16:33.36 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:36.38 | *** join/#asterisk bchia (~Adium@nat/digium/x-zhmaccmpauqehhsq) |
| 16:58.51 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 17:02.54 | *** join/#asterisk k610 (~Instantbi@host-78-129-3-116.brutele.be) |
| 17:05.57 | Brokedcomputer | [TK]D-Fender - So I have the sample scripts in that folder now, and I have duplicated Asterisk.conf and left it as is, could you help me set it up? |
| 17:06.34 | [TK]D-Fender | You shouldn't have to mess with asterisk.conf |
| 17:06.52 | [TK]D-Fender | first see if your phone has registered to your SIP peer |
| 17:06.58 | *** join/#asterisk thehar (~thehar@diddlebox.thehar.com) |
| 17:07.10 | Brokedcomputer | I would need to restart my server to make the asterisk.conf enabled right? |
| 17:07.24 | [TK]D-Fender | just do "service asterisk restart" |
| 17:07.28 | [TK]D-Fender | from OS CLI |
| 17:07.29 | Brokedcomputer | *This is a brand new file, asterisk in my setup didn't have it. |
| 17:07.31 | Brokedcomputer | oh ok |
| 17:07.32 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 17:07.52 | [TK]D-Fender | And you'd better make sure that when you put configs there they went in with the right permissions... |
| 17:08.00 | [TK]D-Fender | should be asterisk:asterisk for them |
| 17:08.04 | Brokedcomputer | yup |
| 17:08.14 | Brokedcomputer | Ok its restarted. |
| 17:08.23 | Brokedcomputer | And the user and group is asterisk |
| 17:08.46 | [TK]D-Fender | ok, restart and do "sip show peers" to make sure * has even picked up your entires. then "sip show peer THEPEER" for each and pastebin it |
| 17:09.14 | *** join/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
| 17:09.42 | Brokedcomputer | I don't think it has, I get that there is no such command. |
| 17:10.44 | Brokedcomputer | Is there a preferred way to install asterisk that I can start from again? |
| 17:10.53 | [TK]D-Fender | show us |
| 17:10.56 | Brokedcomputer | Since I still know very little, |
| 17:11.41 | Brokedcomputer | http://pastebin.com/SW26ABy0 |
| 17:13.36 | [TK]D-Fender | I don't see all the other sample files in your folder as per our previous discussions... |
| 17:13.50 | Brokedcomputer | Sorry, I didn't update that list, |
| 17:13.53 | [TK]D-Fender | And specifically not asterisk.conf which even tells * where your MODULES are. |
| 17:14.14 | [TK]D-Fender | trash all that old stuff and do not repaste. All new. from here on. |
| 17:14.44 | ghost75 | centos has asterisk 1.8 already in? |
| 17:15.02 | [TK]D-Fender | No. |
| 17:15.22 | Brokedcomputer | Here is the latest list http://pastebin.com/L7mAiQ7k |
| 17:16.01 | [TK]D-Fender | -rwxrwxr-x 1 asterisk asterisk 1286 Nov 13 2009 modules.conf.sample |
| 17:16.09 | [TK]D-Fender | need this to be non-sample as well... |
| 17:16.27 | [TK]D-Fender | just rename them all to normal except for yuor own custom ones. |
| 17:16.41 | [TK]D-Fender | this will save the file-by-file failure this path is leading on... |
| 17:16.42 | Brokedcomputer | Oh ok |
| 17:16.47 | Brokedcomputer | ahh |
| 17:17.10 | jeffspeff | having an issue with paging now... using asterisk ver 11.0 http://pastebin.com/FVmxhEjL it seems that it wants to use app_confbridge to mix the audio instead of app_meetme like i think it used to. |
| 17:17.36 | kaldemar | jeffspeff: that's a change made in 11. |
| 17:17.39 | *** join/#asterisk sertaconay (~sertacona@unaffiliated/sertaconay) |
| 17:18.13 | jeffspeff | kaldemar, is there any documentation for proper configuration of app_page for 11? |
| 17:19.18 | file | you can't configure app_page - there is nothing to configure, do you have a confbridge.conf? |
| 17:19.37 | jeffspeff | file, yes, that's what i'm using for conferencing |
| 17:19.47 | file | weird. |
| 17:20.10 | file | did you remove the defaults? |
| 17:20.18 | jeffspeff | file, yes |
| 17:20.24 | file | ah |
| 17:20.33 | file | that would be why |
| 17:21.09 | jeffspeff | so, is it looking to use the default user or the default bridge? |
| 17:21.15 | jeffspeff | or both? |
| 17:21.25 | file | default_bridge |
| 17:27.32 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 17:28.56 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
| 17:29.57 | Brokedcomputer | Alright there [TK]D-Fender |
| 17:30.08 | Brokedcomputer | Done that, and I restarted the server. |
| 17:30.30 | Brokedcomputer | YAY, sip show peers worked :P |
| 17:30.48 | Brokedcomputer | clear |
| 17:30.51 | Brokedcomputer | oops. :P |
| 17:31.34 | kannan | originate a local channel with AMI , to stream an audio file into a meetme room is working well. however, re-direct is causing the channels to hangup. i want to re-direct 1 participant to a IVR , out of the meetme, and hangup the other |
| 17:32.36 | gimp__ | kannan, http://www.voip-info.org/wiki/view/Asterisk+cmd+ChannelRedirect ? |
| 17:32.43 | Brokedcomputer | http://pastebin.com/6a1NmxXu <- this is the most recent test [TK]D-Fender |
| 17:33.11 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:33.36 | Brokedcomputer | I really gotta remember to remove the old stuff, clear doesnt get rid of it.. |
| 17:34.02 | Brokedcomputer | What you wanted to see is at the very bottom |
| 17:35.16 | [TK]D-Fender | Brokedcomputer, (Unspecified) <- hasn't registered yet. |
| 17:35.25 | Brokedcomputer | Yeah I saw that, |
| 17:35.40 | Brokedcomputer | How do I fix that? |
| 17:35.50 | [TK]D-Fender | Brokedcomputer, Now we don't know if the phones are even configured right on their end. so "sip set debug on" and restart a phone and see if a registration attempt comes in |
| 17:36.40 | Brokedcomputer | I almost garentee that its not configured right on the phone's side... |
| 17:37.36 | Qwell | <Qwell> It would help if the phones themselves were configured first. |
| 17:37.40 | kannan | gimp__ , thanks |
| 17:38.07 | kannan | in re-direct thru AMI, we cannot use Asterisk variables ? even if global vars? |
| 17:38.15 | Brokedcomputer | It doesnt seem to want to pull my configuration files from the tftp server, but I am not sure what I am missing, I have the sep and sip files in there, and the firmware. |
| 17:38.32 | Brokedcomputer | and I have named them with the mac addresses |
| 17:38.37 | [TK]D-Fender | Brokedcomputer, Last you mentioned was having the files in an FTP specific folder. |
| 17:39.02 | [TK]D-Fender | Brokedcomputer, You should probably show ALL of that new setup and an attempt with a TFTP client to grab them to make sure it's even sane |
| 17:39.06 | Brokedcomputer | Ah yes, I reenabled the tftp program and the phone see's the server, but can't find its files. |
| 17:39.08 | jeffspeff | file, adding the default_bridge made the conference work, but it shows the following errors in output http://pastebin.com/gugsMSFL |
| 17:39.12 | [TK]D-Fender | Brokedcomputer, Prove the lowest levels first |
| 17:39.20 | Brokedcomputer | Hmmmm |
| 17:39.28 | Brokedcomputer | Ok, so I need a tftp client? |
| 17:39.38 | [TK]D-Fender | Clearly |
| 17:39.38 | file | that I don't know why it would immediately be doing that |
| 17:41.18 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 17:44.21 | Brokedcomputer | Hey [TK]D-Fender I got it, and it stated, 7blocks transferred in 1 second. |
| 17:44.32 | Brokedcomputer | I am using tftp32 |
| 17:44.48 | [TK]D-Fender | So you grabbed teh files OK? |
| 17:44.52 | Brokedcomputer | Yeah |
| 17:45.44 | [TK]D-Fender | Brokedcomputer, Ok, now we're down to misconfiguration of the files themselves or that the phone isn't pointing to it properly |
| 17:45.51 | Brokedcomputer | Fair enough |
| 17:46.17 | [TK]D-Fender | Brokedcomputer, Ciscos have 2 different config file styles depending on what version of firmware you'er on. I don't have direcet experience with that part however. |
| 17:46.21 | Brokedcomputer | when the phone loads up it says configuring VLAN, then goes to my router, then tries the tftp server |
| 17:46.42 | Brokedcomputer | By IP address is what I mean when I say tries. |
| 17:47.07 | fireman_biff | Is there a "pri set debug on" equivalent for a non-PRI T1? |
| 17:47.14 | Brokedcomputer | It loads the 803 firmware, |
| 17:47.46 | Brokedcomputer | So I agree, I think its the settings in one of my files. |
| 17:49.23 | [TK]D-Fender | Sounds like. |
| 17:49.53 | [TK]D-Fender | Brokedcomputer, No traffic at all means that it isn't like a bad user/pass .. it's that in it's current state it isn't trying to contact the server at all (SIP) |
| 17:50.08 | Brokedcomputer | That sounds right, |
| 17:50.23 | Brokedcomputer | So do you have any experience to look at my sip settings in the files? |
| 17:50.50 | [TK]D-Fender | As I said, no direct experience with Cisco configs.... |
| 17:51.02 | Brokedcomputer | :P |
| 17:51.11 | [TK]D-Fender | Anyone else here able to help this fellow out with his Cisco configs? |
| 17:51.20 | ghost75 | no |
| 17:51.31 | Brokedcomputer | Thanks for all your help though, got me places I wouldnt have been yet :p |
| 17:51.40 | ghost75 | i only have cisco ata |
| 17:51.41 | Brokedcomputer | Could we setup a software sip phone for testing? |
| 17:52.01 | Brokedcomputer | Worry about the complexities of cisco later? |
| 17:52.08 | Qwell | ~book |
| 17:52.08 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:52.10 | Qwell | You could, sure. |
| 17:52.13 | Brokedcomputer | I mean since we know the config files are loading :P |
| 17:52.19 | *** join/#asterisk karl370 (~karl370@12.139.131.2) |
| 17:52.25 | ghost75 | are cisco phones really different? |
| 17:52.30 | Brokedcomputer | Not sure,. |
| 17:52.59 | ghost75 | you can even use some mobiles as sip client |
| 17:53.11 | [TK]D-Fender | Brokedcomputer, Of course you coudl test the * side with jsut a softphone. Go for it |
| 17:53.15 | Brokedcomputer | Id rather on my computer :P |
| 17:53.28 | [TK]D-Fender | ~x-lite |
| 17:53.30 | Brokedcomputer | awesome, ... this is going to be interesting... |
| 17:53.33 | [TK]D-Fender | ~xlite |
| 17:53.34 | infobot | [~xlite] X-Lite is a free SIP soft-phone for Windows, Mac OS X, and Linux that can be downloaded from http://www.counterpath.com |
| 17:53.43 | [TK]D-Fender | ~zoiper |
| 17:53.43 | infobot | [~zoiper] Zoiper (formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, Mac OS X, and Linux that can be found at http://www.zoiper.com |
| 17:53.50 | drmessano | Zoiper+ |
| 17:53.51 | [TK]D-Fender | ^ probably a better choice |
| 17:54.00 | ghost75 | both free? |
| 17:54.14 | drmessano | The OLD X-Lite was pretty awesome for testing. The new one sucks. Zoiper is great |
| 17:54.17 | drmessano | Free |
| 17:54.40 | drmessano | Actually, the OLD, OLD, OLD X-Lite. I think that was 3 OLDs ago |
| 17:55.50 | jmetro | the 3cx softphone is pretty nice from what i've used. |
| 17:55.57 | Brokedcomputer | My wife was distracting me, hadn't downloaded the x-lite yet, I will get zoiper |
| 17:56.08 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 17:56.26 | ghost75 | where are getting those softphones the phonebook from |
| 17:56.34 | jmetro | ~3cx |
| 17:56.34 | [TK]D-Fender | themselves |
| 17:56.35 | ghost75 | ldap? |
| 17:56.43 | jmetro | aw theres no ~3cx macro |
| 17:58.00 | *** join/#asterisk doulos1 (~bcalhoun@71-14-6-250.static.gwnt.ga.charter.com) |
| 17:58.21 | *** join/#asterisk mbrit (~mbrit@186.120.97.194) |
| 17:59.20 | Brokedcomputer | WOW just got a HUGE bunch of text from the sip debug. :P |
| 18:01.15 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 18:01.33 | *** join/#asterisk volga629 (~volga629@host7.pythian.com) |
| 18:02.16 | volga629 | Hello Everyone, res_rtp_asterisk.c: RTCP SR transmission error to ip:20403, rtcp halted Operation not permitted, what is mean ? |
| 18:02.55 | Brokedcomputer | Hmmm how do I setup an extension for the software sip phone under extensions? |
| 18:05.33 | Brokedcomputer | I ask because I can see how to setup a hardware phone, but not a softphone? |
| 18:07.07 | ghost75 | is the same i think |
| 18:07.35 | Brokedcomputer | I am just not sure how the mac address plays in though, is that my ethernet mac or some software number? |
| 18:07.50 | ghost75 | you dont need |
| 18:08.08 | Brokedcomputer | in sip.conf I have this |
| 18:08.09 | Brokedcomputer | [](school-phone) |
| 18:08.09 | Brokedcomputer | username=craig |
| 18:08.21 | Brokedcomputer | what goes in the [] brackets then? |
| 18:08.28 | ghost75 | the extension |
| 18:08.42 | Brokedcomputer | So 101? |
| 18:09.03 | ghost75 | if 101 is your ext |
| 18:09.40 | Brokedcomputer | Hmmm |
| 18:09.47 | ghost75 | http://agix.com.au/blog/?p=2656 |
| 18:10.01 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:11.51 | Brokedcomputer | So under extensions, do I need the ()'s? exten => 102,1,Dial(SIP/0019AA735FBA) ; Phone 3 - Software based Uber-Machine |
| 18:14.11 | [TK]D-Fender | Brokedcomputer, yes |
| 18:14.25 | Brokedcomputer | So what goes in there for a software sip phone? |
| 18:15.03 | [TK]D-Fender | Brokedcomputer, you also don't have to name yuor peers like MAC addresses. Some sample guides may show it like that because it is *a* concept that might be helpful for your lookups when configuring other things, but you could jsut as easily have named it [fred] |
| 18:15.23 | Brokedcomputer | OH |
| 18:15.42 | jpsharp | The same. Asterisk doesn't care if it is talking to a softphone, an IP phone, or a cow at the end of a tin can & strings. |
| 18:15.45 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-oghlaqvhitbfgvzs) |
| 18:15.57 | Brokedcomputer | hahahaha |
| 18:17.02 | Brokedcomputer | AHHHHHHH this makes more sense now. :) |
| 18:19.35 | Brokedcomputer | So this is what you wanted to see all along in the one test, http://pastebin.com/LWdMqDiA |
| 18:21.53 | Brokedcomputer | This is what I am getting from the software phone, http://pastebin.com/fdAGe6mq |
| 18:23.50 | chris_n | wonders how one assigns an ip to a cow at the end of a tin can & string |
| 18:24.10 | Brokedcomputer | :P |
| 18:24.11 | chris_n | and it is static or dhcp? |
| 18:24.51 | [TK]D-Fender | US-AS-AD-01*CLI> sip show peer 102 |
| 18:24.54 | ghost75 | makes asterisk talk to cow |
| 18:25.14 | [TK]D-Fender | To: "102"<sip:craig@192.168.1.25;transport=UDP> |
| 18:25.19 | jpsharp | chris_n: ipv6 |
| 18:25.24 | [TK]D-Fender | Your phone thinks its username is "craig" |
| 18:25.35 | Brokedcomputer | Yeah |
| 18:25.47 | [TK]D-Fender | but your sip.conf peer username well .. doesn't seem to be right... |
| 18:25.47 | Brokedcomputer | SO it should be 102? |
| 18:25.59 | Brokedcomputer | hmmm |
| 18:26.23 | Brokedcomputer | This is the setting currently |
| 18:26.25 | Brokedcomputer | ; define another device name using the same template |
| 18:26.25 | Brokedcomputer | [102](school-phone) |
| 18:26.25 | Brokedcomputer | username=craig |
| 18:26.32 | [TK]D-Fender | For starters I'd remove defaultuser, and make it just [102] across the board |
| 18:26.39 | [TK]D-Fender | And ditch templates |
| 18:26.47 | [TK]D-Fender | like "(school-phone) |
| 18:26.53 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 18:27.05 | [TK]D-Fender | just do the parms direct in your peer. Better to learn it explicit to start |
| 18:27.05 | Brokedcomputer | I don't belive I have a default user set? |
| 18:27.16 | [TK]D-Fender | username/defaultuser in the sip.conf entry |
| 18:27.17 | [TK]D-Fender | ^ |
| 18:27.22 | [TK]D-Fender | Def. Username: craig |
| 18:28.19 | Brokedcomputer | So does default user override the 102 setting? |
| 18:31.26 | Brokedcomputer | Ahhh so I changed the user name to 102 and it registered, |
| 18:31.38 | Brokedcomputer | thouhg, I still cannot make calls :P hahahhaa |
| 18:34.15 | [TK]D-Fender | baby steps, and you're working at it. this is very good. |
| 18:34.22 | [TK]D-Fender | now to look at a failed call. |
| 18:34.32 | ghost75 | anyone knows how i could sort an php multiarray by a-z ? |
| 18:34.58 | [TK]D-Fender | Again, * CLI + SIP DEBUG will tell you what entry it matches and show what it's trying to dial and in what context |
| 18:39.52 | Brokedcomputer | Ok, I have two softphones setup to line 102, and 103, |
| 18:40.27 | Brokedcomputer | Now, I know that I have not done everything yet, is it possible to make a call between them, or do I need something else? |
| 18:41.19 | Brokedcomputer | Ahhh sorry, I missed the look at a failed call part. |
| 18:41.28 | Brokedcomputer | I will post the cli |
| 18:42.25 | Brokedcomputer | http://pastebin.com/qkDgAfXy |
| 18:44.43 | [TK]D-Fender | Brokedcomputer, There is no call attempt in there |
| 18:44.52 | Brokedcomputer | hmm ok one moment |
| 18:45.53 | Brokedcomputer | Should be the last set in the paste, I am not sure where they start and end... http://pastebin.com/79TVkQ8a |
| 18:46.42 | Brokedcomputer | From: "Craig"<sip:102@192.168.1.25;transport=UDP>;tag=0b6d3e45 |
| 18:46.42 | Brokedcomputer | To: <sip:103@192.168.1.25;transport=UDP> |
| 18:46.53 | [TK]D-Fender | Brokedcomputer, Looking for 103 in LocalSets (domain 192.168.1.25) |
| 18:47.04 | Brokedcomputer | yup, |
| 18:47.12 | [TK]D-Fender | [Nov 1 18:44:57] WARNING[18646]: chan_sip.c:5511 create_addr: No such host: softphone |
| 18:47.21 | [TK]D-Fender | "core set verbose 10" |
| 18:47.23 | [TK]D-Fender | do it again. |
| 18:47.37 | [TK]D-Fender | You are missing basic verbose information which will show you dialplan execution there |
| 18:47.57 | [TK]D-Fender | looks like you're doing dial(SIP/softphone) in there |
| 18:48.17 | [TK]D-Fender | And that that peer HASN'T registered yet so * has no idea how to contact them. |
| 18:48.29 | Brokedcomputer | http://pastebin.com/Qi83jDgG |
| 18:48.37 | [TK]D-Fender | Or... actually sorry, that it doesn't even exist as a sip.conf entry |
| 18:48.46 | Brokedcomputer | Oh so I have the wrong name somewhere |
| 18:51.34 | Brokedcomputer | http://pastebin.com/9Fug80Ui Fixed that, no audio though? |
| 18:52.05 | [TK]D-Fender | Now with SIP debug please... |
| 18:53.36 | Brokedcomputer | http://pastebin.com/21HF667S More info than I could copy?! |
| 18:54.52 | [TK]D-Fender | you copied your buffer BACk to CLI |
| 18:55.03 | Brokedcomputer | ... sorry |
| 18:55.53 | Brokedcomputer | What was the command for setting debug to 10? |
| 18:56.09 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 19:02.54 | Brokedcomputer | http://pastebin.com/LFaSg2yi <- found the setting, and did a sip show peer 102 just before I initiated the call. What do you think? |
| 19:03.14 | [TK]D-Fender | "sip set debug on" |
| 19:03.31 | [TK]D-Fender | <PROTECTED> |
| 19:03.37 | [TK]D-Fender | though this seems to say it all... |
| 19:03.55 | Brokedcomputer | ? |
| 19:04.07 | [TK]D-Fender | you need several settings for SIP to work where NAT is involved |
| 19:04.13 | Brokedcomputer | What is that IP? |
| 19:04.25 | Brokedcomputer | Thats my external? Why is that in there? |
| 19:04.44 | [TK]D-Fender | You are testing with a remote device, no? |
| 19:04.50 | Brokedcomputer | no |
| 19:04.58 | Brokedcomputer | its local, along with all the systems |
| 19:05.55 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 19:08.14 | [TK]D-Fender | new call with SIP debug please |
| 19:08.23 | Brokedcomputer | ok |
| 19:10.26 | Brokedcomputer | Updated http://pastebin.com/LFaSg2yi |
| 19:11.23 | Brokedcomputer | Is there some setting in Zoiper that chooses the external IP? |
| 19:11.43 | ghost75 | veronika is your wife? |
| 19:11.48 | Brokedcomputer | Yes |
| 19:13.23 | [TK]D-Fender | that is not the complete call |
| 19:13.33 | Brokedcomputer | http://pastebin.com/XJWWjJzG |
| 19:14.25 | [TK]D-Fender | STILL not a complete call... or even part of one there |
| 19:17.41 | Brokedcomputer | Call started, somewhere in there, http://pastebin.com/seEh6VJd |
| 19:18.23 | Brokedcomputer | http://pastebin.com/FMsrycrT Call completed, though I am not sure where they meet up, I also have a middle portion while the call was in place if you need it. |
| 19:19.53 | Brokedcomputer | Not sure why it is using my external to register the phone. this is annoying. |
| 19:20.47 | [TK]D-Fender | set_destination: set destination to 184.71.199.202:51415 |
| 19:21.02 | [TK]D-Fender | Seems that Zoiper is loking at your external interface a bit much... |
| 19:21.07 | Brokedcomputer | yeah... |
| 19:21.10 | [TK]D-Fender | make sure your peers have "directmedia=no" |
| 19:21.30 | Brokedcomputer | is that in sip.cong? |
| 19:21.32 | Brokedcomputer | conf* |
| 19:21.37 | [TK]D-Fender | and "nat=yes" (just to lock them down even though they aren't |
| 19:21.43 | [TK]D-Fender | Yes, this is kinda dirty... |
| 19:21.48 | [TK]D-Fender | and yes, sip.conf |
| 19:22.37 | Brokedcomputer | Should I copy that to IAX as well? |
| 19:25.12 | Brokedcomputer | http://pastebin.com/tvdertV6 |
| 19:27.17 | Brokedcomputer | This is what my sip.conf looks like http://pastebin.com/CyVk28h2 |
| 19:28.31 | Brokedcomputer | ... ... why is it looking at my external... :P hahahah this so strange. :P |
| 19:34.17 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v025-135.mobile.uci.edu) |
| 19:35.45 | ghost75 | what is directmedia doing? |
| 19:36.56 | ghost75 | that its forcing 1:1 connection without nat ? |
| 19:37.46 | file | stares at wall |
| 19:38.48 | [TK]D-Fender | broHave you tested? |
| 19:46.03 | jpsharp | ghost75: directmedia says the phones direct their RTP streams directly at each other, rather than using Asterisk as a middleman. Directmedia does not work well when NAT is involved. |
| 19:48.45 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v025-135.mobile.uci.edu) |
| 19:55.29 | ghost75 | jpsharp: is that related only to phones or also to peers ? |
| 19:56.26 | [TK]D-Fender | phones vs peers" is not a proper use of terms |
| 19:56.32 | [TK]D-Fender | SIP is SIP. |
| 19:56.56 | [TK]D-Fender | Wrong IP's are wrong IP's. |
| 20:01.12 | jpsharp | ghost75: sip endpoint to sip endpoint, whether the endpoints be a SIP phone on someone's desk, an ITSP, or an ipv6 based cow running a SIP stack. |
| 20:01.35 | [TK]D-Fender | MOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOo |
| 20:01.42 | jpsharp | 200 MOOO! |
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| 20:18.46 | Brokedcomputer | [TK]D-Fender any idea's why its looking at my external instead of internal? |
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| 20:22.37 | [TK]D-Fender | I asked about your having tested with the new setup and didn't get debug to confirm |
| 20:22.46 | [TK]D-Fender | unfortunately time is up on my side. back later... |
| 20:23.19 | philfry | so i have a conference line and if i try to dial it while on a ivr it seems to try to use one of the ivr entries instead of going to the conference line |
| 20:24.51 | philfry | for instance lets say one of the entries for the ivr is 1 and the conference line is 103 it will go to the ivr entry instead of the conference line |
| 20:34.50 | ectospasm | philfry: pastebin your dialplan |
| 20:34.52 | ectospasm | ~pb |
| 20:34.52 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 20:36.31 | Brokedcomputer | ~info |
| 20:37.51 | Brokedcomputer | Well, Thanks for all the help guys, I do appreciate it, I need a break as well :P Gonna go play some games :P |
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