| 00:00.05 | hesco1 | does asterisk have any dependencies on python? |
| 00:01.35 | hesco1 | I've been getting congestion on both inbound and outbound calls all day and when I got a chance to check on it, sip show peers showed no connections to either my devices or my providers. |
| 00:02.31 | WIMPy | Maybe it's epidemic. I had the same issue today. |
| 00:02.41 | WIMPy | Well, technically yesterday. |
| 00:02.55 | WIMPy | But if you're looking for a solution we need input. |
| 00:03.08 | hesco1 | last night I was updating python for another service I host on this box and I wondered if their might be some relationship |
| 00:03.43 | WIMPy | Extremely unlikely. |
| 00:04.13 | hesco1 | I'm late for a dinner date and would certainly appreciate some help thinking through this, but that will have to wait, I'm afraid. |
| 00:04.42 | hesco1 | Is it just you and me? or has everyone been hit with this? |
| 00:05.06 | hesco1 | I'll check back in after dinner. Thanks. |
| 00:05.32 | WIMPy | For me the box was running out of FDs for some unknown reason. Very unlikely that's related to anyone else. |
| 00:21.54 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
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| 01:46.24 | hesco1 | WIMPy: so what is an FD? |
| 01:46.44 | jpsharp | file descriptor |
| 01:46.52 | hesco1 | Back from dinner and my peers still have not managed to register. |
| 01:47.01 | hesco1 | ah. |
| 01:48.47 | hesco1 | I have a rather large hard drive but am using less than 40gb of it. I doubt that is my issue, but just in case, how is it I would query for that? |
| 01:49.17 | hesco1 | and is that quota applied system wide, or by partition? |
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| 01:49.18 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:52.35 | jpsharp | Well, file descriptors aren't necessarily tied down to hard drive space. |
| 01:53.45 | jpsharp | TCP and UDP connections also use up file descriptors. |
| 01:54.59 | hesco1 | I get this: cat /proc/sys/fs/file-nr |
| 01:54.59 | hesco1 | 1216 0 294409 |
| 01:57.31 | jpsharp | When your calls fail, do you see any kind of error messages on your Asterisk console? |
| 01:58.35 | jpsharp | It is highly unlikely that FD limits are your problem. Unless you have some other scripts/code running rampant on your server or you're handling hundreds of calls per second. |
| 01:59.36 | hesco1 | I'm seeing lots of this: Registration for 'every itsp vendor trunk configured' timed out, trying again (Attempt #82) |
| 01:59.53 | hesco1 | that is not my issue then. |
| 02:01.28 | hesco1 | Afraid I have the most drastically underutilized pbx this side of the Mississippi. |
| 02:02.38 | jpsharp | That sounds more like a network problem. Can your PBX host machine ping the outside world? |
| 02:03.18 | hesco1 | I tracerouted one of my trunk proxies from the pbx and the pbx from another server. |
| 02:03.49 | hesco1 | traceroute found my server. My traceroute of the proxy failed, but long after it had left my local data center |
| 02:04.19 | jpsharp | Time to turn on sip debug and see if something is complaining. |
| 02:04.24 | hesco1 | I was thinking at first it was a network issue related to my host's firewall. |
| 02:04.37 | hesco1 | sip debug 3 <-- like that? |
| 02:05.05 | jpsharp | sip set debug on |
| 02:06.39 | *** join/#asterisk gusto (~gusto@ppp-93-104-81-177.dynamic.mnet-online.de) |
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| 02:08.11 | jpsharp | You should see your asterisk box send a register message to the far end, get a "407 Unauthorized" message in reply, then it will send another register message with credentials, and the far end should reply with "200 OK" if the registration was successfull. |
| 02:12.00 | volga629 | Hello Everyone, After all this troubleshooting discovered that all issue is on snom 370 which refuse take lldp staff from switch. If I set vlan tag id manually it reboots and all working if I let got trough lldp nothing not working that just ... |
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| 02:30.56 | hesco1 | took me a bit to sanitize the paste, but sip set debug on gave me this: http://paste.scsys.co.uk/208836 |
| 02:31.16 | hesco1 | I'm not seeing any response codes in the mix, neither 407, nor 200 |
| 02:31.46 | hesco1 | jpsharp: would you mind taking a glance at that and letting me know what you make of it, please? |
| 02:32.01 | jpsharp | Something is eating SIP packets. |
| 02:32.33 | jpsharp | Either eating the outbound packet or the inbound response. |
| 02:38.58 | hesco1 | a sip packet eating monster is quarding my network? |
| 02:39.42 | hesco1 | can I poison a sip packet in hopes of killing it? |
| 02:40.18 | jpsharp | I don't think so. |
| 02:42.32 | hesco1 | any thoughts on other strategies for resolving this issue and putting my telephony network back online? |
| 02:48.45 | jpsharp | Well, you have to figure out where things are dying. If you have root access to a box way off your network, you can tell your asterisk box to register to it and then you can log into that box to see if you see the SIP registry packets. |
| 02:49.28 | jpsharp | If you can, then you have something blocking inbound packets. If not, you have something blocking outbound packets. |
| 02:50.23 | jpsharp | Or, failing that, you can try to register against my asterisk box and I'll tell you if I see the packets. |
| 02:52.14 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 02:53.58 | jpsharp | pick a random username and try to register via sip to asterisk.fivecats.org and I'll tell you if I see the packets. |
| 02:54.53 | hesco1 | thanks, setting that up now, will alert you when I'm ready for your attention on that one |
| 03:02.11 | hesco1 | I added a sip trunk in the freepbx interface, but its taking forever to reload the configuration. |
| 03:03.41 | hesco1 | dialplan reload at the cli runs w/o a hitch, even though the fpbx interface is still trying to reload from before I hit the cli with the reload |
| 03:10.22 | jpsharp | I haven't seen any indication of your box registering here. |
| 03:10.48 | jpsharp | Saw my iphone register over 3G, so I know external internet access works. |
| 03:50.03 | *** join/#asterisk cmendes0101 (~nn@pool-98-112-76-215.lsanca.fios.verizon.net) |
| 03:51.30 | cmendes0101 | I'm having trouble setting up ChanSpy. I'm trying to monitor a call that is a sip user into an IVR. When calling ChanSpy it only picks up the sip users audio and not the IVR. |
| 03:51.46 | cmendes0101 | From what I saw ChanSpy should be picking up both, is that correct? |
| 03:58.19 | jpsharp | Depends on the option you pass ChanSpy. |
| 04:00.02 | jpsharp | You have to pass it "b" to listen to both sides, otherwise it only passes you audio from the channel you specify in the ChanSpy command. |
| 04:01.04 | cmendes0101 | Ok let me try that. I thought that ment something different and didn't see any other option that would be useful so had been passing no options |
| 04:03.41 | cmendes0101 | hmm with that enabled seems to not find the channel |
| 04:21.37 | cmendes0101 | aw ok got it with B. Thanks |
| 04:22.00 | jpsharp | Ah, okay. |
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| 06:05.48 | hesco1 | jsharp: I'd imagine not. It was only now that I've been able to get fpbx to complete a config reload. In the intervening time, I tore out the debian packaged ruby and reinstalled rubyy from source. Can't imagine that had anything to do with it though. My Voicepulse trunks are registered: OK (67 ms); but I've seen this before. They will pretend to register for a moment right after a reload or restart, but then go away almost immedia |
| 06:06.33 | [TK]D-Fender | that has nothing to do with registration |
| 06:06.38 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 06:07.46 | hesco1 | sip show registry shows only lots of 'Request Sent' and |
| 06:07.46 | hesco1 | asterisk.fivecats.org:5060 N hesco 120 Unregistered |
| 06:08.08 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:08.09 | hesco1 | which was the trunk I set up to test against jsharpe's server. |
| 06:08.26 | jpsharp | I did see the registration attempts. |
| 06:08.38 | jpsharp | But just starting 3 minutes ago. |
| 06:08.53 | hesco1 | great. so network out is working I guess. |
| 06:09.03 | jpsharp | [Oct 1 02:05:32] NOTICE[8497]: chan_sip.c:25765 handle_request_register: Registration from '<sip:hesco@asterisk.fivecats.org>' failed |
| 06:09.14 | hesco1 | now your server gives me: Request Sent |
| 06:10.57 | jpsharp | I'm getting the request and replying with "401 Unauthorized". |
| 06:11.03 | jpsharp | Which is what should happen. |
| 06:11.25 | hesco1 | how crazy would it be to rip out this 1.6.2.11 source install and rebuild * from the new source 1.8 tarball? |
| 06:12.54 | jpsharp | I think you've got a firewall issue. |
| 06:14.19 | jpsharp | I don't think you're getting any SIP replies from anywhere. |
| 06:14.28 | hesco1 | I fear intrusion. When I checked, my uptime was less than five days and my firewall was down. |
| 06:14.44 | jpsharp | do an "iptables -L" and pastebin the results. |
| 06:14.56 | hesco1 | I'm running iptables -L now and still waiting for it to respond. Its taking forever |
| 06:15.20 | hesco1 | here we go, coming your way |
| 06:17.52 | hesco1 | http://paste.scsys.co.uk/208852 |
| 06:19.26 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 06:19.33 | hesco1 | ACCEPT udp -- anywhere anywhere udp dpts:sip:sip-tls |
| 06:19.33 | hesco1 | I'm not seeing anything here which would block sip traffic. |
| 06:21.17 | *** join/#asterisk bipul (~vipul@unaffiliated/bipul) |
| 06:21.29 | jpsharp | Are you running fail2ban or something along those lines? |
| 06:24.45 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 06:27.38 | hesco1 | not that I am aware |
| 06:27.56 | hesco1 | would that show up in a ps listing? |
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| 06:28.09 | jpsharp | yeah. |
| 06:28.43 | hesco1 | then no, locate can't find anything like that either. |
| 06:29.07 | *** part/#asterisk jsjc (~Adium@221.Red-83-46-237.dynamicIP.rima-tde.net) |
| 06:29.26 | jaxon007_ | Guys, we have to purchase one server which can handle 100 Concurrent Conference calls with Each conference has 3 participants. Please guide me what type of server configuration suitable for us. |
| 06:30.04 | hesco1 | hmm, that looks like a useful tool. perhaps not for the pbx, but for some of our other servers. |
| 06:32.25 | jpsharp | jaxon007_: A big machine and running Asterisk 10 so you can use confbridge rather than meetme. |
| 06:33.25 | *** join/#asterisk Vince-0 (~androirc@196-215-188-244.dynamic.isadsl.co.za) |
| 06:33.48 | jpsharp | hesco1: I have no idea what's up. I'm still thinking it is a network/firewall issue, but I can't find anything pointing to iptables. |
| 06:35.27 | jaxon007_ | jpsharp: we are using asterisk 1.8 for our Applications. what should be configuration of server which can handle all calls. |
| 06:35.37 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 06:36.18 | jpsharp | I think asterisk 8 has confbridge. At least a dual core 2.5Ghz. |
| 06:36.37 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 06:37.57 | jpsharp | Couple of gigs of ram. Hard disk. All that jazz. |
| 06:40.04 | jaxon007_ | If I use meetme instead of ConfBridge. Will this required Higher configuration? If Yes please mention required configuration. |
| 06:42.21 | hesco1 | jpsharp: There is an appliance firewall between my server and the world. My agreement with the guy I rent rack space from is that he has me in his DMZ. My first suspicion was that something had changed with his configuration and I have written and called him, but so far without a response. So I started with a traceroute and was able to reach the box and trace from the box, which left me doubting my first inclination. But he's on my |
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| 06:46.08 | hesco1 | I was able to traceroute to hostmeup up in Canada (my server is in a DC in Atlanta GA). |
| 06:57.51 | hesco1 | [TK]D-Fender, jsharpe: than you both for your time tonight. I need to get some sleep before I have to get up and do it all again. Hopefully Ferris' appliance firewall is to blame and he can easily remedy this issue. We shall see in the morning (if I can get him to answer my calls). Otherwise I will be back to scratching my head on this one again tomorrow. |
| 07:01.12 | jpsharp | Yes, I'm certain it is the firewall. |
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| 07:29.59 | As001 | Hello I can see callerid when someone calls me on my pstn line but when I call dahdi/1 (telephone device) I can just see incoming. Softphone twinkle can see callerid... What this Panasonic TG1611FX need to show caller id ? |
| 07:30.47 | As001 | I tried everything loadzone defaultzone cidsignalling bell dtmf reverse polarity ... |
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| 07:38.57 | din3sh | hello all |
| 07:39.12 | din3sh | am experiencing occasional call drop with this msg |
| 07:41.36 | din3sh | asterisk[6320]: segfault at 0000000000000038 rip 00002aaac169e6b5 rsp 0000000041c23450 error 4 |
| 07:41.51 | din3sh | any idea why i would get this? |
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| 08:14.54 | wdoekes | din3sh: try to get a core dump from the crash and get a backtrace |
| 08:15.16 | wdoekes | and.. if you're not running the latest version of the branch, update to the latest one |
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| 08:21.39 | EmleyMoor | My Samsung Galaxy Nexus won't register with Asterisk any more, it seems - it just times out. Can't see anything on asterisk about it trying, but wondering if I can find out more... |
| 08:22.53 | *** part/#asterisk As001 (~uros@93-86-179-141.dynamic.isp.telekom.rs) |
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| 08:26.08 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 08:27.50 | EmleyMoor | Hmmm... looks like it's the phone |
| 08:40.04 | *** join/#asterisk din3sh (~din3sh@41.212.201.55) |
| 08:47.57 | EmleyMoor | Anyone here ever used Zoiper for Android? |
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| 08:56.21 | Marquel | morning. short question: can asterisk bind its sip-interface to a specific ip-address and use a sip-provider (as client) on a different IP-address? |
| 08:58.32 | WIMPy | You can bind to one or all IPs, not multiple. Not sure how to understand the 2nd part. |
| 09:00.16 | Marquel | WIMPy: asterisk registers itself to an external sip-provider. and that external sip-provider is not reachable from the ip-interface bind-addr points to. |
| 09:02.31 | Marquel | WIMPy: if that won't work, i'll bind to all interfaces... |
| 09:02.34 | *** join/#asterisk vlad_starkov (~vlad_star@194.186.188.92) |
| 09:05.58 | din3sh | WIMPy: i get occasional call drops with this in /var/log/messages ---> kernel: asterisk[6118]: segfault at 0000000000000038 rip 00002aaabd4be6b5 rsp 00000000401bd1c0 error 4 |
| 09:06.05 | din3sh | any idea? |
| 09:16.25 | WIMPy | Marquel: I haven't tried, but if you use tcp I'd expect that to work. |
| 09:16.37 | WIMPy | ~collectdebug |
| 09:16.37 | infobot | collectdebug is probably a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 09:16.44 | WIMPy | din3sh: ^^ |
| 09:18.22 | din3sh | thnx |
| 09:20.38 | Marquel | WIMPy: UDP only. but then that was just a question if anybody knows for sure. thanks for your time :) |
| 09:21.13 | din3sh | my log file has reache 1.4Gb |
| 09:21.14 | din3sh | :o |
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| 09:58.56 | hachque | I'm trying to get my SIP client to connect to Asterisk |
| 09:59.08 | hachque | but in the asterisk log it shows *CLI> [Oct 1 05:58:06] NOTICE[12222]: chan_sip.c:26004 handle_request_register: Registration from '<sip:redpointsoftware.com.au>' failed for '59.167.198.201:61049' - No matching peer found |
| 09:59.14 | hachque | and I don't really understand what that means |
| 10:03.16 | wdoekes | hachque: you're missing a username in the From/To header. your sip client is misconfigured |
| 10:07.18 | hachque | wdoekes: okay I added the 6000@ part to the account id |
| 10:07.22 | hachque | and then set the user to host = dynamic |
| 10:07.36 | hachque | now the client has the same "no suitable credential" error (using pjsua for testing) |
| 10:07.45 | hachque | but there's no messages in asterisk to even say that the registration was attempted |
| 10:08.19 | hachque | http://hastebin.com/kuhoqaxelu.avrasm that is the message in pjsua |
| 10:20.10 | *** join/#asterisk vlad_starkov (~vlad_star@188.123.252.83) |
| 10:24.11 | WIMPy | Marquel: Ooos. Sorry. Must have dreamt the TCP thing. No it won't work with UDP. |
| 10:24.15 | *** join/#asterisk Da-Geek (~Da-Geek@exch.imparando.com) |
| 10:24.20 | WIMPy | Mayebe with some iptables hackery. |
| 10:27.29 | wdoekes | iptables hackery will break the Via |
| 10:49.33 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 10:51.09 | mcolombo | hi all! |
| 10:51.39 | mcolombo | can asterisk send update package when i resume call from hold? |
| 10:56.44 | WIMPy | chan_sip doesn't seem to signal hold. |
| 10:56.54 | WIMPy | Which can be very annoying. |
| 10:59.21 | ectospasm | how does Asterisk know to play MOH then? |
| 11:00.03 | WIMPy | That's the other direction. |
| 11:04.59 | ectospasm | WIMPy: but, but... the hold is initiated at the phone! How does Asterisk know to play MOH if it never gets a hold signal? |
| 11:06.57 | mcolombo | phone send a re-invite to the asterisk |
| 11:07.05 | mcolombo | @ectospasm |
| 11:07.14 | *** join/#asterisk ChanServ (ChanServ@services.) |
| 11:07.14 | *** mode/#asterisk [+o ChanServ] by kornbluth.freenode.net |
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| 11:07.38 | mcolombo | but asterisk, does not send update/re-invite to the carrier/trunk sip |
| 11:08.38 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 11:08.58 | WIMPy | ectospasm: I said *chan_sip* doesn't seem to *signal* (as in send) hold. |
| 11:09.19 | ectospasm | WIMPy: ah, gotcha... |
| 11:09.26 | ectospasm | almost time for bed, methinks... |
| 11:10.42 | WIMPy | So if you use SIP from your Asterisk to some external conference and get a call, that's likely going to cause an epic fail. |
| 11:27.12 | *** join/#asterisk wonderworld (~ww@dsdf-4db5e168.pool.mediaWays.net) |
| 11:28.09 | hachque | i have asterisk set up with a simple configuration: http://hastebin.com/ramekalisa.ini |
| 11:28.25 | hachque | however, I can't connect to it with pjsua with pjsip-apps/bin/pjsua-i686-pc-linux-gnu --id=\<sip:620\> --registrar=sip:sip.redpointsoftware.com.au --username=620 --password=test |
| 11:40.17 | kasanop | hachque: have you read the pjsua manual? http://www.pjsip.org/pjsua.htm#acc |
| 11:43.23 | hachque | hmm I just tried with pjsip-apps/bin/pjsua-i686-pc-linux-gnu --id sip:620@sip.redpointsoftware.com.au --registrar sip:sip.redpointsoftware.com.au --realm \* --username 620 --password test instead |
| 11:43.31 | hachque | but still getting request timed out |
| 11:43.57 | hachque | that configuration I pasted is in a file called "/etc/opensips/sip.conf" |
| 11:44.03 | hachque | and I'm running asterisk -ccccv |
| 11:44.30 | hachque | http://hastebin.com/rudepicimu.apache |
| 11:44.36 | hachque | that's the log output of asterisk |
| 11:44.40 | jaxon007_ | we are using asterisk 1.8 for our Applications. what should be configuration of server which can handle all calls. there will be 100 Concurrent call in conference and Each conf has 4 Participants. |
| 11:49.01 | *** join/#asterisk robl^ (~robl^@pdpc/supporter/active/robl) |
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| 11:55.56 | kasanop | hachque: your asterisk didn't load any modules including chan_sip since it failed to load modules.conf |
| 11:56.09 | hachque | ah okay |
| 12:11.11 | hachque | I've added a modules.conf with the following content: http://hastebin.com/gekiyuvaxu.ini, but I still don't get any response from the server |
| 12:12.57 | hachque | this stuff is really complex :/ |
| 12:16.12 | *** join/#asterisk deo (~dnepangue@203.177.214.75) |
| 12:17.01 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
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| 12:31.56 | hachque | woohoo got the client registereing with the server |
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| 12:50.47 | carrar | woohoo |
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| 12:53.31 | [TK]D-Fender | When I feel heavy metal! |
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| 12:55.33 | bulkorok | whoohoo! |
| 12:55.34 | carrar | turn it up to 11 |
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| 13:34.12 | *** join/#asterisk irgendwer4711 (~irgendwer@reactos/tester/irgendwer4711) |
| 13:34.58 | irgendwer4711 | hi, I am using Asterisk 1.6 and have a problem with BLF. The led is blinking continously, even when phone is idle. |
| 13:35.16 | *** join/#asterisk kannan (~chatzilla@123.238.229.114) |
| 13:35.31 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 13:35.59 | kannan | hello, is it possible to set asterisk so that someone can simpy dial its IP address from a SIP phone and automatically send that to a dialplan context? |
| 13:36.33 | nunne | Does anyone have any experience with Juniper SGS5 firewall?? We have disabled SIP ALG and manually forwarded SIP + RTP span. But still 1-way audio on outgoing calls. Seems like it's re-writing inside/outside ports for some reason... Anyone have any input on this? |
| 13:37.09 | creativx | ssg-5 |
| 13:37.29 | nunne | sorry ;) |
| 13:38.01 | creativx | in case you were googling for sgs5.. :> |
| 13:38.31 | nunne | Ahh, no ;) I was googling it yesterday when i was out at this specific customer. Just trying to remember the name :D |
| 13:38.43 | [TK]D-Fender | kannan, [general] , "allowguest=yes", "context=generlschmucks" |
| 13:38.53 | nunne | seems like a solid product, good features.. Just dont understand how it can fall on something as simple as a plain port-forward |
| 13:39.11 | kannan | [TK]D-Fender , oj thats sounds easy enough thank you |
| 13:40.14 | irgendwer4711 | BLF-experts here? :-D |
| 13:43.00 | [TK]D-Fender | irgendwer4711, Are you going to tell us any useful details likee what PHONE you're even working with? Or an actual version of * if not an actual branch? Go SIP DEBUG to show us? Phone configs? |
| 13:43.54 | [TK]D-Fender | ~pb |
| 13:43.54 | 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. |
| 13:43.55 | irgendwer4711 | siemens openstage 40 |
| 13:43.56 | [TK]D-Fender | ^^^^^^^^ |
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| 13:46.30 | irgendwer4711 | http://pastebin.ca/2237946 |
| 13:46.44 | irgendwer4711 | its partially working |
| 13:47.11 | hachque | hey guys |
| 13:47.21 | hachque | I have linphone clients connected to asterisk |
| 13:47.25 | hachque | and the users defined in sip.conf |
| 13:47.28 | irgendwer4711 | it stops flashing, after the other phone hangs up, begin to flash later |
| 13:47.28 | hachque | but they can't call each other |
| 13:48.05 | hachque | http://hastebin.com/yabojegubo.ini |
| 13:48.13 | hachque | that's the user entries |
| 13:48.23 | hachque | I also have entries in extensions.conf but the extensions aren't even recognised |
| 13:48.26 | irgendwer4711 | [TK]D-Fender: console shows the hints and subscriptions |
| 13:48.38 | [TK]D-Fender | irgendwer4711, Which you should be showing us... |
| 13:49.07 | irgendwer4711 | 192.168.2.19 19 205b1dae62c4f8d 22@BLF-gs Idle dialog-info+xml <none> |
| 13:49.14 | [TK]D-Fender | hacpastebin the failed call for us to see with SIP DEBUG enabled. |
| 13:49.27 | irgendwer4711 | phone 19 is watching phone 22 |
| 13:49.42 | [TK]D-Fender | irgendwer4711, full dump of "core show hints". Do no waste time on partial displays |
| 13:51.26 | irgendwer4711 | http://pastebin.com/hW6BUPBb |
| 13:52.17 | irgendwer4711 | seems to be corecct |
| 13:52.24 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 13:53.19 | [TK]D-Fender | irgendwer4711, well if the phone is flashing then it's likely the phone's problem. What specific version are you running? |
| 13:54.24 | irgendwer4711 | <PROTECTED> |
| 13:54.42 | irgendwer4711 | I had setup a little testing box with it |
| 13:55.03 | [TK]D-Fender | irgendwer4711, that is an ancient version. Forget about support for it... or any 1.6 as it is.... |
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| 13:55.13 | [TK]D-Fender | irgendwer4711, Upgrade |
| 13:55.17 | irgendwer4711 | not possible |
| 13:55.35 | hachque | okay |
| 13:55.38 | [TK]D-Fender | irgendwer4711, Then I guess you'll have to live with it |
| 13:55.40 | hachque | we have users being able to dial each other |
| 13:55.43 | hachque | but they can't hear each other |
| 13:55.58 | [TK]D-Fender | hachque, Feel like telling us anything about your setup? |
| 13:56.14 | irgendwer4711 | <PROTECTED> |
| 13:56.14 | hachque | it's pretty much a default set up with just some users defined in the sip.conf |
| 13:56.20 | hachque | and some extensions in extensions.conf |
| 13:56.20 | santa0536 | guys can I have a question about asterisk t38 negotioation process? |
| 13:56.35 | [TK]D-Fender | irgendwer4711, If you say that 1.8 works .... how is it a CONFIG problem? |
| 13:56.59 | [TK]D-Fender | hachque, No, I mean REAL information .... like what NETWORKING is involved. |
| 13:57.08 | irgendwer4711 | I have read of many other with a working 1.6 and BLF |
| 13:57.12 | hachque | it's a server sitting at sip.redpointsoftware.com.au |
| 13:57.22 | hachque | we're connecting to it over the internet using linphone and csipsimple on android |
| 13:57.54 | [TK]D-Fender | hachque, So far I'm betting that clients are NAT'd an you have not configured any of this properly for it. |
| 13:58.15 | hachque | wouldn't the client software autodetect that? :/ |
| 13:58.25 | [TK]D-Fender | hachque, "directmedia=no", "nat=yes", and make sure if your * doesn't have a public IP on it that you've set the rest of that up properly |
| 14:00.19 | hachque | by * you mean sip.redpointsoftware.com.au? |
| 14:00.30 | [TK]D-Fender | 8 = ASTERISK |
| 14:00.31 | [TK]D-Fender | * |
| 14:01.11 | hachque | oh fair enough ;P |
| 14:01.38 | hachque | we've put our public IP addresses into the clients |
| 14:01.45 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:01.45 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:01.48 | hachque | and set directmedia=no, nat=yes onto each of the user entries in sip.conf |
| 14:01.53 | hachque | and reloaded sip |
| 14:01.57 | hachque | still no audio though |
| 14:02.12 | hachque | is there any commands I can run to give you more information? |
| 14:02.16 | santa0536 | Ok, I guess somebody will help :) I've found out that asterisk 1.8 performs a fall back to old codec via REINVITE after a 488 response to a T.38 REINVITE. Is it really nesessary? rfc3261 says like: |
| 14:02.16 | santa0536 | > If the other party does not accept the change, he sends an error response such as 488 (Not Acceptable Here), which also receives an ACK. However, the failure of the re-INVITE does not cause the existing call to fail - the session continues using the previously negotiated characteristics. |
| 14:05.03 | *** part/#asterisk irgendwer4711 (~irgendwer@reactos/tester/irgendwer4711) |
| 14:06.51 | [TK]D-Fender | hachque, "sip set debug on" |
| 14:10.17 | hachque | [TK]D-Fender: http://hastebin.com/yijifoniku.txt |
| 14:10.28 | hachque | that's the full output of making a call on the asterisk server |
| 14:11.29 | hachque | clients are 59.167.198.201 and 118.209.123.44 |
| 14:11.54 | hachque | server is 74.207.247.199 (sip.redpointsoftware.com.au) |
| 14:12.42 | [TK]D-Fender | hackeron, Check yoru firewalls on the server. 5060/10000-20000 (typical) all UDP need to be open |
| 14:12.53 | [TK]D-Fender | hachque, ^ |
| 14:13.11 | Katty | yawns, stretches |
| 14:13.14 | [TK]D-Fender | hachque, The rest looks OK |
| 14:13.17 | [TK]D-Fender | Katty, Mew. |
| 14:13.18 | Katty | pamples things |
| 14:13.23 | carrar | pokes Katty in the tummy |
| 14:13.23 | Katty | [TK]D-Fender: g'morning |
| 14:13.27 | Katty | carrar: ohai |
| 14:13.41 | hachque | [TK]D-Fender: I only have 5060 open in UDP, so opening 10000-20000 now |
| 14:14.06 | kannan | [TK]D-Fender , sip uri dial works fine thnaks again |
| 14:14.15 | carrar | Ohayoo gozaimasu! |
| 14:14.27 | [TK]D-Fender | hachque, SIP is noly a call setup protocol on 5060. the audio is RTP over a separate UDP range. 1 port per audio per direction, plus others for DTMF depending, etc |
| 14:14.55 | [TK]D-Fender | kannan, You're welcome |
| 14:14.59 | hachque | OMG it works! |
| 14:15.26 | carrar | OMG voip is so complicated! |
| 14:15.41 | Katty | errmagerddd verrip |
| 14:15.54 | hachque | thanks <3 |
| 14:17.13 | [TK]D-Fender | hachque, You're welcome |
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| 14:44.49 | hachque | hey guys |
| 14:45.01 | hachque | is it possible to set up asterisk in a situation like https://docs.google.com/drawings/d/1XgTG8IGE7dPXMBGcP6ThxDGrsds2gmCtfVvYqZBmhgY/edit |
| 14:45.15 | hachque | the VPS running asterisk and connecting to user 1 / 2 is now working |
| 14:45.41 | hachque | but i'm interested in permitting some users to use another VoIP device to call landlines / mobiles |
| 14:46.32 | carrar | You can do almost anything |
| 14:46.58 | hachque | okay cool |
| 14:47.48 | francisvgarcia | does anyone knows when the stable version of Asterisk 11 will be available? |
| 14:47.57 | drmessano | When its ready |
| 14:48.01 | [TK]D-Fender | francisvgarcia, Next eventually ... sharp |
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| 15:01.21 | *** join/#asterisk LemensTS (~matthew@adsl-99-126-109-24.dsl.stl2mo.sbcglobal.net) |
| 15:02.16 | LemensTS | how do i pass a variable on a transfered call? I set callerid to $callnum on an incoming call, and when i trasnfer that call to another sip phone i want to set $callnum as the callerid that shows up on its lcd instead of the sip phone that transfered the call. Asterisk 1.8 |
| 15:02.52 | LemensTS | do i need to set it as a global variable or a channel variable or something else? |
| 15:03.08 | mjordan | LemensTS: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information |
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| 15:11.16 | kasanop | LemensTS |
| 15:11.19 | kasanop | oops |
| 15:11.36 | kasanop | try 'o' option in Dial application |
| 15:14.45 | LemensTS | mjordan / kasanop: thanks I am looking at both your suggestions. |
| 15:15.08 | LemensTS | hopefully the o option works, it is less complicated :)_ |
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| 15:20.15 | dbeskoek | hi |
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| 15:26.51 | Katty | looks in |
| 15:28.19 | Katty | what's the word. |
| 15:29.41 | leifmadsen | bird is the word |
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| 15:38.40 | dbeskoek | running 1.8.15 there are some call-scenarios where "200 OK" (after INVITE) is sent by * to a device with missing contact-header, seems to be related to #17258 |
| 15:39.29 | dbeskoek | I'm currently unable to add a comment to this ticket |
| 15:42.12 | leifmadsen | dbeskoek: only the owner of the issue can comment at this point |
| 15:42.25 | leifmadsen | you might have an "Enter Feedback" button at the top |
| 15:42.34 | leifmadsen | but you likely can't see that unless you're a bug marshal |
| 15:42.39 | *** join/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 15:45.26 | dbeskoek | should I open a new one ? |
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| 15:50.08 | leifmadsen | no |
| 15:50.22 | leifmadsen | you could use the asterisk mailing list |
| 15:50.37 | leifmadsen | it's a failure in the jira workflow unfortunately which was never addressed |
| 15:51.12 | Qwell | If you're actually referring to 17258, you're talking about a 2+ year old ticket. |
| 15:51.38 | leifmadsen | Qwell: which has been updated as recently as Sep 24 |
| 15:52.06 | Qwell | Jul 10, 2010.. |
| 15:52.30 | leifmadsen | 24/Sep/12 11:20 AM |
| 15:52.40 | Qwell | leifmadsen: mantis |
| 15:52.49 | Qwell | ASTERISK-16026 |
| 15:52.55 | leifmadsen | I assumed he was talking about ASTERISK-17258 |
| 15:53.09 | leifmadsen | I realize the issue is old |
| 15:53.11 | Qwell | shrugs |
| 15:53.14 | leifmadsen | the comments is still recent |
| 15:53.20 | leifmadsen | s/is/are/ |
| 15:53.37 | leifmadsen | someone was asked to test against 1.8 |
| 15:53.41 | leifmadsen | so the issue could still be valid |
| 15:54.29 | *** part/#asterisk LemensTS (~matthew@adsl-99-126-109-24.dsl.stl2mo.sbcglobal.net) |
| 15:56.04 | dbeskoek | I think a "200 OK" after an INVITE should contain in any case a contact-header. the problem is only visible with a softphone and only 1 of 10 calls. After seeing the Jira-ticket I found the same 401/ACK-timing-problems. So this issue is still valid |
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| 16:04.30 | b1shop | can anyone recommend some US SIP providers that offer 1-8xx dial-ins |
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| 16:11.22 | ChannelZ | b1shop: voip.ms, vitelity... I think pretty much all of them do |
| 16:15.50 | b1shop | ChannelZ: thanks. I've only set up asterisk with fxo cards. never sip. |
| 16:17.36 | *** join/#asterisk nny (~Scott@174.107.223.14) |
| 16:20.13 | nny | I am setting up a system where a user can login via Agentlogin remotely or receive calls through their registered phone. This is an exclusive scenario, the peer will not be registered if they dial in. In this scenario, can I 1.) use the same ID for the sip peer as the agent and 2.) setup my dialplan to try both simultaneously? |
| 16:20.43 | nny | The agents will not be put into queues per se, each one is being sent calls via the AMI |
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| 16:32.15 | nny | hmm can Agent credentials be added via AMI? |
| 16:32.25 | nny | I don't see a related command in the AMi docs |
| 16:33.23 | kannan | while doing SIP URI dialling, when the asterisk server is on a public IP and the incoming context dials the call to a sip user (xlite) it works fine. The same thing does not work when i put the asterisk server behind a NAT, with port fwd'ing 5060 and rtp 10000-2000 on router |
| 16:33.50 | kannan | the call is established , but no audio at all |
| 16:33.59 | [TK]D-Fender | kannan, that is an improper NAT setup then |
| 16:34.10 | [TK]D-Fender | has nothing to do with "SIP URI" |
| 16:34.56 | kannan | i see |
| 16:35.05 | [TK]D-Fender | kannan, Your peers should all be "nat=yes", "directmedia=no", also under [general]. Set your localnet & externip, etc |
| 16:35.19 | kannan | ok thanks |
| 16:39.04 | nny | hey [TK]D-Fender working on a new setup, we have a remote database with users added to our system via a generation script. I want to add the ability to have them login as agents vs. their phones and send them calls via AMI. I have been reading up on the queue substructure, is it my understanding that 1.) agents can only be called via queue 2.) you cannot dynamically add agents via AMI and 3.) to attempt both the agent logged in via "call-in" or via their phone |
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| 16:39.35 | nny | just making sure I am not over engineering the process |
| 16:42.47 | [TK]D-Fender | nny, Dial(Agent/1234) |
| 16:42.55 | nny | [TK]D-Fender: ahh thanks |
| 16:43.08 | [TK]D-Fender | nny, Keeping in mind this implies you're on an OLD setup |
| 16:43.16 | nny | [TK]D-Fender: it's 1.8 |
| 16:43.26 | [TK]D-Fender | chan_agent was supposed to be dead last I checked |
| 16:44.04 | [TK]D-Fender | In favour of AQM/RQM |
| 16:45.01 | nny | [TK]D-Fender: that may be better |
| 16:45.20 | nny | [TK]D-Fender: I am importing values from a database, I could just use RQM |
| 16:46.49 | nny | [TK]D-Fender: the thing I am trying to avoid is using queues when each agent will be sent calls via AMI, I may have to create a queue for each agent |
| 16:48.28 | [TK]D-Fender | You should come up with another strategy.... |
| 16:49.05 | [TK]D-Fender | And a more complete call flow description to go around it |
| 16:49.43 | nny | [TK]D-Fender: let me pastebin it so I don't flood channel, I am researching using realtime as well |
| 16:52.47 | Corydon76-home | nny: in terms of your original plan, there is the capability to add via AMI, with QueueAdd |
| 17:01.32 | nny | I made an outline here, there are some unknowns as far as design ideas http://pastebin.com/AkqJJ2bs |
| 17:02.03 | nny | Corydon76-home: I assume this is used to add the queue definition but I would still need to define the agent member elsewhere |
| 17:02.41 | Corydon76-home | Uh, no, you can't add queues that way, only queue members |
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| 17:04.53 | Corydon76-home | btw, I use VMAuthenticate when I need to auth users for things like this |
| 17:04.55 | nny | Corydon76-home: yeah, this is essentially a 1 to 1 relationship, the agent is sent calls directly, the queue isn't needed if it can be avoide ( afaik apart from agent_chan it can't?) |
| 17:05.22 | Corydon76-home | Since it comes with its own method of changing the password, I figure it takes a lot of the trouble out of maintenance for me |
| 17:05.48 | Corydon76-home | Chan_agent is GONE, so forget agents |
| 17:05.50 | nny | Corydon76-home: we have a local database with credentials, I am using MYSQL to verfy the dial in part. The password is generated and sent via the web portal (and presented to the user) we use md5passwords for sip from the same database |
| 17:05.59 | nny | that's what I hear |
| 17:06.27 | Corydon76-home | Queue members are just SIP peer dialstrings |
| 17:07.09 | nny | I believe I have an idea how to approach it now |
| 17:07.12 | Corydon76-home | Seems like you're reimplementing Queue in a way |
| 17:08.16 | nny | Corydon76-home: I am going to have the agent dial in and enter thier user/pass. When they do they'll be dynamically added as an agent. I can then use AMI to also add that agent to it's own queue and send calls to it, does this sound sane? |
| 17:08.32 | Corydon76-home | Yes |
| 17:08.38 | nny | Assuming I can dynamically add agents via the dialplan |
| 17:09.03 | Corydon76-home | You also may want to specify joinempty=no on all your queues... |
| 17:09.22 | Corydon76-home | via the dialplan is with the app AddQueueMember |
| 17:09.34 | nny | yeah seems smart |
| 17:10.50 | nny | ok gonna make a test setup here and see what needs adjusting. I'll have to create each Queue manually before adding via AMI or can I generate each queue as needed? |
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| 17:11.39 | Corydon76-home | If you want to generate queues as needed, I think realtime is the only way to do it |
| 17:12.25 | nny | Corydon76-home: yeah may be superfluous I will generate the queues at the same time I generate the sip credentials. They are used to the 1 minute lag on new accounts etc. |
| 17:12.27 | Corydon76-home | otherwise a reload would wipe them out |
| 17:12.53 | nny | some day i'll learn a proper realtime language ;P |
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| 17:20.52 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 17:22.31 | jeffspeff | I just finished writing a nice little script to edit the iptables in realtime on my * box with info fed from the vpn logs... I'm calling it "Asterisk IPtables Dynamic Security" aka AIDS. So, let it be known that AIDS is actually good for your phone system! |
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| 17:24.00 | nny | That sounds like a post modification of the Holisitic IPTables Vanguard already out there |
| 17:24.15 | jeffspeff | nny, never heard of that |
| 17:25.12 | nny | jeffspeff: it was another poorly placed joke. Have you seen fail2ban? |
| 17:25.38 | jeffspeff | nny, lol, i've read a little about it, but never used it |
| 17:26.09 | nny | jeffspeff: it does the same thing essentially. It cralws logs for auth errors (like asterisk logs) and dynamically drops the source IP via iptables |
| 17:26.28 | nny | it also works with SSH etc. |
| 17:27.16 | jeffspeff | nny, this is more proactive instead of reactive... all of my remote phone users also have vpn accounts. this gets their public ip from the vpn log and sets that network to be allowed on the * server |
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| 17:27.26 | nny | jeffspeff: ahh nice |
| 17:28.04 | nny | jeffspeff: seems like a nice way to allow traffic from that IP without the VPN layer clogging things up |
| 17:28.39 | jeffspeff | yep |
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| 17:33.19 | nny | Corydon76-home: with a single queue member can I just eliminate the timeout and retry cycle? I really just want the system to try the single agent, if it fails for some reason then hangup. Not sure what values work for that |
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| 17:34.12 | nny | I have maxlen=0 |
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| 17:51.24 | nny | hmm does AgentLogin exist in 1.8? |
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| 17:58.34 | nny | I guess my question is what's the best method to have a user dial in and log in as an agent/wait for calls in 1.8 |
| 17:58.48 | nny | I assumed it was AgentLogin |
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| 18:04.14 | madduck | is it true that storing sip peers in realtime ODBC means that certain features won't work? |
| 18:05.24 | [TK]D-Fender | nny, as I said, that is just gone in 1.8 |
| 18:05.27 | [TK]D-Fender | (IIRC) |
| 18:05.30 | nny | [TK]D-Fender: ahh |
| 18:05.51 | nny | [TK]D-Fender: ok so if I want the agent to have the ability to call in to a number and wait for calls, do I have to do some channel bridging etc? |
| 18:06.31 | [TK]D-Fender | nny, Now I suppose you could rebuild somethin similar by leaving them parked on MoH and then using AMI to redirect them around, etc. A lot of work for sure |
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| 18:07.25 | nny | [TK]D-Fender: what's the 1.8 alternative? am i proper f**ked? |
| 18:07.36 | [TK]D-Fender | I just told you... |
| 18:07.45 | [TK]D-Fender | A lot of dirty work |
| 18:07.47 | nny | [TK]D-Fender: basically they need to be able to dial in as agents. Wow, that's bogus, why remove functionality lol |
| 18:08.21 | nny | [TK]D-Fender: so there's no native way for asterisk to allow dial in agents.. that's odd. |
| 18:08.43 | [TK]D-Fender | chan_agent = bai bai. Can't dial an agent when one doesn't exist |
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| 18:10.15 | nny | [TK]D-Fender: oddly that still exists in make menuselect for 1.8. So what if someone just wants to dial in and receive calls on that channel, the only way is to have AMI bridge each call to the waiting channel? Sounds like a horrible kludge |
| 18:10.27 | [TK]D-Fender | yup |
| 18:10.31 | nny | w t f |
| 18:10.33 | nny | why |
| 18:10.34 | nny | ? |
| 18:11.58 | nny | I am still confused, the whole point of this exercise was to allow agents an alternative to using a SIP peer. Now I find out that's basically a hack |
| 18:12.53 | nny | is frustrated as hell |
| 18:14.12 | nny | [TK]D-Fender: so having a user dial into a number and then have calls bridged to it wouldn't even use queues or agents, just channel bridge hackery |
| 18:14.52 | nny | which means there's no fail safes for multiple calls, or anything. WTF, jesus I feel like this PBX just got demoted |
| 18:15.02 | Corydon76-home | nny: because chan_agent is extremely buggy and prone to crashing, and we don't have a good way to fix it. |
| 18:16.37 | nny | Corydon76-home: is there some kind of general outline or idea on how to implement dial in agents? I feel lost after our first conversation |
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| 18:17.17 | Corydon76-home | Not that I know of |
| 18:18.06 | Corydon76-home | Everybody tends to use the callback mode |
| 18:18.34 | Corydon76-home | Well, you could use a conference call, I suppose |
| 18:18.47 | nny | oh yeah |
| 18:18.51 | nny | i have seen that before |
| 18:19.05 | nny | AgentCallbackLogin is deprecated, surely you don't mean that |
| 18:19.07 | nny | er removed |
| 18:19.31 | Corydon76-home | No, mean using AQM/RQM |
| 18:19.53 | nny | what is AQM? I understand Realtime Queue Management, but what's the A? |
| 18:20.03 | Corydon76-home | AddQueueMember |
| 18:20.08 | nny | oh |
| 18:20.25 | nny | makes more sense. Yeah but if the queue member is a dial in channel, that option is off the table |
| 18:20.41 | nny | I have seen the conference hack with old school stand alone dialers |
| 18:21.02 | nny | so agent joins conference, AMI sends call to confernece channel |
| 18:21.06 | nny | conference |
| 18:21.19 | Corydon76-home | with RedirectChannel, yes |
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| 18:23.08 | Corydon76-home | Agent hears the join signal to know to answer the call |
| 18:24.03 | Corydon76-home | And then you either Redirect the caller back out to a survey or you just hangup the channel to drop them out of the conference |
| 18:24.39 | nny | so I understand: 1.) Agent joins conference. 2.) AMI calls intended number 3.) AMI redirects channel to Meetme (with no announce, etc). 4.) Agent recieves notification that another person has joined, (will they hear ringing or the person?) |
| 18:24.52 | nny | I mean that would work, I may have to ask some questions as I implement it |
| 18:25.25 | Corydon76-home | Actually, step 3 is conditional between MusicOnHold or the conference, depending upon whether the agent is already with another caller |
| 18:25.46 | Corydon76-home | and you need a mutex around that test and join op |
| 18:26.00 | nny | Corydon76-home: they will be initiating calls, so there should only be a 1 to 1 relationship |
| 18:26.24 | Corydon76-home | Wait, the agent is calling out? |
| 18:26.27 | nny | Corydon76-home: I will have the dialplan check for current members of the conf call and fail if it's 2 |
| 18:26.35 | nny | Corydon76-home: the agent is dialing out via AMI (through a web app) |
| 18:27.07 | nny | Corydon76-home: essentially initiating the call through an AMI request |
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| 18:28.10 | Corydon76-home | Ah, fun. I'd still keep the join sound for the conference |
| 18:28.20 | nny | Corydon76-home: indeed, be awkward without it |
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| 18:28.32 | nny | Corydon76-home: I am thinking I can do the redirect through the dial plan |
| 18:28.37 | Corydon76-home | Unless you're doing the answer notify on the same web page |
| 18:29.11 | nny | Corydon76-home: probably not, I am not the web dev and they are a bit slow. I am doing this stand alone as a proof of conecept that it works and letting them mangle the AMI calls till they get it right |
| 18:29.34 | nny | I wish that was different lol, trust me |
| 18:29.44 | nny | so I am thinking |
| 18:29.57 | Corydon76-home | Well, you're going to get the answer notify or conf join notify via AMI, anyway, as an event |
| 18:30.23 | nny | Corydon76-home: good point, i'll try to implement that in my test scenario as some kind of update |
| 18:31.20 | nny | Corydon76-home: if the AMi sends a call to local channel, does it progress after it's answered or while it is ringing? |
| 18:32.11 | nny | or do I have a choice there? |
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| 18:33.20 | Corydon76-home | I honestly don't know |
| 18:33.20 | citywok | nny: assuming there isn't something i've missed, then when it's answered the call will progress through the dialplan. |
| 18:33.52 | citywok | we dial all of our calls through the AMI with a local channel dialing the agent's SIP phone, and the other side dialing outbound |
| 18:33.54 | Corydon76-home | Do you mean progress through the dialplan, or do you mean when it will send the Progress indication? |
| 18:34.40 | nny | Corydon76-home: through the dialplan, I think citywok answered it |
| 18:35.56 | nny | So AMI dials 8435551212 and sends it to my local channel along with a variable to tell me what agent (conference) it's destined for. Asterisk checks that the conference has 1 occupant and redirects the channel to the conference. The agent hears a join sound and responds with "hello, blah blah". |
| 18:38.02 | citywok | what are you trying to accomplish? |
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| 18:42.25 | nny | citywok: sorry one sec, thanks for asking |
| 18:42.29 | Corydon76-home | citywok: he's got an Agent calling in once, then making several calls |
| 18:42.38 | citywok | Ah, I scrolled up. It looks like you are trying to make it os the agent can call in to the PBX, sit in a conference, hit a "dial" button somewhere that will tell the PBX to make a call and what conference they are in to bridge them to? |
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| 18:44.54 | nny | citywok: Corydon76-home reqoringt he plan one sec |
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| 18:49.05 | citywok | yea, confbridge or meetme is probably the way to go. when the person dials in have them enter their "pin" which will be their predetermined conference pin (before meetme, that way you can verify it), and mark them as "in" the meetme conference, and when they leave mark them as out of it to avoid problems |
| 18:49.21 | nny | citywok: ok changed it a bit. |
| 18:49.38 | nny | citywok: the best way for this to work is to bridge the channel to the user as the call is placed (while ringing/connecting etc) |
| 18:50.20 | nny | citywok: can I just originate a call via AMI and redirect it to an agent waiting on another channel (that isn't a conference)? I may have them pointed at MOH while they wait (per [TK]D-Fender's suggestion) |
| 18:50.20 | citywok | yea makes sense. you could get the same effect by playing ringing tone as the hold music, and as soon as the person is joined the ringing will go away |
| 18:51.24 | citywok | i imagine you could originate a call with a uniqueid, then perform a redirect on it and get that same effect |
| 18:51.39 | nny | citywok: I think the feedback of failed calls (not all calls go through) would be better as an audio experience similar to dialing themselevs. I would normally have some fancy addition to their portal but I don't control that aspect and for continuity between using a sip phone or dial in |
| 18:51.48 | citywok | except you will probably lose the agent when the call is over, and they would need to dial back in |
| 18:52.17 | citywok | yea, makes sense |
| 18:52.17 | nny | citywok: hmm no way around that? |
| 18:52.35 | citywok | nny you could work around it by having a jump back to meetme after the dial(whoevertheagentiscalling) |
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| 18:52.59 | citywok | so they get put back in the conference until the next call, and then redirected to the direct channel |
| 18:53.09 | citywok | it sounds really, really ghetto tbh, but i can picture how it would work mentally. |
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| 18:53.35 | nny | citywok: i agree, sadly I only have ghetto as options here |
| 18:54.07 | nny | citywok: could I just jump back to the music on hold app or is conference needed? |
| 18:55.07 | nny | i.e. Dial in -> moh -> AMI dials (with option j) the call -> call ends -> agent sent back to moh via the jump/ repeat? |
| 18:55.29 | citywok | Agent Calls In, placed on MoH app or conf -> Presses Dial (8002662278) -> AMI originates (actually a redirect) to local/8002662278, the local channel sends the person back to MoH at the end of hte call. |
| 18:55.57 | citywok | i would use a redirect to create the actual dial to get the effect you want |
| 18:56.10 | nny | citywok: hmm I like the local redirect |
| 18:56.37 | citywok | Yea, I think that's the best way I can think of. |
| 18:56.42 | nny | assuming I can tell AMI which channel to redirect |
| 18:56.46 | citywok | you can |
| 18:57.04 | citywok | http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Redirect |
| 18:57.23 | nny | citywok: I mean pass that information to the side using AMI |
| 18:57.45 | citywok | When the agent dials in have htem enter their "passcode" and ASTDB(CHANNELS/PASSCODE) |
| 18:57.54 | nny | citywok: ooh |
| 18:57.57 | citywok | set the channel they are on in astdb, and use that via the dial |
| 18:58.01 | nny | citywok: clever |
| 18:58.03 | citywok | ;) |
| 18:58.23 | citywok | once they hang up clear the thing and you know they are no longer there and throw an error on the dial button |
| 18:58.46 | nny | ok gonna get started, thank you citywok |
| 18:59.04 | citywok | yep. i bet with that information you can do it in an hour or less. lol. |
| 18:59.32 | nny | citywok: yeah it seems a lot simpler now. I have some areas I haven't mentioned, but they are more dialplan semantics during the call than over all design |
| 19:00.25 | citywok | yea, i kinda want to do it just to see how long it would take to pull off. |
| 19:00.49 | nny | citywok: the AMI side is gonna be some simple php code I put up to test it, that will be the part that holds me up, my business partner is the php wiz |
| 19:01.04 | nny | citywok: let me see how far I can get stand alone for testing |
| 19:01.20 | citywok | nny it's a very simple fsockopen() and you just pass in the AMI commands via fputs() |
| 19:01.42 | nny | citywok: yeah let me see I already have an example php dial page setup |
| 19:01.55 | nny | should be able to mangle it to recirect |
| 19:01.57 | citywok | sec, i've got one |
| 19:01.58 | nny | redirect |
| 19:03.33 | citywok | nny send me a PM |
| 19:03.42 | citywok | for some reason i can't open a PM window, don't want to paste in here :p |
| 19:04.25 | nny | citywok: sent |
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| 19:19.03 | hesco1 | [TK]D-Fender, jsharpe: Well, I spoke with the guy hosting my 1u in his rack. He logged in to his appliance and assured me that my server remains in his DMZ. I just returned to the source code left in /usr/local; and rebuilt asterisk 1.6 from scratch, but now I don't even have a sip command at the *CLI> prompt. I'm quickly running out of ideas for what to try. Will have to head into the office now. Will return to this project once |
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| 19:19.46 | hesco1 | guess my next step might be an asterisk 1.8 source install and start over yet again. |
| 19:22.15 | [TK]D-Fender | Maybe you should actually look at the problem |
| 19:22.28 | [TK]D-Fender | you know. .... prove you have configs. try loading the module by hand... |
| 19:22.35 | hesco1 | maybe I was too impatient: I now see: SIP channel loading... |
| 19:22.46 | hesco1 | how would I load a module by hand? |
| 19:23.05 | [TK]D-Fender | module load [themodule] |
| 19:23.13 | [TK]D-Fender | "chan_sip.so" |
| 19:23.31 | hesco1 | config list shows all expected sip*.conf files |
| 19:24.41 | hesco1 | ok, that loaded sip |
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| 19:24.49 | hesco1 | and made sip commands available again |
| 19:25.23 | Qwell | citywok: for future reference, /query |
| 19:25.46 | nny | citywok: getting closer, may have some questions about the php script and how it relates to the local channel |
| 19:26.22 | hesco1 | and sip show registray shows all my ITSP vendors, with state: Request Sent or Unregistered |
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| 19:32.26 | nny | citywok: when you see this: I am reading over the AMI command structure, should i execute the redirect via the AMI field Application or via the context on connect variable? |
| 19:34.37 | nny | citywok: I think per our original discussion the redirect sends the channel to local XXXXXXXXXX where it is then dialed |
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| 19:51.31 | nny | http://pastebin.com/h7HqfhY1 is returning a broken pipe error, what did I fail at? |
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| 19:54.13 | nny | nm my channel variable needs some work, working directly with the channel name now |
| 19:54.42 | nny | now trying to figure out how to send this to the local channel, Executing [18436842813@agent-local:1] Dial("SIP/42-00060502", "18436842813@vitel-outbound2,,rg") in new stack failed |
| 19:57.33 | nny | http://pastebin.com/XvQRLdfc is wrong, trying to figure out why. |
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| 19:58.24 | nny | it's trying to dial the redirect channel, then the channel. I did this wrong but not sure why |
| 19:59.59 | nny | oh jesus |
| 20:00.01 | nny | i am an idiot |
| 20:00.03 | nny | SIP/EXTEN |
| 20:00.07 | nny | lol |
| 20:01.27 | nny | wishes I could take back my irc channel spam and hide my blatant error |
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| 20:06.20 | oquidave | hello guys |
| 20:06.51 | oquidave | am trying to load the chan_mobile.so module, but i get ... ERROR[3544]: chan_mobile.c:4164 mbl_load_adapter: Skipping adapter blue. Voice setting must be 0x0060 - see 'man hciconfig' for details. |
| 20:07.49 | oquidave | i've a "Cambridge Silicon Radio Bluetooth Dongle (HCI mode)" bluetooth usb dongle on my opensuse12.1 with kde4.9. it's recognized |
| 20:08.13 | oquidave | so when i start asterisk and try to load the mobile module, i get that error |
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| 20:12.37 | nny | citywok: pretty much works hahahah |
| 20:13.11 | nny | citywok: i'll pm you the pastebin of my dialplan and the changes to my php AMI call. I am still working with the later to fix some small issues but overall it works |
| 20:15.39 | drmessano | oquidave: See if this affects you: https://issues.asterisk.org/jira/browse/ASTERISK-16357 |
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| 20:26.00 | oquidave | drmessano: kind of ...but not very helpful. but thanks |
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| 20:30.11 | mjordan | oquidave: so the patch did not change anything? |
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| 20:42.51 | tompaw | Blame Canada! |
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| 21:47.46 | ibercom | Anybody know what's means "app_dial.c: Unable to forward frametype: 2". I have a lot of errors. |
| 22:02.37 | pabelanger | ibercom, Warnings != Errors |
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| 22:12.04 | mjordan | ibercom: A frametype of 2 corresponds to a voice frame |
| 22:13.09 | mjordan | ibercom: you probably have something that is attempting to send media (voice) to a channel before it has been answered, and the channel technology does not support the delivery of media prior to it being answered |
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| 22:26.28 | ibercom | mjordan, I only use dahdi and sip, I will review dialplan, not agi. That I have to look out for? |
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| 22:30.23 | ibercom | pabelanger, but a lot of warnings isn't good. I need look for the cause. |
| 22:31.31 | pabelanger | ibercom, are you having problems? |
| 22:33.07 | mjordan | ibercom: you should find out what channel is upset that media has been played onto it, i.e., what are you dialing. You should find out if said channel supports early media. |
| 22:33.10 | ibercom | No problem, but I think that warnings are 'dangerous'. |
| 22:36.40 | ibercom | The primary support early media, I have 4 PRI. Do I need to put early media in sip ? |
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| 22:40.18 | ibercom | mjordan, Normally, I do dial and player, not always Answer the channel. Is it automatic ? |
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