| 00:01.01 | dlublink | pabelanger, I assume 58 seconds because the Event duration was about 58000 in the packet, and I seem to remember that value being in milliseconds |
| 00:02.46 | dlublink | my next step is I did rm -rf /etc/asterisk and am going to setup a single extension for a test call. |
| 00:05.34 | dlublink | pabelanger what further information need I give so you can help me |
| 00:08.06 | pabelanger | dlublink: well you need to narrow down where the issue lies. With our Asterisk tests, we rely a lot on DTMF to pass information between 2 asterisk instances. That being said, DTMF is pretty well tested in those scenarios. |
| 00:08.29 | dlublink | Any suggestions as how to narrow it down ? |
| 00:09.12 | pabelanger | test phone to asterisk, then asterisk to asterisk. SendDTMF() |
| 00:10.42 | *** join/#asterisk digilink (~digilink@vps.stephennet.net) |
| 00:11.09 | dlublink | ok, I tested phone to asterisk, it works because I see the dtmf events in the console ( I use logger dtmf to see it ). |
| 00:12.48 | dlublink | I think I'll try senddtmf |
| 00:23.09 | dlublink | Phone to asterisk is ok, the packet sniff shows that the packets are perfecet |
| 00:23.11 | dlublink | perfect* |
| 00:31.26 | dlublink | pabelanger : I added this to extensions.conf |
| 00:31.26 | dlublink | exten => 22,1,Dial(SIP/114186939930@pri1.omnity.net,30,D(wwwwwwww123889047143#)) |
| 00:31.49 | dlublink | I dialed 22 from my SIP phone, and the second asterisk server received "113800774" |
| 00:36.46 | dlublink | the packet sniff shows that the DTMFs are missing from the server |
| 01:15.20 | nix8n82 | dlublink, are these two servers on the same network? |
| 01:16.36 | nix8n82 | are you doing any jitter buffering? |
| 01:25.45 | *** join/#asterisk wonderworld (~ww@port-92-201-22-157.dynamic.qsc.de) |
| 01:35.02 | *** join/#asterisk nix8n82 (~nate@24.143.27.157) |
| 01:41.13 | ectospasm | mmm, jitter butter |
| 01:47.29 | *** join/#asterisk nix8n82 (~nate@24.143.27.157) |
| 01:48.13 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 01:52.24 | *** join/#asterisk Tech_Travis (~Travis@cpe-76-168-191-127.socal.res.rr.com) |
| 01:54.26 | dlublink | Yes |
| 02:00.01 | pabelanger | dlublink: create a peer in sip.conf, Then you can Dial(SIP/peername/114186939930) |
| 02:00.02 | pabelanger | is cleaner |
| 02:00.25 | dlublink | ok |
| 02:03.39 | dlublink | Ok, I'll try that |
| 02:05.58 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 02:09.42 | *** join/#asterisk micols (~ident@rlogin.dk) |
| 02:11.58 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 02:22.39 | *** join/#asterisk joshaidan (~brianj@24.109.210.41) |
| 02:25.23 | *** join/#asterisk JunTao2 (~asdf@CPE0021296828b2-CM00111ae6f860.cpe.net.cable.rogers.com) |
| 02:25.47 | JunTao2 | do you guys think this is a good ATA product/deal? http://www.telephonydepot.com/Catalog/Grandstream-Analog-Adapters/Grandstream-HandyTone-503-HT503 |
| 02:32.05 | *** join/#asterisk luisfelice (~luisfelic@190.39.213.145) |
| 02:34.13 | luisfelice | Hi all, I have an issue with two asterisk systems. I have one 1.4 with Zaptel and the other 1.8 with dahdi, when an analog extension of the ast1.4 calls an analog extension of the 1.8 always the analog extension of the ast1.4 listen bad and the extension of the ast1.8 listen perfect. |
| 02:34.47 | luisfelice | Any idea? |
| 02:36.16 | pabelanger | JunTao2: No, grandstream does not get good reviews around here. |
| 02:37.26 | JunTao2 | pabelanger is that a fact eh. Why not? Any other recommendations? HOw about this cheap one http://www.thevoipconnection.com/store/catalog/Grandstream-HandyTone-HT-286-p-16211.html |
| 02:38.07 | *** join/#asterisk titter (~Justin@173.14.85.173) |
| 02:38.50 | pabelanger | JunTao2: cheap and VoIP don't go hand in hand :) Have you looked at the SPA-3102? |
| 02:39.18 | JunTao2 | nope |
| 02:39.49 | pabelanger | Linksys sells it |
| 02:39.57 | pabelanger | ~spa3102 |
| 02:40.34 | *** join/#asterisk bongfrog (~winston@c-67-165-236-171.hsd1.co.comcast.net) |
| 02:43.48 | *** join/#asterisk luisfelice_ (~luisfelic@190.39.213.145) |
| 02:44.08 | luisfelice | Sorry I lost my connection, anyone can help me? |
| 02:51.20 | pabelanger | luisfelice: define bad |
| 02:52.05 | luisfelice | pabelanger: like cut audio |
| 02:52.26 | pabelanger | luisfelice: how are you connecting the 2 asterisk boxes together? |
| 02:52.35 | luisfelice | pabelanger: IAX2 |
| 02:52.52 | luisfelice | pabelanger: each one have voip phones that works fine |
| 02:53.04 | luisfelice | pabelanger: inclusive calling voip to analog |
| 02:53.11 | pabelanger | luisfelice: more then likely a bandwidth issue |
| 02:53.13 | luisfelice | pabelanger: te problem is only analog to analog |
| 02:53.40 | pabelanger | are you using the same codecs for all your calls? |
| 02:53.49 | luisfelice | pabelanger: yes |
| 02:53.57 | luisfelice | pabelanger: ulaw |
| 02:54.09 | pabelanger | luisfelice: jitterbuffer enabled? |
| 02:54.19 | luisfelice | pabelanger: yes |
| 02:55.15 | pabelanger | I would confirm you are using the same codecs around all connections. I can't see analog making a difference in this setup |
| 02:55.34 | luisfelice | pabelanger: I thought maybe can be an incompatibility between calls originated by zaptel and dahdi |
| 02:56.26 | pabelanger | luisfelice: no, your calls are not Zaptel to dahdi directly, they both interface to IAX2 |
| 02:57.20 | luisfelice | pebelanger: yes I know, but that is the only thing different, I try codecs, check bandwidth, etc etc |
| 02:57.40 | luisfelice | pebelanger: allways the ast1.4 analog listen bad |
| 02:58.33 | luisfelice | pebelanger: inclusive a PSTN call from the ast1.4 originated in the analog extension of the ast1.8 listen bad |
| 02:58.43 | luisfelice | pebelanger: at the PSTN side |
| 02:59.11 | pabelanger | and analog to voip? |
| 02:59.23 | luisfelice | works good |
| 02:59.55 | pabelanger | ~collectdebug |
| 02:59.55 | infobot | somebody said collectdebug was a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 03:00.19 | pabelanger | luisfelice: pb a debug log from asterisk 1.4 for both a good and bad call ^ |
| 03:02.06 | luisfelice | IAX debug? |
| 03:02.32 | pabelanger | luisfelice: yes, follow the wiki page and enable IAX debug |
| 03:02.38 | luisfelice | ok |
| 03:18.08 | *** join/#asterisk luisfelice (~luisfelic@190.39.213.145) |
| 03:20.21 | luisfelice | pebelanger: I saw this many times: [Apr 23 03:10:28] DEBUG[8209] chan_zap.c: Dropping frame since I'm still dialing on Zap/1-1... |
| 03:21.08 | pabelanger | luisfelice: pb log |
| 03:21.11 | luisfelice | ok |
| 03:26.00 | luisfelice | http://pastebin.com/sfyaPJVA |
| 03:28.26 | pabelanger | luisfelice: Did you answer the call? |
| 03:28.34 | luisfelice | yes |
| 03:30.37 | pabelanger | luisfelice: is this the full debug log? Because it is not showing that Zap answered the call.... can could be the reason for your audio issue, since frames are getting dropped |
| 03:32.14 | luisfelice | oh sorry no, i did it bad, let me pb again |
| 03:37.54 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 03:39.41 | luisfelice | http://pastebin.com/jrsCeX7s |
| 03:48.34 | *** join/#asterisk radic (~radic@dslb-178-002-228-100.pools.arcor-ip.net) |
| 03:56.43 | luisfelice | any idea? |
| 04:02.09 | luisfelice | I will comeback later |
| 04:12.02 | *** join/#asterisk luisfelice (~luisfelic@190.39.213.145) |
| 04:12.10 | *** join/#asterisk Josejulio (~patty@189.202.4.236.cable.dyn.cableonline.com.mx) |
| 04:12.51 | luisfelice | pabelange: I am back, did you got time to see the log? |
| 04:17.21 | Josejulio | Good day, sorry for the long post: i'm gathering info for a project i'm working on, we wanna link our system with an asterisk server, one of the things i'm looking for is a module that "hook" on the iax2 auth process, what i intend is to use other server to do the auth stuff (like, user sends user/pass to asterisk, asterisk sends it to a server B, server B process it and answers YES/NO, and so asterisk answers to the user YES/ |
| 04:17.22 | Josejulio | this possible with a module or should i modify the asterisk-core? Thanks for reading. |
| 04:21.31 | *** part/#asterisk luisfelice (~luisfelic@190.39.213.145) |
| 04:32.54 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 04:36.17 | *** part/#asterisk joshaidan (~brianj@24.109.210.41) |
| 04:45.54 | *** join/#asterisk Ean (~Ean@unaffiliated/ean) |
| 04:47.51 | *** join/#asterisk exes (~exes@wisdom-eth0.enasis.net) |
| 04:48.34 | exes | random question, could Asterisk accept a text message and, using either a plugin or external script, allow me to interact with that text message over XMPP? |
| 04:51.15 | *** join/#asterisk luisfelice (~luisfelic@190.39.198.95) |
| 05:13.30 | *** join/#asterisk luisfelice (~luisfelic@190.39.198.95) |
| 05:20.43 | *** join/#asterisk jplank (~G_Bove@208-104-67-26.dyn.fttp.comporium.net) |
| 05:26.02 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 05:32.07 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 05:37.13 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 05:43.48 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 05:45.07 | kuku | If I have hints, but 0 watchers, after I restart asterisk... but watchers show up after the phone sits there for a while and makes a few calls - what could be the issue ? |
| 05:46.39 | kaldemar | you tell us |
| 05:47.02 | jplank | phone not subscribing soon enough? |
| 05:47.10 | kuku | makes sense. |
| 05:47.21 | kaldemar | the phones subscribe to hints. they do it in intervals. restart clears subscriptions. |
| 05:47.25 | jplank | what does a debug show? |
| 05:47.28 | kuku | This is a 6757i phone running on 1.8 |
| 05:47.40 | jplank | what kaldemar said |
| 05:48.08 | kuku | so I need to force the phone to subscribe to hints more often |
| 05:48.31 | jplank | or restart asterisk less |
| 05:48.56 | jplank | should probably figure out why your restarting it so often, and stop doing that |
| 05:49.42 | kuku | i did it on purpose. |
| 05:49.59 | kuku | b/c it never subscribed to my parking lot hints. |
| 05:50.04 | jplank | I understand that |
| 05:50.59 | kuku | So its only subscribing to extensions - not my parking long hints. I have no clue where to go from here to have the phone subscribe to the arking lot hints. whoa, I think i just said the same thing twice, i must be tired. |
| 05:57.21 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 05:58.55 | kaldemar | kuku: poke the phones. you can't make them subscribe from asterisk. |
| 06:04.24 | kuku | yeh - no subscriptions |
| 06:05.02 | kuku | cisco 7940's just re-registered. |
| 06:15.58 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 06:19.16 | *** part/#asterisk Tech_Travis (~Travis@cpe-76-168-191-127.socal.res.rr.com) |
| 06:28.47 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 06:34.01 | p3nguin | What the hell are parking lot hints? |
| 06:35.54 | p3nguin | I know what a parking lot is, and I know what hints are, but I have never used the two terms in conjunction in the manner in which you have. |
| 06:39.36 | *** join/#asterisk OldMonk (~raju@122.176.204.66) |
| 06:39.40 | OldMonk | hi |
| 06:40.53 | OldMonk | got an asterisk server (1.6.2.9) with 4000 sip peers, about 1000 of them connected. i'm seeing mass connects from the clients, followed by lag and then disconnect. after a while the client connects again. any idea why this could be happening, or how to mitigate it? |
| 06:42.29 | ectospasm | that's an *old* version of Asterisk. And, the Asterisk 1.6.2* branch was deprecated Thursday, which means nothing but security fixes until it's end of life next year |
| 06:43.39 | OldMonk | no option at the moment: waiting for debian packages. |
| 06:43.47 | OldMonk | messages are of the form: [Apr 23 12:13:30] NOTICE[27465] chan_sip.c: Peer '21007219' is now UNREACHABLE! Last qualify: 52 |
| 06:43.53 | p3nguin | They are in the asterisk repo. |
| 06:44.08 | p3nguin | current, several branches. |
| 06:44.13 | ectospasm | OldMonk: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 06:44.22 | ectospasm | There are Digium-sponsored Debian repos |
| 06:44.44 | OldMonk | ectospasm: will do that if i'm sure that will solve this problem |
| 06:44.55 | jplank | I'd think if you were admining a box with 4000 extensions, you'd know how to build from source |
| 06:45.08 | OldMonk | i do |
| 06:45.14 | ectospasm | OldMonk: there's always the change log (-; |
| 06:45.46 | jplank | OldMonk: the first thing to do is see if the phones are replying to the OPTIONS message |
| 06:45.55 | jplank | are all the phones inside the same network or remote? |
| 06:46.01 | OldMonk | well, i'm trying to first figure out if it's a bug in the client, in asterisk or a fault in the LAN |
| 06:46.10 | OldMonk | jplank: same building. multiple VLANs |
| 06:46.16 | jplank | what does a capture show you? |
| 06:46.32 | jplank | are you seeing responses to the the OPTIONS message? |
| 06:46.32 | OldMonk | on the asterisk box? |
| 06:46.39 | jplank | yea |
| 06:47.11 | jplank | Asterisk will send the client an OPTIONS message, and then client needs to respond |
| 06:47.11 | OldMonk | man, there's so much traffic on that box it'll be next to impossible to isolate one missed OPTIONS response! |
| 06:47.28 | jplank | thats what captures and filters are for |
| 06:47.36 | p3nguin | You can debug by ip or by peer. |
| 06:48.05 | jplank | if you can't start troubleshooting from the right place, it will take you forever to figure out the issue |
| 06:48.10 | jplank | install wireshark |
| 06:48.35 | jplank | tshark -i any port 5060 and host 1.2.3.4 |
| 06:49.02 | OldMonk | for me tcpdump with capture to file may be easier. i can perl out the OPTIONS and responses |
| 06:49.03 | ectospasm | <3 wireshark |
| 06:49.11 | OldMonk | OPTIONS will go to port 5060, right? |
| 06:49.16 | ectospasm | yes |
| 06:49.26 | jplank | thank you ectospasm I usually get flack for saying that instead of debug :) |
| 06:49.45 | jplank | OldMonk: are all the peers going unreachable at the same time? |
| 06:49.57 | OldMonk | not all, some 60-70 out of 1000 at a time |
| 06:50.11 | jplank | are they ones within the same VLAN? |
| 06:50.22 | OldMonk | need to check... holdon |
| 06:50.35 | ectospasm | jplank: using CLI debug for SIP is like digging a massive hole with an ice cream tasting spoon! |
| 06:51.15 | jplank | I concur, but I always get it thrown in my face that you also get CLI output |
| 06:51.25 | p3nguin | It can be done, but takes a long time? |
| 06:51.37 | ectospasm | Once someone showed me the wireshark way, I only go back to the Asterisk CLI debug if I need to see non-SIP-debug CLI messages (since tcpdump et al. don't capture that) |
| 06:51.50 | OldMonk | nope, different VLANS |
| 06:51.52 | jplank | agreed |
| 06:52.27 | ectospasm | jplank: granted, we always request both, just in case one or the other is incomplete |
| 06:52.29 | OldMonk | jplank: different VLANs |
| 06:52.31 | jplank | OldMonk: there's any one of a hundred reasons that could be happening |
| 06:52.48 | jplank | start with taking a capture and seeing if the phones are even responding to the options messages |
| 06:53.09 | jplank | it could be as simple as a bad cable somewhere |
| 06:54.34 | OldMonk | woah, here's something right away: http://pastebin.ca/2049617 |
| 06:54.36 | jplank | if you turn qualify off, do the phones start working? |
| 06:54.42 | OldMonk | picked a client at random |
| 06:54.51 | jplank | well there's your problem |
| 06:55.05 | jplank | check yo network |
| 06:55.05 | OldMonk | why TF would that happen? |
| 06:55.44 | OldMonk | umm, this looks more like a client problem... network issue would just time the packet out, this looks like the client sending a PORT_UNREACH |
| 06:56.08 | jplank | are you getting one way audio? |
| 06:56.09 | p3nguin | That reminds me of a firewall. |
| 06:56.34 | jplank | what does the 5060 traffic look like? there's no reinvites going on, are there? |
| 06:56.46 | OldMonk | jplank: hmm, it's possible... users say calls take ages to mature with blank gaps |
| 06:57.14 | OldMonk | anyone got a quick tcpdump expression to filter out the actual voice traffice? |
| 06:57.25 | jplank | I gave you wireshark |
| 06:57.37 | OldMonk | p3nguin: no firewalls between host and client |
| 06:57.56 | OldMonk | jplank: no gui access to the server, just ssh |
| 06:58.33 | jplank | you don't need gui for wireshark |
| 06:58.45 | jplank | you said debian right? |
| 06:58.49 | OldMonk | jplank: yes |
| 06:58.51 | jplank | apt-get wireshark |
| 06:58.54 | jplank | or whatever |
| 06:59.14 | p3nguin | You can use tshark, which is part of wireshark, and is CLI based. |
| 06:59.27 | OldMonk | ah, ok |
| 06:59.35 | jplank | (2:48:35 AM) jplank: tshark -i any port 5060 and host 1.2.3.4 |
| 06:59.40 | p3nguin | man 1 tshark |
| 07:00.58 | p3nguin | It writes in libpcap format, so you can read the saved file in the same way as you would a tcpdump saved file. |
| 07:01.27 | jplank | -w to save to a file |
| 07:01.39 | jplank | tshark -i any -w capture.cap port 5060 and host 1.2.3.4 |
| 07:02.08 | OldMonk | installing |
| 07:05.10 | OldMonk | still one issue though: there's not guarantee that host 1.2.3.4 will go unreachable |
| 07:05.46 | jplank | capture all port 5060 traffic and just watch the CLI until a host goes unreachable |
| 07:06.09 | jplank | 5060 traffic isn't like UDP, even with 1000 peers, it wont be too huge |
| 07:06.33 | OldMonk | this is tcp port 5060? |
| 07:06.43 | jplank | no UDP |
| 07:06.54 | jplank | I'm sorry, I meant isn't like RTP |
| 07:07.07 | OldMonk | cool |
| 07:07.27 | jplank | though, with that much data I recommend after making the capture file he copy locally to your machine and open it with the wireshark gui, will be a lot easier to parse with that many clients |
| 07:07.50 | OldMonk | will do that at end of working day |
| 07:08.01 | OldMonk | then try to correlate against the asterisk logs |
| 07:08.21 | jplank | there you go |
| 07:20.24 | *** join/#asterisk cerberus_za (~coert@dsl-185-106-48.dynamic.wa.co.za) |
| 07:31.49 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 07:39.48 | OldMonk | hmm, what would this mean? 0.000000 10.0.36.116 -> 10.0.10.130 SIP Request: OPTIONS sip:in_17626@10.0.10.130 |
| 07:39.54 | OldMonk | 0.002017 10.0.10.130 -> 10.0.36.116 SIP Status: 404 Not Found |
| 07:42.09 | jplank | hmm |
| 07:42.18 | jplank | what kind of endpoint is it? |
| 07:43.52 | *** join/#asterisk gmc (~gmc@freenode/sponsor/gmc) |
| 07:44.08 | OldMonk | qutecom |
| 07:44.50 | jplank | do you see ANY proper responses from that model phone to the options message? |
| 07:45.00 | gmc | hi all.. anyone care to share some insight onto why my dialplan results in Spawn extension (xs4all-prive, s, 3) exited non-zero on 'SIP/xs4all-00000003' after step the first Dial isn't answered?? http://pastebin.com/ED5fhDf3 |
| 07:45.22 | OldMonk | this is from phone to asterisk |
| 07:45.40 | OldMonk | asterisk is 10.0.10.130 |
| 07:45.44 | jplank | ye |
| 07:45.45 | jplank | a |
| 07:45.52 | jplank | does it ever properly respond? |
| 07:46.05 | OldMonk | to OPTIONS? sure, it sends 100 then 200 |
| 07:47.08 | jplank | it responds to options with a 100 trying? |
| 07:47.16 | OldMonk | yes |
| 07:47.36 | OldMonk | <PROTECTED> |
| 07:47.41 | OldMonk | <PROTECTED> |
| 07:47.41 | OldMonk | <PROTECTED> |
| 07:48.23 | jplank | I never seen a 100 trying as a response to a options request |
| 07:48.55 | jplank | I can't even imagine why it would send a 100 trying, but that can't be right |
| 07:49.12 | jplank | but I don't know if that follows the RFC or not |
| 07:49.55 | jplank | maybe there is a method inside options that the phone doesn't like? |
| 07:50.03 | ectospasm | no, 100 Trying and 200OK does not follow from an OPTIONS |
| 07:50.16 | ectospasm | they follow from an INVITE |
| 07:50.19 | ectospasm | check the SIP ID |
| 07:51.11 | jplank | does the issue ONLY happen with qutecom endpoints? |
| 07:55.51 | ectospasm | RFC 3261 specifies that (in this case) the endpoint should return either a 200 OK, 486 (Busy Here), etc. The CSeq. of the second message should be <some number> OPTIONS |
| 07:58.01 | OldMonk | we have only qutecoms here |
| 07:58.15 | OldMonk | that's a linux softphone, btw -- used to be wengophone |
| 07:58.20 | ectospasm | it may respond with other 4xx messages |
| 07:58.47 | jplank | would a 404 response be valid if the one of the methods wasn't supported? |
| 07:58.49 | ectospasm | I'd need to see a pcap |
| 07:59.26 | ectospasm | jplank: it could, yes. I can't remember when I saw this, but I think a 404 is a valid response to an OPTIONS method |
| 07:59.48 | OldMonk | ectospasm: for just one client or the whole load? |
| 08:01.09 | jplank | OldMonk: can you isolate the part of the capture showing 1) the 404 and 2) a 100 reply to a options? |
| 08:01.10 | gmc | hi all.. anyone care to share some insight onto why my dialplan results in Spawn extension (xs4all-prive, s, 3) exited non-zero on 'SIP/xs4all-00000003' after step the first Dial isn't answered?? http://pastebin.com/ED5fhDf3 |
| 08:01.23 | gmc | whoops.. sorry 'bout that |
| 08:01.33 | jplank | gmc: you don't have enough info in that pastebin |
| 08:01.41 | jplank | will need the cli output too |
| 08:01.59 | ectospasm | OldMonk: I would just need to see one dialog. In wireshark, choose an OPTIONS message, and filter only that dialog, then save the filter as its own pcap. |
| 08:02.10 | ectospasm | I dunno how to do it with tshark |
| 08:02.22 | jplank | ectospasm: are you asking to do that for the 404 or the 100? |
| 08:03.27 | ectospasm | jplank: basically, I'm asking for the entire OPTIONS dialog. Should be the OPTIONS message and its response. |
| 08:03.39 | ectospasm | ...I'm only expecting two packets there |
| 08:04.06 | jplank | I'd love to see the cseq on that options/100/200 dialog, I bet its not right |
| 08:07.24 | gmc | jplank: http://pastebin.com/jFVNSGKM |
| 08:07.30 | gmc | need sip debug as well? |
| 08:08.14 | OldMonk | ectospasm: say that again, please? |
| 08:09.11 | jplank | gmc: how much time is there between lines 14 and 15 |
| 08:09.26 | OldMonk | hmm, can't seem to get the sequence numbers |
| 08:09.30 | ectospasm | OldMonk: say what again? |
| 08:09.31 | OldMonk | in tshark |
| 08:09.36 | gmc | jplank: about 30 seconds |
| 08:09.42 | OldMonk | ectospasm: preciesly what do you want to see? |
| 08:09.49 | gmc | 25 i'd say |
| 08:09.59 | gmc | phones are ringing |
| 08:10.15 | ectospasm | OldMonk: I want to see a complete OPTIONS dialog. I want to see the OPTIONS message, and its response. |
| 08:10.44 | jplank | woonkamer and kantoor actually ring right? |
| 08:11.27 | gmc | yep, both phones are ringing |
| 08:12.15 | jplank | if you comment out line 6, does it go to VM properlY? |
| 08:13.20 | OldMonk | ectospasm: will pull it out and give it to you in a few minutes |
| 08:13.31 | gmc | lemme try |
| 08:14.21 | jplank | actually, aren't you not suppose to use the r option in dial if your using progress()? |
| 08:14.28 | jplank | isn't that what progress is doing |
| 08:14.55 | gmc | hrmz.. i tried with and without progress actually |
| 08:15.05 | jplank | leave progress, drop the r |
| 08:15.19 | gmc | progress was a recent addition after reading stuff on voip-info.org |
| 08:15.20 | gmc | ok |
| 08:15.29 | gmc | btw, doesn't go to voicemail, same non-zero thing |
| 08:16.49 | gmc | jplank: also, leaving out r and leave in progress.. same result.. |
| 08:17.02 | gmc | bummer, have an appointment to make.. |
| 08:17.04 | ectospasm | OldMonk: I dunno about tshark, but in wireshark you can select an OPTIONS request, and follow the UDP stream |
| 08:17.07 | gmc | tnx for your help so far! |
| 08:18.04 | jplank | something has to be wrong with the dial, I'm just not seeing it, play around with it, I'm sure its something stupid |
| 08:18.37 | jplank | try upping the ring time, make sure the call doesn't end sooner then the timer |
| 08:19.13 | gmc | ah.. hrmz.. when i do Answer() first it does work |
| 08:19.19 | gmc | except i don't get a ringtone anymore when calling in |
| 08:20.04 | jplank | will the phone generate the ring tone? |
| 08:20.14 | gmc | plus, of course, it then starts charging even if we don't answer |
| 08:20.18 | ectospasm | ringtone or ringback? |
| 08:20.18 | jplank | can you drop progress, and use R? |
| 08:20.23 | jplank | ringback |
| 08:20.25 | gmc | R capital? |
| 08:20.27 | jplank | I asusme |
| 08:20.44 | gmc | with the Answer you mean? |
| 08:20.46 | jplank | instead of generating the ringback, it tells the phone to do it |
| 08:20.49 | jplank | Dial |
| 08:20.57 | jplank | oh, no |
| 08:21.02 | jplank | thats from a patch isn't it? |
| 08:21.23 | jplank | ignore my R suggestion |
| 08:21.46 | gmc | ack.. |
| 08:21.49 | gmc | there's Ringing() though |
| 08:22.20 | jplank | try it |
| 08:23.01 | ectospasm | Ringing() is a hack |
| 08:23.08 | ectospasm | ...but if it works... |
| 08:24.05 | gmc | it doesn't though :) |
| 08:24.23 | jplank | your dropped the r right? |
| 08:24.33 | gmc | yep |
| 08:24.53 | gmc | right i now really have to go for that appointment :) |
| 08:24.55 | gmc | tnx again.. |
| 08:24.57 | jplank | you left the answer? |
| 08:25.41 | jplank | answer(), ringing(), dial(sip/1&sip/2), dial(sip/3&sip/4), Voicemail() |
| 08:25.43 | OldMonk | well fsck |
| 08:36.05 | OldMonk | ectospasm: here's one: does this help? http://pastebin.ca/2049656 |
| 08:37.20 | ectospasm | 404 Not Found is a valid response to OPTIONS |
| 08:37.44 | OldMonk | ectospasm: how about at the end: the 100/200 response? |
| 08:37.46 | ectospasm | Notice the Allow: line in the 404 message. |
| 08:38.23 | ectospasm | that's bizarre |
| 08:39.29 | ectospasm | 100 Trying is usually for INVITE, but in this case for the OPTIONS. Weird. I suppose it's not technically illegal, since Asterisk doesn't behave poorly to it. |
| 08:39.44 | ectospasm | OldMonk: why are you concerned about OPTIONS messages anyway? |
| 08:39.57 | ectospasm | They usually are safely ignored by the admin |
| 08:40.06 | OldMonk | i'm not, someone here pointed out that it could be a problem |
| 08:40.35 | OldMonk | ectospasm: also, you'd asked for a complete OPTIONS stream. is this enough? |
| 08:40.52 | ectospasm | yes. But I think I got caught in the middle of the conversation. |
| 08:41.20 | ectospasm | ...so I don't remember what the original problem was, and I'm too lazy to scroll up (tl;dr) |
| 08:41.49 | OldMonk | problem was clients getting mass connected, lagged, disconnected, connected again |
| 08:42.25 | ectospasm | OldMonk: do you have qualify set for all of them? |
| 08:42.31 | OldMonk | yes |
| 08:42.37 | ectospasm | what qualify value? |
| 08:42.44 | OldMonk | just "yes" |
| 08:42.52 | ectospasm | OK, try setting them to "no" |
| 08:43.18 | OldMonk | oh, ok |
| 08:43.26 | ectospasm | You won't get the nice "OK (XXXms)" message in "sip show peers", but Asterisk won't treat them as lagged |
| 08:43.36 | ectospasm | ...and then disconnect them after a certain period |
| 08:43.41 | ectospasm | "yes" means 2000ms |
| 08:43.43 | ectospasm | or 2sec |
| 08:43.55 | OldMonk | yup, that's what it's timing them out after now |
| 08:43.59 | ectospasm | with 4000 SIP endpoints |
| 08:44.11 | OldMonk | only 1000 or so are up, actually |
| 08:44.32 | OldMonk | nearer 900 to be more precise |
| 08:44.39 | ectospasm | yeah, that's still a lot. Asterisk takes a fair bit of massaging to support that many simultaneous registrations |
| 08:44.58 | OldMonk | would SER in the middle help? |
| 08:45.11 | ectospasm | maybe, depends on where the bottleneck is |
| 08:45.27 | OldMonk | i don't even know if it's my LAN acting up -- that's new too |
| 08:45.59 | ectospasm | yeah, there are many variables involved here, any of which could be the culprit (or collude with other problems to manifest this) |
| 08:47.05 | jplank | turning qualify off is just a bandaid though |
| 08:48.29 | jplank | wait, something is messed up |
| 08:49.13 | jplank | why is the phone sending asterisk a SIP options message? |
| 08:49.19 | jplank | thats what's being 404'd |
| 08:51.08 | jplank | of course asterisk is going to 404 those options requests |
| 08:54.03 | ectospasm | huh? Which is the Asteirsk IP, and which are the phones'? |
| 08:54.22 | OldMonk | asterisk is 10.0.10.130 |
| 08:54.43 | jplank | unless I'm reading it backwards, look at the first message |
| 08:54.52 | jplank | OPTIONS sip:in_14376@sip-concentrator-1.noida.foo.com SIP/2.0 |
| 08:54.57 | jplank | User-Agent: qutecom/rev-fdc30c5c1582-trunk |
| 08:55.10 | jplank | SIP/2.0Â 404Â Not Found |
| 08:55.15 | jplank | Server: Asterisk PBXÂ 1.6.2.9-2 |
| 08:55.50 | ectospasm | yeah |
| 08:55.56 | ectospasm | Asterisk is responding correctly |
| 08:56.07 | ectospasm | ...with the 404 |
| 08:56.17 | jplank | your softphone shouldn't be sending options to asterisk. |
| 08:56.28 | jplank | I don't know if thats actually causing the unreachable, but I guess its possible |
| 08:56.48 | jplank | OldMonk: my advise, find a new softphone |
| 08:56.59 | jplank | s/advise/advice/ |
| 08:58.19 | OldMonk | reinstallation on 1000+ debian desktops? ouch! |
| 08:59.18 | jplank | use ser to proxy the packets and make them play nice then |
| 08:59.19 | ectospasm | you don't have a way to manage all of these from a central location? Why did you install Debian on 1000 desktops if you didn't? |
| 08:59.42 | OldMonk | we do have ssh ;) |
| 08:59.55 | ectospasm | ...and scripting (-; |
| 09:00.11 | OldMonk | that too... that's how each qutecom gets configured remotely |
| 09:00.17 | jplank | would probably be easier to figure out the underlining problem |
| 09:00.36 | jplank | its probably a misconfig in qutecom |
| 09:01.21 | jplank | did you try #qutecom? |
| 09:01.35 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 09:02.01 | OldMonk | nope, i just found out that's the qutecom sending OPTIONS is a problem |
| 09:02.21 | ectospasm | sounds like you should be able to disable that in the config |
| 09:02.28 | jplank | heh, they actually have a rule in #qutecom that you must say hello when you join and goodbye before you leave |
| 09:03.21 | OldMonk | ...and do they respond with 404 when you say HELO? ;) |
| 09:03.53 | jplank | heh, you should go find out |
| 09:04.32 | jplank | maybe they'll respond with a 250 |
| 09:11.11 | OldMonk | also getting these fairly regularly: [Apr 23 14:40:51] WARNING[3661] chan_sip.c: Maximum retries exceeded on transmission 444800427@10.0.34.133 for seqno 21 (Critical Response) -- See doc/sip-retransmit.txt. |
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| 09:14.46 | jplank | would need to see the capture for that |
| 09:16.45 | OldMonk | making one, will pull it out in the evening |
| 09:17.01 | OldMonk | don't want to choke the eth with long transfers right now |
| 09:19.55 | OldMonk | oh excellent, tshark is logging whole decoded sip packets now |
| 09:21.18 | OldMonk | bbl |
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| 12:04.00 | fatalfury1981 | Good Morning at all. I've big problem with hangup detection with analog line, dahdi with digium TDM400P. I live in italy, i've tried set busydetect.. busycount polarity reverseal.. loadzone default zone are set up with italy.. Is there any way with asterisk cli works in debug mode for understand what happens in asterisk when the caller hungup? |
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| 13:17.42 | gmc | re |
| 13:17.46 | gmc | jplank: yep, the answer was still there |
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| 14:10.36 | gmc | jplank: for the record, problem wasn't with asterisk |
| 14:10.48 | gmc | turned out the provider had voicemail enabled, and thus stopped trying after 30 seconds...... |
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| 14:45.41 | JunTao2 | would I be able to connect an FXO ATA device to any of the FXS phone plugs in my house, which are currently plugged in with phones? That way I can pick up the phone downstairs and also the signal still would go to upstairs to the ATA device which connects to asterisk? |
| 14:59.20 | wolfe | question |
| 14:59.32 | wolfe | does Asterisk have any type of transrating services? |
| 14:59.54 | wolfe | I've used asterisk since about 05' and casually have issues on networks which are limited in BW |
| 15:00.13 | wolfe | can asterisk do any transrating or no? |
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| 15:18.30 | Freeaqingme | ~book |
| 15:18.31 | infobot | For more information about the Asterisk book, see ~thebook |
| 15:18.34 | Freeaqingme | ~thebook |
| 15:18.34 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
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| 15:31.15 | gruvfunk | what's the easiest method to generate a UUID/GUID for an IVR session? |
| 15:33.11 | gruvfunk | I found this, but it requires launching a separate process and keeping it running in background: http://lists.digium.com/pipermail/asterisk-users/2007-August/193880.html |
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| 15:51.03 | drfreeze | Anyone know how to adjust sidetone on polycom 550 phones? |
| 15:51.37 | drfreeze | In the sip file there is control for 330,430,450,650, but no 550 |
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| 16:10.39 | cusco_ | when I place a .call file it goes trough a channel, then on connect it goes to the specified context and extension... |
| 16:11.05 | cusco_ | if chanel is local and I Set(var) can I somehow re-use that var on the specified context extension ? |
| 16:18.34 | gruvfunk | @cusco_ I believe a Channel Variable will persist through the life of that channel (call) |
| 16:18.59 | gruvfunk | Otherwise, look at Global Variables (SetGlobalVar) |
| 16:21.12 | gruvfunk | er.. deprecated, use Set(GLOBAL= |
| 16:21.46 | gruvfunk | Can anyone in here confirm that a Channel Variable persists through contexts? |
| 16:22.16 | cusco_ | gruvfunk: im not sure but it seems that Extension: bla will not belong the the same channel ??? |
| 16:22.40 | cusco_ | or I don't understand |
| 16:23.44 | cusco_ | gruvfunk: take a look here: http://lists.digium.com/pipermail/asterisk-users/2011-April/261750.html |
| 16:24.55 | gruvfunk | I believe what you want to do is set a global variable then, in the [globals] section |
| 16:25.21 | cusco_ | with several calls that var will be overwritten... |
| 16:25.31 | cusco_ | right? |
| 16:25.43 | gruvfunk | I suppose |
| 16:26.12 | gruvfunk | My understanding is a channel stays the same through the life of the call |
| 16:26.19 | cusco_ | yes in mine too |
| 16:26.32 | gruvfunk | so a channel variable should work? |
| 16:26.54 | gruvfunk | perhaps if you expand on what you are trying to achieve? |
| 16:26.59 | cusco_ | but read my example call file in that mailing list url... |
| 16:27.19 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 16:27.33 | cusco_ | ok, so I place a .call file to a channel that has Queue() |
| 16:27.58 | cusco_ | once some one answers that queue, then is pointed to the Context and Extension specified on that .call file |
| 16:29.03 | gruvfunk | so when someone answers the queue, the call just gets passed to another context/extension? |
| 16:29.26 | cusco_ | when that happens the original channel hungs up, and that someone follows trough that new dialplan |
| 16:29.56 | cusco_ | yes |
| 16:30.10 | cusco_ | and in this new context/extension I cannot get the peer number |
| 16:30.15 | cusco_ | of who answered the call |
| 16:30.39 | gruvfunk | hmm.. complex, hope somebody else in here can help |
| 16:31.14 | cusco_ | I thought I should be able to use MEMBERINTERFACE |
| 16:31.25 | cusco_ | ok thansk for trying anyway |
| 16:31.38 | cusco_ | gruvfunk: regarding your question I really dunno what uuid and guid are |
| 16:31.49 | cusco_ | I use ${UNIQUEID} as unique identifier |
| 16:32.10 | gruvfunk | sure, is there a better solution to your problem? (I find myself asking that question a lot - is there an easier way to achieve what you want, instead of .call -> Queue -> context |
| 16:33.34 | cusco_ | well... .call files would be good enough here.. |
| 16:33.54 | cusco_ | We have a bd with list of phone numbers to outbound |
| 16:33.58 | gruvfunk | right, but when somebody answers the Queue, why is that not the "Agent" |
| 16:34.03 | gruvfunk | why pass it off? |
| 16:34.15 | cusco_ | hu? |
| 16:34.33 | cusco_ | the queue member answers the queue.. |
| 16:34.33 | gruvfunk | when the Queue is answered, is it human? |
| 16:34.40 | cusco_ | yes |
| 16:34.50 | cusco_ | then he automatically gets dialed to outbound |
| 16:35.09 | gruvfunk | so it's a call back? |
| 16:35.16 | cusco_ | sort of yes |
| 16:35.42 | cusco_ | so... outbound contact -> call file -> queue -> answer -> dialout |
| 16:37.18 | gruvfunk | as I try to find a better way - forums suggest a script for call back |
| 16:37.21 | cusco_ | so I would like to set a var on dialplan that Channel points to, and re-use it in Extension |
| 16:37.38 | cusco_ | script.. as in agi? |
| 16:37.49 | gruvfunk | that's what I'm reading, agi, php, perl |
| 16:38.01 | cusco_ | well... |
| 16:38.05 | gruvfunk | but... why don't you write your Channel variable to AstDB, and then call it back out? |
| 16:38.08 | cusco_ | gah |
| 16:38.35 | cusco_ | never used astdb manually |
| 16:38.55 | gruvfunk | in the dialplan |
| 16:38.58 | cusco_ | wouldn't it get overwritten too ? |
| 16:39.23 | cusco_ | if more than one call at the same time exists? |
| 16:39.35 | gruvfunk | depends, need some logic |
| 16:40.03 | gruvfunk | with AstDB you have groupings called "families" with valies identified by "keys" |
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| 16:40.49 | cusco_ | it seems a new world to explore |
| 16:40.57 | cusco_ | Ill just look it up on voip info |
| 16:41.51 | gruvfunk | maybe something like Set(DB(customers/${UNIQUEID}=${CALLERID(num)}) - I have no idea if that works |
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| 16:42.03 | cusco_ | uniqueid will change too |
| 16:42.05 | cusco_ | lol |
| 16:42.47 | gruvfunk | tough weekend, everybody seems unavailable for Easter :) |
| 16:43.03 | cusco_ | true _= |
| 16:43.04 | cusco_ | :) |
| 16:43.58 | cusco_ | Im thinking in setting a global var using destination nr-var and then cut it back... |
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| 16:44.59 | cusco_ | but this doesn't make sese |
| 16:45.04 | cusco_ | channel is the same |
| 16:45.17 | gruvfunk | a UUID/GUID is a 128 bit long number that can be used to identify something, like UNIQUEID, except longer 00000000-0000-0000-0000-000000000000 |
| 16:45.33 | cusco_ | ow.. |
| 16:45.40 | gruvfunk | more random |
| 16:45.48 | cusco_ | isn't there a function to retrieve/generate one in dialplan? |
| 16:46.03 | gruvfunk | we're using UUID's to track session #, user ID, etc |
| 16:46.12 | gruvfunk | is there? that's what I'm looking for! |
| 16:46.54 | cusco_ | well, how do you know uuid exists? |
| 16:47.37 | gruvfunk | it's a public standard |
| 16:47.43 | cusco_ | ah.... |
| 16:48.01 | gruvfunk | so the only way I found to generate one is via AGI perl script |
| 16:48.09 | cusco_ | can't you just manually generate one? |
| 16:48.15 | gruvfunk | but it requires me to run that perl script in the background, creats zombie processes sometimes hangs |
| 16:48.31 | gruvfunk | probably - my perlFu is not good |
| 16:48.34 | cusco_ | like ${UNIQUEID}-${EPOCH}-${CALLERID(num)} |
| 16:48.40 | cusco_ | in dialplan, Imean |
| 16:48.48 | gruvfunk | it has to be UUID conformant |
| 16:48.56 | cusco_ | ah right.. |
| 16:49.01 | gruvfunk | 128 bits, 32 characters, 8-4-4-4-12 |
| 16:49.51 | gruvfunk | shame.. my googleFu is good, appears Freeswitch uses UUID's natively |
| 16:50.06 | cusco_ | o.O |
| 16:50.09 | gruvfunk | it may be time to start learning something new |
| 16:50.11 | cusco_ | hold a sec |
| 16:51.54 | cusco_ | is it only digits? |
| 16:51.57 | cusco_ | or alphanumeric? |
| 16:53.22 | cusco_ | alphanumeric according to wikipedia |
| 16:53.33 | cusco_ | err.. hex |
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| 16:55.03 | cusco_ | so what about using perl or php agi ? |
| 16:55.25 | x86 | mmmmmm perl AGI |
| 16:55.29 | cusco_ | in the past I used something like... |
| 16:56.41 | cusco_ | System(php -q /some/path/echo.php ${CHANNEL}); -- where this echo.php takes the cannel argument, opens socket to manager, and sets a channel variable |
| 16:56.53 | cusco_ | probably a long way but seems that php can generate uuid |
| 16:57.29 | cusco_ | x86: are you familliar with .call files ? |
| 16:57.30 | cusco_ | :p |
| 16:57.33 | cusco_ | needs help |
| 17:03.59 | cusco_ | gruvfunk: |
| 17:04.52 | cusco_ | using dumpchan I see that Channel (where the call goes first) has channel name: Local/210332450@ZonNew-Outbound-66c7;2 ... and Context/Extension (where the call goes after) has channel name Local/210332450@ZonNew-Outbound-66c7;1 |
| 17:05.07 | cusco_ | so it seems that the first is a subchannel of the second ? |
| 17:05.09 | cusco_ | o.O |
| 17:07.37 | *** join/#asterisk ph8 (ph8@unaffiliated/ph8) |
| 17:07.54 | cusco_ | I do not understand... |
| 17:11.48 | drfreeze | Man, it's hard to tell the difference between sidetone and echo |
| 17:13.18 | Kobaz | so, this was fun |
| 17:13.32 | Kobaz | i had a deadlock caused this morning by core restart when convenient |
| 17:18.06 | cusco_ | I had one like that too last weak.. |
| 17:18.14 | cusco_ | I had a core restart gracefully |
| 17:18.25 | cusco_ | but there was a call in queue during night that nobody picked up... |
| 17:18.36 | cusco_ | so it never restarted and refused new calls |
| 17:18.37 | cusco_ | lol |
| 17:19.33 | cusco_ | Kobaz: any good with call files? |
| 17:20.56 | Kobaz | never used them |
| 17:21.04 | Kobaz | well |
| 17:21.10 | Kobaz | you don't want to use core restart gracefull |
| 17:21.27 | Kobaz | because it will block new calls until all the calls finish |
| 17:21.40 | Kobaz | core restart when convenient is the way to go... but i gotta try and reproduce this deadlock |
| 17:22.20 | Kobaz | call files are basically a file with call originate options |
| 17:22.40 | *** join/#asterisk sourcode (~code@ppp-58-8-150-241.revip2.asianet.co.th) |
| 17:25.26 | *** join/#asterisk sizzers (~k_killah@c-68-41-174-94.hsd1.mi.comcast.net) |
| 17:26.10 | cusco_ | Kobaz: yes but seems that I cannot re-use vars in them |
| 17:26.31 | cusco_ | it originates a channel and then upon connect points it to a context/extension |
| 17:26.51 | cusco_ | http://lists.digium.com/pipermail/asterisk-users/2011-April/261751.html |
| 17:30.48 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 17:31.04 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 17:31.08 | Kobaz | when you use local channels, you will get the ;1 leg and the ;2 leg |
| 17:31.18 | Kobaz | they are connected to each other, they are not sub channels |
| 17:31.36 | *** part/#asterisk sizzers (~k_killah@c-68-41-174-94.hsd1.mi.comcast.net) |
| 17:32.10 | Kobaz | looks like you're trying to communicate between channels |
| 17:32.19 | Kobaz | there are several ways of doing that |
| 17:32.41 | Kobaz | you can used the SHARED function |
| 17:33.06 | Kobaz | this has nothing really do to with call files, it's just how local channels work |
| 17:34.14 | Kobaz | oh also. you need to put all your sets on one line seperated by commas, that's another problem |
| 17:35.02 | cusco_ | ow, ok |
| 17:35.16 | cusco_ | so I need to use SHARED.. |
| 17:35.41 | Kobaz | if you want to set a var on ;1 and be able to have ;2 use it (or vice versa) then yeah |
| 17:36.13 | cusco_ | :( ok |
| 17:36.20 | cusco_ | I will try that |
| 17:36.20 | Kobaz | Set: __PARTNER=ZonNew-Outbound,NUMBER=210332450 |
| 17:36.24 | Kobaz | so do that instead |
| 17:36.26 | cusco_ | yes, thanks for that |
| 17:36.30 | Kobaz | and you'll get your initial vars |
| 17:36.34 | cusco_ | didn't get there yet :p |
| 17:37.25 | Kobaz | you'll have to know the channel name in order to use shared |
| 17:38.24 | cusco_ | yes that was my question now |
| 17:38.30 | cusco_ | I was just reading that... |
| 17:39.35 | cusco_ | I can use prefix |
| 17:39.39 | cusco_ | should be OK |
| 17:40.00 | cusco_ | since it looks like current channel name.. |
| 17:40.01 | cusco_ | right? |
| 17:45.57 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 18:00.08 | cusco_ | argh... |
| 18:02.53 | *** join/#asterisk sourcode (~code@ppp-58-8-150-241.revip2.asianet.co.th) |
| 18:05.25 | Kobaz | mm, you might be able to use a prefix, some stuff you can |
| 18:05.35 | Kobaz | i dont remmeber offhand |
| 18:08.17 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:21f:5bff:fe37:c2c9) |
| 18:08.44 | cusco_ | voip info states that I can |
| 18:15.34 | Kobaz | for knowing exactly you can look at the source code |
| 18:17.49 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:19.02 | cusco_ | [Apr 23 19:18:36] ERROR[17291]: func_global.c:149 shared_read: Channel 'Local/210332450@ZonNew-Outbound-7488' not found! Variable 'foo' will be blank. |
| 18:19.27 | cusco_ | and before that: -- Executing [210332450@ZonNew-Outbound:8] Set("Local/210332450@ZonNew-Outbound-7488;2", "SHARED(foo)=bar") in new stack |
| 18:20.01 | cusco_ | :( |
| 18:21.14 | cusco_ | if I specify the channel name exactly, it works |
| 18:21.16 | cusco_ | gah... |
| 18:21.49 | cusco_ | Im using ${SHARED(foo,${CHANNEL:0:$[${LEN(${CHANNEL})} - 1]}2)}); |
| 18:22.03 | cusco_ | but this is assuming I want the same channel name leg 2 |
| 18:22.07 | cusco_ | ... |
| 18:22.21 | cusco_ | isn't there another way? |
| 18:22.43 | cusco_ | to share a variable? |
| 18:27.16 | Kobaz | my group variables patch |
| 18:27.24 | Kobaz | or global variables (which are evil) |
| 18:27.56 | Kobaz | anyways |
| 18:28.02 | Kobaz | before we go deeper into the rabbit hole |
| 18:28.08 | Kobaz | what exactly are you trying to do |
| 18:30.35 | cusco_ | lol |
| 18:30.55 | cusco_ | I tried globals.. but seems that I cannot unset them |
| 18:30.58 | cusco_ | thats really bad |
| 18:31.38 | cusco_ | well... call file that will place a call on queue, when a member answers he gets to dialout to client |
| 18:36.58 | Kobaz | you can clear them |
| 18:37.13 | Kobaz | there's nothing to delete variables.. but you can Set(var="") |
| 18:37.38 | Kobaz | the problem with globals also, is that you need a scheme to make sure each channel has it's own |
| 18:37.49 | Kobaz | which is my my group variable stuff is really cool |
| 18:38.54 | cusco_ | :| |
| 18:39.11 | cusco_ | well for now, all I want is to know the peer number of who answered the queue |
| 18:40.03 | cusco_ | so... outbound contact -> call file -> queue -> answer -> dialout |
| 18:40.23 | cusco_ | I did that setting them to "" |
| 18:40.49 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 18:40.57 | cusco_ | but then: dialplan show globals, I have loads of 1303581808.4556-ext= |
| 18:41.19 | JunTao2 | can asterisk do voicemail and other pbx automated call agents when I have a set up at home like POTS LAndline->ATA FXO Device->PBX ? |
| 18:41.40 | cusco_ | JunTao2: I don't see why not |
| 18:43.22 | cusco_ | Kobaz: what does your group variables do?» |
| 18:43.52 | Kobaz | you know how yo ucan put a channel into a group |
| 18:43.59 | Kobaz | and the groups go away when the channels go away |
| 18:44.11 | cusco_ | never used that.. |
| 18:44.13 | JunTao2 | cusco_ so this would work http://www.telephonydepot.com/Grandstream-HandyTone-503-HT503 ? |
| 18:44.18 | Kobaz | well i added variable support, for groups... so you can attach a variable to a group.. and you can access the variables by knowing the group |
| 18:44.26 | Kobaz | and when the channels go away, the variables go away... no cleanup |
| 18:44.45 | cusco_ | nice |
| 18:44.54 | cusco_ | that would work |
| 18:44.57 | Kobaz | yeap |
| 18:44.58 | cusco_ | why isn't that in mainstream? |
| 18:45.16 | Kobaz | because i still have some work to do on it |
| 18:45.28 | Kobaz | and it won't be in mainline until 1.10 anyway |
| 18:45.53 | Kobaz | https://reviewboard.asterisk.org/r/464/ |
| 18:46.16 | cusco_ | JunTao2: sorry dunno about hardware... but if you can connect to it, then you can do whatever from dialplan |
| 18:46.59 | cusco_ | Kobaz: well I would really rather not apply patches... |
| 18:47.20 | *** join/#asterisk hamidqa (~hamid@85.133.201.3) |
| 18:47.24 | hamidqa | hi |
| 18:47.32 | hamidqa | i have quasdtion about astrik |
| 18:47.32 | cusco_ | for now I will use IMPORT and relly on channelname;2 instead of ;1 |
| 18:47.49 | Kobaz | cusco_: if you work with asterisk long enough, all you will have are patches |
| 18:47.53 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 18:48.14 | Kobaz | cusco_: i have about 2000 lines of changes to my current asterisk for bug and crash fixes, and new features i wrote |
| 18:48.36 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 18:48.41 | cusco_ | Kobaz: well yes we used to alter a little source code in past times, but now we can always get to do whatever trough dialplan |
| 18:49.17 | hamidqa | i have a two wifi network with two deferent dhcp address when sip mobile user switch betwen those network the voice connection is droped by my old sip server is astrik support this (do not drop call on wifi switch |
| 18:50.57 | *** join/#asterisk geek (~killown@unaffiliated/geek) |
| 18:51.28 | hamidqa | ? |
| 18:53.23 | hamidqa | any one? |
| 18:54.30 | hamidqa | hello |
| 18:54.41 | hamidqa | any help |
| 18:55.48 | cusco_ | I don't know.. |
| 18:56.25 | *** join/#asterisk JonnyD_work (~Jon@173.226.80.154) |
| 18:58.02 | hamidqa | finally thank for answer cusco_ |
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| 19:06.38 | *** part/#asterisk killown (~killown@unaffiliated/killown) |
| 19:30.03 | *** join/#asterisk ghostmediapro (~shawn@71.23.182.164) |
| 19:31.01 | ghostmediapro | Qwell: http://pastebin.com/TerfgAiu |
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| 19:41.56 | cusco_ | retrieve_conf ? |
| 19:47.17 | ghostmediapro | yes |
| 19:49.41 | cusco_ | what is that? |
| 19:53.21 | ghostmediapro | where freepbx loads your config file and sync with asterisk |
| 20:06.29 | *** join/#asterisk tomaw_ (tom@freenode/staff/tomaw) |
| 20:10.03 | *** join/#asterisk Mo (5e652b65@gateway/web/freenode/ip.94.101.43.101) |
| 20:11.32 | *** join/#asterisk movozo (5e652b65@gateway/web/freenode/ip.94.101.43.101) |
| 20:12.09 | movozo | Hey Guys. Really need your help on this one. Donation should be possible for a Solution... |
| 20:13.11 | movozo | Using Asterisk as a STUN and Registar for voip. But we have a really bad lag Problem when pickung up the Phone. About 2-5seconds of lag, after that everything is fine |
| 20:13.46 | movozo | sorry, when calling somebody |
| 20:14.06 | movozo | you will not hear his or her name, even they take time to say it. |
| 20:20.11 | movozo | nobody on? bad time? |
| 20:21.20 | gruvfunk | exit |
| 20:21.21 | gruvfunk | quit |
| 20:22.13 | cusco_ | hi |
| 20:27.46 | movozo | hi |
| 20:28.51 | *** join/#asterisk geek (~killown@unaffiliated/geek) |
| 20:30.56 | *** join/#asterisk geek (~killown@unaffiliated/geek) |
| 20:38.39 | cusco_ | sorry movozo can't help you |
| 20:38.50 | cusco_ | I never used asterisk as STUN registrar |
| 20:38.54 | cusco_ | how does that work... ? |
| 20:39.04 | cusco_ | only outbounds go trough asterisk? |
| 20:40.21 | movozo | ope |
| 20:40.22 | movozo | nope |
| 20:40.25 | movozo | inbounds, too |
| 21:23.54 | *** join/#asterisk movozo (~Mo@94.101.43.101) |
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| 22:15.09 | *** join/#asterisk tacloban (~bgiles@c-24-18-11-138.hsd1.wa.comcast.net) |
| 22:16.36 | tacloban | can anyone confirm that there are known intermittent issues with asterisk on inbound google voice calls? |
| 22:16.58 | tacloban | i've seen some blogs on it, but no solutions yet, afaik |
| 22:17.29 | tacloban | i'm using asterisk 1.8.3.2, btw |
| 22:22.17 | tacloban | i also had a working configuration as of about 3 weeks ago, and I did not change anuything on my config |
| 22:22.27 | tacloban | but things are very flaky now |
| 22:24.12 | pabelanger | tacloban: It is a free service, so plan around in |
| 22:31.51 | monzsca | tacloban: yes, i'm having inbound google voice problems too |
| 22:32.14 | monzsca | phones ring, but don't seem to answer |
| 22:35.07 | *** join/#asterisk micols (~ident@rlogin.dk) |
| 22:45.08 | Freeaqingme | monzsca, you need to send a dtmf digit 1 before it's answered |
| 22:45.39 | monzsca | yes |
| 22:46.03 | monzsca | but the big problem is often the phones ring but i get no press 1 prompt and the caller continues to ring until voicemail |
| 22:46.21 | monzsca | about all i've found is this http://michigantelephone.wordpress.com/2011/03/22/google-voice-incoming-calls-broken-with-asterisk-1-8/ |
| 22:48.02 | monzsca | works sometimes, sometimes not |
| 23:00.54 | tacloban | same problem here |
| 23:01.05 | tacloban | inbound works sometimes |
| 23:01.18 | tacloban | when it does not work, it goes to the google voice mail |
| 23:01.33 | tacloban | used to work like a champ |
| 23:02.00 | tacloban | i noticed 1.8.3.3 is out now, so I am building that now |
| 23:02.12 | tacloban | i'll see if it makes a difference |
| 23:02.15 | monzsca | yes, the changelog doesn't say anything about gv |
| 23:02.19 | tacloban | true |
| 23:02.32 | tacloban | but, i am trying anything at this point |
| 23:03.07 | tacloban | like monzsca, I might just have to break down and get a real paid service |
| 23:03.09 | monzsca | yea, been driving me nuts for few weeks now |
| 23:03.23 | tacloban | i mean pabelanger |
| 23:03.48 | monzsca | i did just find out about the obihai device that apparently still works fine with gv |
| 23:04.00 | tacloban | ya, that blog page mentioned that |
| 23:04.09 | monzsca | that's all i really needed but didn't know about it |
| 23:04.12 | tacloban | but, i already spent $50 bucks on a cisco ATA |
| 23:04.36 | tacloban | not gonna spend another $50 on a device that i have laess control over and may stop working if google keeps messing around |
| 23:04.41 | monzsca | yea i already bought phones and have a ata from using gizmo5 |
| 23:05.01 | monzsca | hope they fix it soon |
| 23:05.02 | tacloban | at least with a generic ATA, i can use it other places too |
| 23:05.07 | monzsca | yes |
| 23:05.10 | tacloban | its been three weeks, |
| 23:05.25 | monzsca | i know, thought at first it had something to do with adding a second gv account |
| 23:05.29 | monzsca | friend of mine has the same trouble |
| 23:05.49 | tacloban | the most frustrating thing is that it does not seem like many people on the asterisk forums are even talking about it |
| 23:06.31 | monzsca | i just found that blog post a few days ago after much searching for weeks |
| 23:06.36 | tacloban | i wish we knew if it was an issue with google or a bug in asterisk |
| 23:07.01 | tacloban | i think someone on the blog commented that freeswitch still works too |
| 23:07.14 | monzsca | yes, so i suspect asterisk problem |
| 23:07.52 | tacloban | i mean, its all free, but i'd be nice if there were some feedback from the devs that they know about it |
| 23:08.08 | tacloban | maybe some idea of a timeline when that think they will get to it |
| 23:08.09 | monzsca | yep |
| 23:08.21 | monzsca | sounds like they know |
| 23:08.29 | tacloban | have you seen something? |
| 23:08.45 | tacloban | like a comment or message in the fourm? |
| 23:09.04 | monzsca | i saw something somewhere that they acknowledged the issue |
| 23:09.32 | monzsca | a comment on that blog post says they're trying to figure it out |
| 23:09.39 | monzsca | thats a month old though |
| 23:09.51 | tacloban | thats good to know |
| 23:10.15 | tacloban | its hard to complain though, its free and opensource |
| 23:10.21 | monzsca | yes |
| 23:10.23 | tacloban | i just dont have an c/C++ skills |
| 23:10.38 | monzsca | thought quite a few people used gv with asterisk though, would expect more info/response |
| 23:10.44 | tacloban | exactly |
| 23:10.53 | tacloban | i thought it was really popular |
| 23:11.34 | monzsca | it still rings my cellphone so i can still get the calls but it's annoying |
| 23:13.38 | tacloban | well, the 1.8.3.3 build is done |
| 23:13.44 | tacloban | gonna restart it |
| 23:15.28 | tacloban | meh, same issue |
| 23:15.59 | monzsca | shame |
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| 23:44.00 | therawr | lala |
| 23:44.29 | Freeaqingme | lili |
| 23:44.54 | Freeaqingme | I'm not sure though I'm willing to sing a song in here |
| 23:45.04 | Freeaqingme | (and I'm fairly sure you guys wouldnt want me either) |