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| 05:11.12 | bhavikpatel6842 | Hi All |
| 05:11.27 | bhavikpatel6842 | I am facing one issue regarding webrtc using ASterisk 11.9.0 |
| 05:11.39 | bhavikpatel6842 | I am trying to integrate SIPml5 demo. |
| 05:11.52 | bhavikpatel6842 | I am not able to sip to sip calls using two chrome browser which is latest version |
| 05:12.19 | bhavikpatel6842 | I refer this link for webRTC : https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 |
| 05:12.55 | bhavikpatel6842 | I made some changes which is define in above link like below : |
| 05:13.05 | bhavikpatel6842 | http://pastebin.com/7KCvtcNf |
| 05:13.55 | bhavikpatel6842 | Sip logs for |
| 05:13.55 | bhavikpatel6842 | Incoming calls : http://pastebin.com/ykeS0JCi |
| 05:13.55 | bhavikpatel6842 | and outbound call log : |
| 05:13.55 | bhavikpatel6842 | http://pastebin.com/e8Ap3bhq |
| 05:14.08 | bhavikpatel6842 | Can any one have idea why calls not working ? |
| 05:20.27 | ChannelZ | Define "not working".. I see in your sip debug that it gets the INVITE, sends one to the other side which reports Ringing but then your debug ends. |
| 05:21.06 | bhavikpatel6842 | but then after not getting any request for any side |
| 05:23.22 | ChannelZ | well I don't know anything about or have any experience with webrtc or that client |
| 05:25.06 | bhavikpatel6842 | ok np but when I dial just for sip to sip call my asterisk CLI just like |
| 05:25.07 | bhavikpatel6842 | <PROTECTED> |
| 05:25.07 | bhavikpatel6842 | <PROTECTED> |
| 05:25.16 | bhavikpatel6842 | and then nothing showing in CLI |
| 05:25.35 | bhavikpatel6842 | here is log for sip when I just enable it. |
| 05:25.35 | bhavikpatel6842 | http://pastebin.com/H2tFmXTF |
| 05:25.43 | bhavikpatel6842 | if you found any issue ? |
| 05:29.25 | ChannelZ | Do you have some console verbose turned on? It's probably telling you it's an invalid extension |
| 05:30.29 | bhavikpatel6842 | Means ? |
| 05:30.54 | bhavikpatel6842 | I think looks like audio issue ? |
| 05:31.44 | ChannelZ | You're trying to call extension 8001 but that extension probably isn't in your dialplan... SIP/2.0 404 Not Found |
| 05:32.14 | ChannelZ | At least that's the main reason I know of for a 404 |
| 05:32.53 | bhavikpatel6842 | 8001 is register extension |
| 05:33.06 | ChannelZ | A device and an extension are two different things. |
| 05:33.55 | bhavikpatel6842 | I am trying to call 8002 to 8001 |
| 05:34.27 | ChannelZ | Although it seems you are using FreePBX which calls them the same thing, so I don't know. I'm not seeing any of your console output so I can't speculate |
| 05:36.43 | bhavikpatel6842 | okey |
| 05:36.50 | bhavikpatel6842 | Please check new logs : http://pastebin.com/Yzx3iiM1 |
| 05:37.00 | bhavikpatel6842 | if you can able to find issue . |
| 05:37.31 | ChannelZ | Doesn't look much different |
| 05:37.41 | ChannelZ | What did you do? |
| 05:38.05 | bhavikpatel6842 | nothing just re -register both extension and tried again. |
| 05:38.28 | ChannelZ | core set verbose 3 |
| 05:39.53 | bhavikpatel6842 | look this lines |
| 05:39.54 | bhavikpatel6842 | Found unknown media description format opus for ID 111 |
| 05:39.54 | bhavikpatel6842 | Found unknown media description format ISAC for ID 103 |
| 05:39.54 | bhavikpatel6842 | Found unknown media description format ISAC for ID 104 |
| 05:39.54 | bhavikpatel6842 | Found audio description format PCMU for ID 0 |
| 05:39.54 | bhavikpatel6842 | Found audio description format PCMA for ID 8 |
| 05:39.54 | bhavikpatel6842 | Found unknown media description format CN for ID 106 |
| 05:39.55 | bhavikpatel6842 | Found unknown media description format CN for ID 105 |
| 05:39.55 | bhavikpatel6842 | Found audio description format CN for ID 13 |
| 05:39.56 | bhavikpatel6842 | Found audio description format telephone-event for ID 126 |
| 05:39.56 | bhavikpatel6842 | Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) |
| 05:39.57 | bhavikpatel6842 | Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|) |
| 05:39.58 | bhavikpatel6842 | Peer audio RTP is at port 192.168.1.13:28230 |
| 05:39.59 | bhavikpatel6842 | <PROTECTED> |
| 05:40.17 | bhavikpatel6842 | do you this line cause some issues ? |
| 05:40.31 | ChannelZ | please don't flood |
| 05:40.57 | bhavikpatel6842 | Oh sry |
| 05:40.58 | ChannelZ | And no that just means it found a bunch of codecs it doesn't care about. In the end it decides on ulaw/alaw. |
| 05:41.41 | ChannelZ | Turn on verbose as I said above and see what it's DOING. |
| 05:41.42 | bhavikpatel6842 | but not getting why its sending 404 |
| 05:43.00 | bhavikpatel6842 | Please check this one : http://pastebin.com/CGMXTU4f |
| 05:44.15 | ChannelZ | That still doesn't look any different. Are you getting this from a logfile or from the console? |
| 05:45.34 | bhavikpatel6842 | from the consol |
| 05:45.43 | bhavikpatel6842 | just set verbose to 3 |
| 05:58.42 | bhavikpatel6842 | Now I am getting Got SIP response 603 "Decline" back from |
| 05:58.51 | bhavikpatel6842 | SIP/8002-00000014 is busy |
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| 06:13.31 | [TK]D-Fender | no 603 in that pastebin |
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| 06:14.26 | bhavikpatel6842 | <PROTECTED> |
| 06:14.28 | bhavikpatel6842 | <PROTECTED> |
| 06:14.28 | bhavikpatel6842 | <PROTECTED> |
| 06:14.28 | bhavikpatel6842 | <PROTECTED> |
| 06:14.28 | bhavikpatel6842 | <PROTECTED> |
| 06:14.28 | bhavikpatel6842 | <PROTECTED> |
| 06:14.28 | bhavikpatel6842 | <PROTECTED> |
| 06:14.29 | bhavikpatel6842 | <PROTECTED> |
| 06:14.29 | bhavikpatel6842 | SIP/8002-00000018 is busy |
| 06:14.31 | bhavikpatel6842 | like this |
| 06:14.41 | [TK]D-Fender | bhavikpatel6842: PASTEBIN |
| 06:14.48 | [TK]D-Fender | STOP FLOODING THE CHANNEL |
| 06:14.55 | bhavikpatel6842 | oh let me give you. |
| 06:15.06 | jblack | bhavikpatel6842: THey want you to paste your logs to pastebin.com instead of here. |
| 06:15.22 | [TK]D-Fender | [02:14]bhavikpatel6842-- Executing [8001@webrtc:1] Dial("SIP/8002-00000017", "SIP/8002") in new stack <- and you're calling YOURSELF |
| 06:15.32 | [TK]D-Fender | 8002 doesn't want to talk to ITSELF |
| 06:15.47 | bhavikpatel6842 | http://pastebin.com/MrJLkASC |
| 06:16.21 | bhavikpatel6842 | I am trying to call 8002 to 8001 |
| 06:16.31 | [TK]D-Fender | that is NOT what that dialplan is doing |
| 06:16.40 | [TK]D-Fender | Dial("SIP/8002-00000017", "SIP/8002") |
| 06:17.00 | [TK]D-Fender | Call FROM SIP/8002 ..... calling SIP/8002 |
| 06:18.43 | bhavikpatel6842 | Oh yes you are right |
| 06:19.09 | bhavikpatel6842 | but after this audio not passing |
| 06:19.25 | bhavikpatel6842 | I set rtp set debug on and showing no logs in CLI |
| 06:19.41 | [TK]D-Fender | Stop trying to make your devices talk to themselves |
| 06:20.01 | [TK]D-Fender | And I see no answer so why should there be RTP? |
| 06:21.27 | bhavikpatel6842 | no now I am able to answer the call and core show channels also seeing connected lines |
| 06:22.34 | bhavikpatel6842 | http://pastebin.com/tm3WDnAc |
| 06:23.51 | [TK]D-Fender | And you aren't loking at the debug to prove whre anything is actually loking... |
| 06:25.55 | bhavikpatel6842 | http://pastebin.com/ydTfKDCL |
| 06:26.00 | bhavikpatel6842 | here is the full log |
| 06:28.40 | bhavikpatel6842 | Let me know what's wrong in audio ? |
| 06:29.05 | [TK]D-Fender | <--- SIP read from WS:192.168.1.13:14944 ---> |
| 06:29.07 | [TK]D-Fender | INVITE sip:8001@192.168.1.31 SIP/2.0 |
| 06:29.19 | [TK]D-Fender | Reliably Transmitting (NAT) to 192.168.1.13:27194: |
| 06:29.21 | [TK]D-Fender | INVITE sip:8001@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 |
| 06:29.33 | [TK]D-Fender | call FROM 192.168.1.13 going TO 192.168.1.13 |
| 06:29.47 | [TK]D-Fender | You seem to have a serious issue with the idea of not calling yourself |
| 06:30.38 | bhavikpatel6842 | to party's connected properly using two browser. |
| 06:30.46 | bhavikpatel6842 | but seems like RTP issue |
| 06:30.57 | [TK]D-Fender | that's the same IP for both ends |
| 06:31.15 | [TK]D-Fender | And there's no way I trust whatever that is |
| 06:32.58 | bhavikpatel6842 | let me give you another logs of live server. |
| 06:33.05 | bhavikpatel6842 | that may be help ful. |
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| 06:34.41 | bhavikpatel6842 | http://pastebin.com/pt343Cmc |
| 06:36.39 | [TK]D-Fender | <--- Transmitting (NAT) to 122.170.89.220:17129 ---> |
| 06:36.41 | [TK]D-Fender | SIP/2.0 100 Trying |
| 06:36.46 | [TK]D-Fender | Reliably Transmitting (NAT) to 122.170.89.220:27194: |
| 06:36.47 | [TK]D-Fender | INVITE sip:8001@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 |
| 06:36.51 | [TK]D-Fender | Sampe IP again. |
| 06:36.55 | [TK]D-Fender | same |
| 06:38.29 | bhavikpatel6842 | so what wrong in my configuration and settings |
| 06:38.30 | bhavikpatel6842 | http://pastebin.com/LXA22kWZ |
| 06:38.34 | [TK]D-Fender | I'm out of time |
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| 07:47.06 | Tuju | i've an udp sip-trunk between two asterisk boxes. and if there is a connection problem, there is always problems to get the trunk re-register itself. nat is involved, any idea what is causing this? |
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| 08:44.10 | As001 | Hello, I listened advice from here to upgrade 1.8.23 to 1.8.27 version, it works but I have still issue with multiple Bridge events in manager interface for some calls. Is it known issue ? Did anyone have similar problem with Queue application and AMI ? |
| 08:46.04 | As001 | I use perl module Component::Client::Asterisk::Manager to connect to AMI and listen on events and send commands, but I don't know why it receives multiple Bridge for same call from time to time. |
| 08:46.29 | WIMPy | Have you looked at them? |
| 08:46.51 | WIMPy | Are you using local cahnnels? reinvites? |
| 08:47.08 | As001 | I am using Queues and agents who are logged in in Queues. |
| 08:48.24 | As001 | I can look at those Birdge events but it's now so easy beacuse of lot of calls and agents and because it does not happen always just 3-4 times a day but when It happens it see 500-1000 Bridges for just one call... |
| 08:48.54 | As001 | I can telnet to AMI and wait for event to show up. |
| 08:48.54 | Nugget | telnet is eeeeeeevil! |
| 08:49.21 | WIMPy | Smells fishy. |
| 08:49.34 | WIMPy | Do you use ICE? |
| 08:49.46 | As001 | no |
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| 08:54.28 | As001 | i read on internet about some bug but in version 10 regarding dtmf, Asterisk do Bridge leave and Bridge join after dtmf signal, but I think it's not my problem as I tried to do that (be an agent and press numbers on xlite, or numbers on clients phone). |
| 08:57.26 | As001 | I meant link and unlink in Bridge |
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| 09:03.30 | As001 | i filtered all events except Bridge and now I see just Bridge events on telnet, when it happen I will know. |
| 09:04.16 | vlad_starkov | Question: Asterisk 11.5.1. is there bug with app_queue.so module? I've changed default strategy in particular queue, then in CLI I made `module reload queue.so`, and it hasn't taken any effect, as `queue show my_queue` still has default strategy. |
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| 09:14.24 | vlad_starkov | Ok this is bug. Will post it to #asterisk-bugs |
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| 09:29.44 | As001 | Whimpy I saw 2 Bridge events on telnet it's asterisk who emit 2 or more bridge for same call. |
| 09:32.09 | As001 | Timestamp differs for 5 seconds and both are Bridgestate: Link |
| 09:33.33 | As001 | http://paste.ubuntu.com/7451548/ |
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| 09:35.26 | michael_work | the bug that crashes on ast_free on music on hold files is so annoying :( |
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| 10:36.03 | As001 | It seems my agents learn trick to go to line2 (on hold) on xlite during a call and then come back to line1 and then Asterisk fires new Bridge. |
| 10:36.39 | r00f | what is the purpose of that trick? |
| 10:36.59 | As001 | to count them call more because I catch Bridge and do calls=calls+1 for agents.. lol |
| 10:37.24 | As001 | now I will fix script to count just first Bridge. |
| 10:37.45 | bulkorok | make a limit... |
| 10:37.50 | r00f | wow. so when it fired 5000 bridges that agent would get the monthly plan completed? |
| 10:38.40 | bulkorok | sip-config: call-limit = number : Number of simultaneous calls through this user/peer. |
| 10:39.03 | As001 | no just team leader will not get angry on him when see he has only 20 calls today. |
| 10:41.28 | As001 | thanks bulkorok I will check that. |
| 10:41.52 | bulkorok | sure |
| 10:43.11 | As001 | bulkorok I have call-limit 1 in real time sip_conf table for all agents. |
| 10:43.33 | bulkorok | so I guess putting on-hlod doesn't count... |
| 10:43.51 | bulkorok | check what xlite does when they opush the hold-button and stop that with asterisk... |
| 10:44.26 | As001 | apologise I don't have for all agents. |
| 10:45.01 | As001 | I will recheck or stop hold button in Asterisk. |
| 11:01.18 | As001 | bulkorok can you point me how can i prevent call from going on hold in Asterisk. call-limit 1 did not fix issue. |
| 11:01.46 | bulkorok | we have to see, what happens when they hit "hold" |
| 11:02.11 | bulkorok | AND dial a new number |
| 11:02.14 | As001 | you mean bridge events emited when they hit hold ? |
| 11:02.25 | bulkorok | trace |
| 11:02.43 | As001 | ok i will go to collect that information. Thanks for help. |
| 11:02.46 | bulkorok | trace, wireshark => voip-calls |
| 11:02.53 | As001 | ok |
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| 11:35.06 | Tuju | how do i make a default route to dialplan? |
| 11:35.18 | Tuju | tried _ and s but neither worked. |
| 11:42.40 | wdoekes | _! <-- matches zero or more characters |
| 11:43.13 | Tuju | mmm, i try that, thanks! |
| 11:43.45 | Tuju | The use of '_!' for an extension is strongly discouraged and can have unexpected behavior. Please use '_X!' instead at line 39 of extensions.conf |
| 11:44.11 | r00f | there is a reason, u know |
| 11:44.15 | Tuju | i'm getting spanked butt red by asterisk. |
| 11:44.53 | Tuju | i wish "ls * " would also give such warning, every time... |
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| 13:31.58 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 13:32.59 | dan_j | Hi. When using DUNDi, if the other server doesn't answer the call, how do I get the dialplan to continue on the original server? |
| 13:33.39 | dan_j | At the moment, all i get is Hungup, No one is available to answer, and Auto fallthrough. |
| 13:34.12 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:34.12 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 13:58.17 | sgriepentrog | dan_j: try dial option g, which make it continue executing dialplan even after a call was answered & hungup. |
| 14:01.13 | dan_j | sgriepentrog: to use dundi, you don't use dial. |
| 14:01.27 | dan_j | You just use switch => DUNDi/context |
| 14:02.49 | file | that is one way of doing it |
| 14:03.45 | dan_j | I'm trying to use Dundilookup and then using the returned info to create a dial, but at the moment, dundilookup is returning nothing. |
| 14:03.47 | dan_j | not sure why |
| 14:03.54 | dan_j | working on it |
| 14:03.57 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-duzkppifhecxctit) |
| 14:03.58 | file | that would be the way to do what you want |
| 14:04.42 | dan_j | ok. thanks for confirming that i'm headed in the right direction. |
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| 14:16.18 | WIMPy | You can just mix the switch and normal dialplan. |
| 14:16.52 | WIMPy | If the switch doesn't take any action, it will start "normal" dialplan. |
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| 15:55.43 | happy-dude | HI all, I was wondering if I can get any help configuring fail2ban for asterisk/ SIP? |
| 15:58.41 | happy-dude | currently, I am looking at this page: http://www.fail2ban.org/wiki/index.php/Asterisk |
| 16:00.36 | *** join/#asterisk aness (~aness@2a02:fe0:c310:8b20:4d64:8276:4cab:9950) |
| 16:02.30 | [TK]D-Fender | Documentation seems to cover a lot. |
| 16:02.40 | [TK]D-Fender | What part of it do you actually need help with? |
| 16:02.55 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 16:06.20 | happy-dude | [TK]D-Fender: at the moment, I'm holding off on implementing until 2pm est |
| 16:06.25 | happy-dude | doing stuff for a school project |
| 16:06.31 | happy-dude | (voip pentest + mitigation stuff) |
| 16:09.14 | *** join/#asterisk navaismo (~navaismo@189.146.15.159) |
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| 16:09.56 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 16:21.02 | ChannelZ | the main thing is the 'action' and modifying it/writing your own to connect to your firewall however it makes sense to on your system. |
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| 16:42.24 | *** join/#asterisk bstarek (~bstarek@41.230.24.208) |
| 16:42.32 | bstarek | Hello everybody |
| 16:42.36 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:42.44 | navaismo | o/ |
| 16:43.17 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 16:44.16 | bstarek | I have been trying to fix this issue that I have with Asterisk 11, I am able to send SIP messages, but when I reply back, the message is not send to the receiver but is echoed back to me. |
| 16:44.31 | bstarek | I dont know if i am clear enough with the explanation? |
| 16:45.03 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 16:45.30 | bstarek | I am willing to post my configuration if needed. |
| 16:47.56 | navaismo | sure post your dialplan and the cli output, sounds like you dont set correctly the recipient |
| 16:51.44 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 16:51.52 | bstarek | navaismo, hello, thank you for answering, should i pastebin it? |
| 16:52.03 | navaismo | yes |
| 16:53.01 | bstarek | here you go, thank you, http://pastebin.com/0KKTAm0C |
| 16:55.49 | navaismo | and the cli output for a real try |
| 16:56.15 | bstarek | here you go: http://pastebin.com/Ddm9QQzF |
| 16:58.55 | [TK]D-Fender | <PROTECTED> |
| 16:58.57 | [TK]D-Fender | <PROTECTED> |
| 16:59.03 | [TK]D-Fender | Sure looks like the response back is messed up |
| 16:59.10 | [TK]D-Fender | it came in with from & to the same |
| 16:59.14 | bstarek | thats exactly the issue |
| 16:59.24 | [TK]D-Fender | look at the actual SIP debug for that exchange |
| 16:59.28 | [TK]D-Fender | "sip set debug on" |
| 16:59.34 | [TK]D-Fender | and pastebin a new attempt |
| 16:59.46 | bstarek | OK let me try that now, thank you guys. |
| 17:00.15 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 17:04.50 | bstarek | here you go |
| 17:04.50 | bstarek | http://pastebin.com/ucgv42b7 |
| 17:04.58 | bstarek | debug has a lot of output |
| 17:04.59 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 17:06.10 | [TK]D-Fender | To: "BOUBOU" <sip:201@192.168.1.76> |
| 17:06.12 | [TK]D-Fender | From: "201" <sip:201@192.168.1.76>;tag=qyznt |
| 17:06.18 | [TK]D-Fender | yup the raw from/to looks messed up |
| 17:06.20 | [TK]D-Fender | try another client |
| 17:08.55 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 17:10.15 | bstarek | sorry, I will now. |
| 17:10.21 | bstarek | what do you suggest? |
| 17:10.38 | bstarek | Did you ever noticed this issue before? |
| 17:10.49 | navaismo | are you sending the msg from the same PC? |
| 17:11.12 | bstarek | No, two different PCs, two different IPs |
| 17:12.29 | navaismo | how do you send the messages? which client? |
| 17:13.09 | bstarek | Twinkle from Linux |
| 17:13.26 | bstarek | 2 differents PCs |
| 17:13.28 | bstarek | Same client |
| 17:14.51 | bstarek | I am going to try "jipsi" client |
| 17:15.33 | SuperNull | what kind of load is g711->g729 on newer gen cpus ? |
| 17:15.47 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 17:15.58 | SuperNull | am i talking tens of calls or hundreds ? |
| 17:17.00 | Qwell | hundreds |
| 17:19.00 | navaismo | does the core show translation works for get that?? |
| 17:20.22 | SuperNull | anyone use those Sangoma enterprise SBCs? were looking to move asterisk to mainly a switching position in the middle and voice/media services only.. |
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| 18:09.44 | kuku | I have asterisk 1.4 ( I know its out of date ). After updating kernels, the fake ring sound, and all the voice recordings that get played back during the IVR are garbled up a bit, but the phone conversation is fine. Any ideas ? |
| 18:12.31 | *** join/#asterisk [sr] (~kvirc@pal-213-228-163-73.netvisao.pt) |
| 18:13.14 | navaismo | ~upgrade |
| 18:13.14 | infobot | Upgrading is easy! Go that way, really fast. If something gets in your way, turn. |
| 18:14.03 | navaismo | ~dahdi |
| 18:14.03 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel (more info at http://www.asterisk.org/dahdi ) and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 18:14.39 | navaismo | ~iax2 |
| 18:14.40 | infobot | hmm... iax2 is http://www.voip-info.org/wiki-IAX |
| 18:14.40 | *** join/#asterisk bmurt (~brendan@static-96-245-76-214.phlapa.fios.verizon.net) |
| 18:14.52 | navaismo | and the sound? |
| 18:15.15 | kuku | so upgrade dahdi ? |
| 18:15.24 | navaismo | e-e-aks? |
| 18:15.37 | navaismo | kuku, not for you im stressing the bot |
| 18:15.53 | kuku | ~what ? |
| 18:15.53 | infobot | idk |
| 18:16.03 | navaismo | about your issue maybe is related with timing sources |
| 18:16.13 | kuku | Its a virtual machien |
| 18:17.35 | navaismo | uh so yea timing issues |
| 18:17.57 | navaismo | ~iax |
| 18:17.57 | infobot | IAX uses port 5036 for the original (deprecated) IAX protocol. Port 4569 is for the IAX2 (current) protocol. IAX is pronounced "Eeks" and stands for Inter-Asterisk eXchange. See http://tools.ietf.org/html/rfc5456 |
| 18:18.09 | navaismo | argh and the sound!! |
| 18:18.37 | navaismo | infobot add http://translate.google.com/translate_tts?tl=en&q=%22Eeks%22 |
| 18:21.31 | *** join/#asterisk kannan (~chatzilla@42.104.61.211) |
| 18:23.15 | kannan | hello all. I need to make a fast decision on speech recognition with asterisk 1.8. certified version. Lumenvox and Vestec used to be on digium store, but i don not see them there now. How does gree google asr compare? In short, which is the best chivce enginge for ASR? |
| 18:24.46 | navaismo | with the payment options you have support and a good ASR engine, with google you have a "fine" (non-)free engine without support |
| 18:37.59 | kannan | the precise challenge i have is, we have been trying with vestec. it sowkrs ok except in the events of longer sequences of digits, like, 12 or 15 digit customer account numbers, the recog rate and experience is really poor |
| 18:38.16 | kannan | works ok, i meant |
| 19:12.23 | kuku | navaismo: So how do I resolve the timing issue ? other than upgrading to the latest? |
| 19:12.42 | *** join/#asterisk Mr_Tac (~TC@94.76.240.191) |
| 19:14.52 | navaismo | well asterisk >1.8 dont need dahdi for timing... but in your case make sure the dahdi/zaptel module is loaded |
| 19:17.15 | *** join/#asterisk kannan (~chatzilla@123.238.108.118) |
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| 19:44.32 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:45.34 | kuku | I currently can't upgrade to >1.8 so I need to make it work. Especially that it worked. |
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| 20:11.01 | dan_j | When using dundi to route a call from one server to another, how can I include the DID that was dialled? Normally I do SIP/PEERNAME/DID, but with the results from DUNDI, i dial IAX2/iaxdundi:password@10.50.0.32/PEERNAME which doesnt leave room for the DID. |
| 20:12.09 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 20:21.12 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 20:22.25 | navaismo | see a sip proxy is better for that |
| 20:26.53 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 20:32.53 | ghoti | So... "sip show inuse" seems to be telling me that I have two channels active on the phone on my desk. That is, the "In use" column shows "2/0/0". I'm not on the phone. What does this mean? |
| 20:33.21 | ghoti | (Am I really *on* the phone in an alternate reality, to which my asterisk install is connected through a subspace SIP gateway?) |
| 20:34.08 | ghoti | was impressed with asterisk already, but this is ... amazing. |
| 20:37.16 | navaismo | core show channels verbose maybe a stucked call |
| 20:37.20 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 20:40.13 | jmmills | Is call identifier logging backported to 1.8? |
| 20:40.29 | jmmills | i.e. so that I can look at a log file, see a call id and find all of the channel actions on it |
| 20:42.11 | navaismo | amm i use the numbers inside the brackets of VERBOSE |
| 20:44.36 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 20:44.53 | *** join/#asterisk lachesis (~lachesis@unaffiliated/lachesis) |
| 20:59.48 | ghoti | navaismo: something does indeed seem to be stuck. Two calls from my extension with a duration of "400:30:3". :-P |
| 21:00.10 | ghoti | One seems to be to extension "*72" (forwarding I think), and theother is to "*98" (voicemail login). |
| 21:00.22 | ghoti | Is there an easy way to clear these from the asterisk commandline? |
| 21:00.54 | dan_j | What characters are valid in a SIP address? Is ^ allowed? |
| 21:01.31 | dan_j | I want to pass two variables to another sip server, so i thought to tag the variables on the end of the peer name, separated with a ^. |
| 21:03.12 | navaismo | ghoti, hangup request SIP/id |
| 21:09.45 | *** join/#asterisk dumby (~dumby@50.8.119.230) |
| 21:13.36 | ghoti | navaismo: I get a "No such command" error... |
| 21:14.38 | ghoti | "channel request hangup" perhaps? |
| 21:14.43 | navaismo | yes |
| 21:15.19 | ghoti | got it. Thanks. |
| 21:27.33 | *** join/#asterisk CeBe (~CeBe@port-92-206-57-178.dynamic.qsc.de) |
| 21:45.50 | dan_j | Hi. If a peer is online, why would ChanIsAvail return a 0 for the status? |
| 21:48.51 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 21:49.37 | dan_j | All I want to do is check if a Peer is online or not. |
| 21:49.40 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-lczyntwtpnrhpqny) |
| 21:58.41 | gushi | Is there an app that will cause asterisk to generate a dialtone? We have some people wardialing through our extension pool, and I want to make them think they've got what they're looking for. |
| 21:58.56 | gushi | (i.e. an outside line) |
| 22:00.18 | gushi | It wouldn't have to respond to anything...just...sound like a dialtone for 10 seconds and hang up. |
| 22:03.49 | *** join/#asterisk viasanctus (~viasanctu@unaffiliated/viasanctus) |
| 22:04.08 | viasanctus | is it possible to see who's calling or not on a sip telephone on asterisk? |
| 22:04.22 | viasanctus | something like ext XXX in call |
| 22:04.25 | viasanctus | a list |
| 22:05.38 | ChannelZ-Wk | depends on how |
| 22:05.49 | viasanctus | needs to be on a sip phone, not the web interface |
| 22:06.12 | viasanctus | management wants to see the dialstatus (or what to name it) |
| 22:07.00 | ChannelZ-Wk | Well there is BLF |
| 22:07.17 | viasanctus | blf? |
| 22:07.44 | viasanctus | ah yes |
| 22:07.46 | ChannelZ-Wk | There's no "standard" way to display random information on a SIP phone but certani phones might give you ways to do it. I know some models basically have little web browsers built into them, whereby you could create a little webapp to display the info |
| 22:08.03 | viasanctus | thats too much |
| 22:08.09 | viasanctus | the busy lamp info is interesting |
| 22:08.36 | ChannelZ-Wk | You just need a phone with lots of line keys, or a sidecar or something. |
| 22:08.42 | viasanctus | yep |
| 22:08.50 | ChannelZ-Wk | depending on the size of the system |
| 22:08.55 | viasanctus | and then configure one of the leds as a blf or extension call ? |
| 22:09.15 | viasanctus | press it, calls, but at the same time it blinks or whatever when the person is in a call |
| 22:09.40 | ChannelZ-Wk | That's more of a SLA thing but yes |
| 22:09.52 | viasanctus | great |
| 22:09.58 | viasanctus | may God bless you |
| 22:10.02 | viasanctus | gets back to work |
| 22:16.24 | *** join/#asterisk navaismo (~navaismo@189.146.15.159) |
| 22:21.15 | viasanctus | doe the cisco 7945G or cisco 7942G work with asterisk? |
| 22:21.25 | viasanctus | most cisco telephones require the cisco call manager |
| 22:21.41 | viasanctus | but for ex the SPA 303 not, you can simply configure it with asterisk |
| 22:21.57 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 22:22.36 | viasanctus | fuck, I need to update firmware |
| 22:22.37 | viasanctus | sigh |
| 22:24.02 | navaismo | better buy digium phones, but if no money buy yealink |
| 22:24.17 | viasanctus | digium |
| 22:24.22 | viasanctus | yealink |
| 22:24.29 | viasanctus | with BLF |
| 22:24.50 | navaismo | almost all phones support blf |
| 22:26.54 | [TK]D-Fender | [18:09]ChannelZ-WkThat's more of a SLA thing but yes <- Which * doesn't really do anyway |
| 22:27.32 | [TK]D-Fender | Polycom > All for most uses |
| 22:28.12 | viasanctus | ok which is the best telephone for asterisk price/quality wise, we have been succesfully using cisco devices |
| 22:28.25 | viasanctus | only prob was the coded, doesn't seem to work well with G722 |
| 22:28.31 | viasanctus | best is G711a |
| 22:28.55 | viasanctus | not sure if it is/was the device or the idsn truncs or the asterisk config |
| 22:29.30 | navaismo | yealink good, cheap and functional. Very Good Digium Phone, its like a polycom with steroids |
| 22:30.10 | viasanctus | yealink it is |
| 22:32.24 | *** join/#asterisk imox (~imox@p57A96BD5.dip0.t-ipconnect.de) |
| 22:33.52 | lvlinux | yealink SIP-T46g is an awesome phone---just got 2 for a client and they are really nice. |
| 22:34.02 | viasanctus | and they can hadnle BLF? |
| 22:34.05 | viasanctus | multiple buttons? |
| 22:37.38 | navaismo | visit the vendor page |
| 22:38.56 | *** join/#asterisk tuxd00d (~tuxd00d@ip24-251-34-116.ph.ph.cox.net) |
| 22:39.20 | navaismo | if you are interested in LED capabilities see this video -->https://www.youtube.com/watch?v=lvOGrA0cYiM |
| 22:39.55 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 22:40.28 | ChannelZ-Wk | [TK]D-Fender: well true, it's all hacked with conferences I guess? Although in *12 it should be much easier to do since everything is basically a conference already. Either way it's something I've never screwed with personally |
| 22:41.46 | [TK]D-Fender | So much worse than that |
| 22:45.00 | [TK]D-Fender | viasanctus: How many other users do you expect to monitor? Is this for a receptionist-type person? |
| 22:48.17 | *** join/#asterisk Alex_Bkash (~atique@124.6.236.252) |
| 22:48.35 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 22:50.38 | viasanctus | yes |
| 22:50.44 | viasanctus | and i think about 10 |
| 22:51.12 | viasanctus | no wait |
| 22:51.14 | viasanctus | 6 |
| 22:51.17 | viasanctus | minimal |
| 22:53.18 | viasanctus | which device is this: http://www.abilis.net/tutorial/7.6/en/ch46s06.html |
| 22:54.01 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 22:54.03 | [TK]D-Fender | looks like a YealinkT-28 + sidecars |
| 22:54.18 | viasanctus | that's the one I'm looking for |
| 22:55.08 | [TK]D-Fender | http://www.telephonydepot.com/Catalog/Yealink-Phones/Yealink-SIP-T28P-6-line-IP-Phone |
| 22:55.28 | [TK]D-Fender | http://www.telephonydepot.com/Catalog/Yealink-Phones/Yealink-Expansion-module-Exp-39 |
| 22:55.38 | [TK]D-Fender | This sidecar looks a lot better.... |
| 22:58.44 | [TK]D-Fender | Mind you a phone like this can do it all in 1 piece and looks a lot nicer esp with poor lighting |
| 22:58.55 | [TK]D-Fender | but for raw phone quality, Polycom > all |
| 22:59.09 | viasanctus | it'll be 711a |
| 22:59.30 | [TK]D-Fender | I'm talking physical build, audio, etc |
| 22:59.34 | [TK]D-Fender | Codec alone isn't everything |
| 22:59.50 | viasanctus | does it matter how clear the voice sounds if the codec is of lower quality |
| 22:59.55 | viasanctus | just a q |
| 22:59.57 | [TK]D-Fender | The Cisco/Linksys SPA's are a tier below for sure |
| 22:59.58 | viasanctus | not a retoric q |
| 23:00.08 | viasanctus | they sound like crap |
| 23:00.27 | [TK]D-Fender | [18:59]viasanctusdoes it matter how clear the voice sounds if the codec is of lower quality <- yes, it does still matter |
| 23:00.58 | viasanctus | we'll if i can convice my colleauge dealing with VOIP in our company, we'll try the yealinks |
| 23:01.00 | [TK]D-Fender | The speaker & mic quality matter a LOT, so does the material build quality, engineering, AEC, etc |
| 23:01.14 | [TK]D-Fender | Yealink is .... ok-ish |
| 23:01.18 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 23:01.34 | [TK]D-Fender | Polycom & Aastra are a clear step above. |
| 23:01.50 | viasanctus | but more expensivE? |
| 23:02.50 | ChannelZ-Wk | Yea get what Yea pay for |
| 23:02.55 | ChannelZ-Wk | HAH! |
| 23:03.04 | viasanctus | can you find me an alternative for the above yealink in polycom? with blf's |
| 23:06.50 | [TK]D-Fender | http://www.ipphone-warehouse.com/Polycom-VVX-Color-Expansion-Module-p/2200-46350-025.htm |
| 23:07.11 | [TK]D-Fender | http://www.ipphone-warehouse.com/Polycom-VVX-Phones-s/768.htm |
| 23:07.16 | [TK]D-Fender | shop around for a combo. |
| 23:07.20 | [TK]D-Fender | This is worth some reading. |
| 23:07.38 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 23:07.38 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 23:07.41 | [TK]D-Fender | That single Aastra I suggested could do the job by itself... |
| 23:07.50 | [TK]D-Fender | and the touch-screen IS really nice actually... |
| 23:08.05 | [TK]D-Fender | Haven't used the Polycom VVX's personally, but rather the SoundPoint IP series |
| 23:08.08 | [TK]D-Fender | And those were solid |
| 23:08.18 | viasanctus | woah the prices!! |
| 23:08.38 | [TK]D-Fender | [19:02]ChannelZ-WkYea get what Yea pay for |
| 23:15.03 | *** part/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 23:16.41 | [TK]D-Fender | out for a while, back later |
| 23:18.12 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 23:19.26 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
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| 23:31.33 | viasanctus | [TK]D-Fender, alright, take care and thank you so much for the feedback! may God bless you |