| 00:01.09 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 00:04.49 | *** part/#asterisk kharwell (kharwell@nat/digium/x-wjbssorcxzbpsgdb) |
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| 02:07.23 | *** join/#asterisk cmendes0101 (~cmendes01@pool-173-60-103-244.lsanca.fios.verizon.net) |
| 03:31.37 | *** join/#asterisk captain118 (~kirk.mcca@50-81-28-72.client.mchsi.com) |
| 03:32.18 | captain118 | Is there a digium support channel? |
| 03:40.28 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 03:49.57 | wyoung | captain118: on their website |
| 03:50.23 | wyoung | although most of the people in here may be able to assist you (unless it is relating to RMA's :)) |
| 04:11.23 | captain118 | I've got their cloud sip trunk and I'm trying to setup the firewall securely. |
| 04:16.49 | *** join/#asterisk juned (~juned@202.131.119.122) |
| 04:38.20 | *** join/#asterisk felix___ (62eba05a@gateway/web/freenode/ip.98.235.160.90) |
| 04:39.11 | felix___ | I'm having issues using Comcast Digital voice fake POTS lines with asterisk. |
| 04:39.24 | felix___ | asterisk 1.8, 11, 12, 13 |
| 04:40.07 | felix___ | DAHDI with Rhino or Xorcom hardware, Sangoma Vega50, doesn't seem to matter |
| 04:40.42 | felix___ | Hardware or software echo cancellation, both have the same issue |
| 04:41.13 | felix___ | Echo during calls inbound and outbound, some voice cutting out |
| 04:41.41 | felix___ | I've tried PBXmate also |
| 04:42.31 | felix___ | The best quality so far has been with mg2 and echocancellation=64 |
| 04:42.44 | felix___ | Anyone out there experience similar issues? |
| 05:03.24 | felix___ | quit |
| 05:03.31 | wyoung | captain118: ok |
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| 08:04.42 | tparcina | voicemail, I can put time in e-mail with ${VM_DATE} |
| 08:05.01 | tparcina | That puts date in format: Friday, January 08, 2016 at 09:02:54 AM |
| 08:05.52 | tparcina | Is there any way I can set this time in format: 09:02 08. 01. 2016. |
| 08:06.05 | tparcina | hh:mm dd. mm. yyyy. |
| 08:10.24 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 08:27.25 | tparcina | Found it! :) |
| 08:27.41 | tparcina | I just head to put "emaildateformat=%H:%M %d. %m. %Y." in voicemail.conf |
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| 09:15.03 | *** join/#asterisk Bordr_ (~Bordr@23-24-156-49-static.hfc.comcastbusiness.net) |
| 09:16.01 | Bordr_ | hagbard you out there? |
| 09:16.22 | Bordr_ | mbecroft you out there? |
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| 09:56.51 | *** join/#asterisk Phil-Work (~Phil-Work@185.28.212.96) |
| 09:57.44 | Phil-Work | what's the safest way to do string comparison in Asterisk? I'm currently comparing the return value from a CURL request to a string like this... same => n,GotoIf($["${agent_validation}" = "OK"]?agent-success) |
| 09:58.11 | Phil-Work | but if ${agent_validation} contains HTML, it triggers an error in ast_expr2.fl |
| 10:07.54 | *** join/#asterisk mohan_ (ca4efb97@gateway/web/freenode/ip.202.78.251.151) |
| 10:14.11 | mohan_ | hi i have a question regarding dial pattern, in the calling queue has 2 different dial patterns 1 starting with 0 and other not with 0, the number which starts with 0 should be dialed as it exists and the number which is not prefixed with 0 should be dialed by prefixing with 0 |
| 10:16.57 | mohan_ | currently i have this in the extensions conf exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _X.,n,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)}) exten => _X.,n,Set(Client_cli=${EXTEN:1}) exten => _X.,n,Set(FILENAME=${Client_cli}_${CALLTIME}.wav) exten => _X.,n,Set(RECORDFILENAME=${FILENAME}) exten => _X.,n,Set(SOUND_PATH=${RECORDING_KINREP}/${RECORDFILENAME}) exten => _X.,n,MixMonitor(${SOUND_PATH}) |
| 10:21.21 | mohan_ | the straight question is can i have more than 1 dial plan in same context |
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| 10:39.12 | hagbard | Bordr_: what sup |
| 10:41.30 | mohan_ | while i read through document understood that there can be more than 1 Dial function in a context with different priorities and each can have different pattern |
| 10:42.22 | Bordr_ | Hey hagbard. Wanted to say thanks for the all the help last night. GREATLY appreciated. Problem turned out to be power supply (entire circuit) loose nuetral wire. Found voltage fluctuating at the outlet. Thats why replacement hardware wasn't helping :).... |
| 10:42.48 | Bordr_ | Last 2 entries - hadn't enabled offsite phones until 1:45 this afternoon. |
| 10:42.48 | Bordr_ | [2016-01-07 10:22:48] NOTICE[2984] chan_sip.c: Peer 'peakteks' is now Reachable. (63ms / 2000ms) |
| 10:42.48 | Bordr_ | [2016-01-07 13:48:42] NOTICE[2984] chan_sip.c: Peer '116' is now Reachable. (46ms / 2000ms) |
| 10:42.51 | hagbard | Bordr_: how the hell did you discover that? didn't someone get shocked? |
| 10:43.25 | Bordr_ | No I ran out of options and started checking everything I could think of. The only commonality after replacing the server was the outlet on the wall... |
| 10:44.37 | Bordr_ | Have been watching and only 1 packet missed, arping is happy without any errors, no dropped calls and not one unreachable since 10:22 (about 17 hours ago). |
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| 11:19.04 | mbecroft | Bordr_: that is crazy |
| 11:21.46 | mbecroft | It would have taken a long time to occur to me that the server could run fine in every respect except for oddly specific IP packet corruption, with multiple different NICs, due to a mains power fault |
| 11:23.16 | Bordr_ | The server is on UPS, switch wasn't. Since I had replaced everything (i mean everything :), there was only one commonality left... Simple check showed 91v spiking to 150 (should have been 120). Check hot and ground - 123v steady. |
| 11:23.53 | Bordr_ | Used an extension cord across room to different circuit and problems haven't returned since. |
| 11:25.00 | Bordr_ | Electrician is coming by in the morning to address further. Requested he replace with GFI/ACFI to prevent issue in the future. |
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| 12:06.10 | mbecroft | Bordr_: still one of the most bizarre faults I've ever heard of |
| 12:07.00 | mbecroft | Especially as the fault started with the phones on the main switching infrastructure, which I imagine is on UPS power even if your test switch was not |
| 12:08.21 | Bordr_ | No, the core switch was on UPS but the POE switch wasn't. UPS is only 1500va and the POE switch cut down the run time... They were supposed to get a new UPS for the switch but never did. |
| 12:08.43 | mbecroft | Ah |
| 12:09.16 | Bordr_ | Been in IT for 15+ years and first time I have seen mains power cause an issue like this, every other time it was identified in under a minute... |
| 12:09.37 | mbecroft | Yeah, pretty insane set of symptoms |
| 12:10.46 | mbecroft | Also that the same fault happened with both the main POE switch and the test switch in response to the unreliable power... |
| 12:10.49 | Bordr_ | After everything was fixed they told me the "soda machine guy" was out and moved the soda machine away from the wall. Thinking he pull the wire to the outlet and knocked the N wire loose. |
| 12:11.29 | Bordr_ | Yes it did. The managed switch and the non-managed both did the same thing. The managed swithc would sometime load pages fast and other times slow. |
| 12:11.46 | mbecroft | Wow. Good work diagnosing that |
| 12:11.48 | Bordr_ | Now I think it was booting up or reloading the management pages due to the power. |
| 12:12.27 | Bordr_ | Thanks, I really appreciate the help. |
| 12:13.14 | mbecroft | Any time |
| 12:13.36 | mbecroft | Are you a regular here or did you just drop on for this? |
| 12:13.43 | Bordr_ | I have a simply stupid question. How are you highlighting the response? You and hagbard both responded and it's highlighted... |
| 12:14.01 | Bordr_ | Not regular, got in here for this, but might hang around a while. |
| 12:14.18 | Bordr_ | Been helping people on centos and fpbx... |
| 12:14.31 | mbecroft | Usually if your nick is mentioned it is highlighted |
| 12:14.39 | mbecroft | e.g. Bordr_ <-- highlighted ? |
| 12:15.40 | Bordr_ | mbecroft yes exactly mbecroft |
| 12:15.46 | Bordr_ | did that highlight your side? |
| 12:15.59 | Bordr_ | (new to irc) sorry for the noob questions... |
| 12:16.37 | mbecroft | Yes |
| 12:17.16 | Bordr_ | Thank you sir! I'll hang around a bit, but for now I'm headed to bed. (5:15 am here in MST) |
| 12:18.26 | mbecroft | Sleep well--well deserved! |
| 12:19.04 | mbecroft | Was going to chat but will perhaps catch you some other day! |
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| 12:24.48 | *** join/#asterisk cusco (~tralala@2001:41d0:1:6caf::cafe) |
| 12:25.10 | cusco | hi |
| 12:26.14 | cusco | http://www.voipsupply.com/digium-1a8b00f can I choose from these 8 ports wich are fxs and wich are fxo, right? |
| 12:27.02 | cusco | basically so this is not a X-Y question: someone wants to place a method of recording calls. 4 landline phones... |
| 12:27.22 | cusco | So I was thinking of a asterisk between the phones and PSTN (POTS) |
| 12:30.56 | *** join/#asterisk chris-lx141_ (3effef3e@gateway/web/freenode/ip.62.255.239.62) |
| 12:33.04 | chris-lx141_ | Hi all |
| 12:36.08 | chris-lx141_ | I'm lokking for some help with my asterisk server. in sip.conf in siptrunk context I have set fromuser=USERNAME but when I'm doing a call in sip header is showing From :sip:<other-peer-name> and is failing to authenticate in sip trunk |
| 12:36.19 | chris-lx141_ | anyone know how to solve that problem ? |
| 12:37.39 | Bordr_ | cusco: you would need to add the fxo/fxs modules. |
| 12:37.48 | Bordr_ | cusco: then it would work |
| 12:38.35 | Bordr_ | chris-lx141_: can you post the config (change the password or just put in xxxxx) |
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| 12:44.45 | marceloamorim | guys, about openr2, there is something ( best practice )I test to send for the list asterisk-r2 ? I can't receive or make calls, seems to be busy all channels. |
| 12:57.55 | chris-lx141_ | Bordr: I do not see any fxo or fxs modules in /usr/lib64/asterisk/modules/ |
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| 13:00.27 | chris-lx141_ | Burdr: heh now I can see that your answer was not for me :-0 |
| 13:02.16 | chris-lx141_ | thats my sip.conf |
| 13:02.17 | chris-lx141_ | [general] context=incoming ; default context for incoming calls allowguest=no ; disable unauthenticated calls srvlookup=yes ; enabled DNS SRV record lookup on outbound calls tcpenable=no ; disable TCP support localnet=192.168.110.0/24 externaddr=XXX.XXX.XXX.XXX register => USERNAME:xxxxx@sip.voipfone.net/USERNAME [voipfone] host=sip.voipfone.net username= |
| 13:04.25 | chris-lx141_ | you can see it here https://bpaste.net/show/943a3573952e |
| 13:05.28 | chris-lx141_ | and in extensions.conf I'm testing with same => n,Dial(SIP/${EXTEN}@sip.voipfone.net) |
| 13:06.28 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
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| 13:06.48 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:13.42 | chris-lx141_ | I'm lokking for some help with my asterisk server. in sip.conf in siptrunk context I have set fromuser=USERNAME but when I'm doing a call in sip header is showing From :sip:<other-peer-name> and is failing to authenticate in sip trunk |
| 13:14.15 | chris-lx141_ | my sip.conf could be seen here https://bpaste.net/show/943a3573952e |
| 13:14.23 | *** part/#asterisk chendy (~alexc@59.40.99.113) |
| 13:14.26 | chris-lx141_ | and in extensions.conf I'm testing with same => n,Dial(SIP/${EXTEN}@sip.voipfone.net) |
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| 13:45.56 | Zogot | ahoy all, belated happy new year |
| 13:50.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:58.12 | wyoung | Zogot: belated happy hanukkah, kwanzaa and christmas too!!! |
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| 14:09.31 | [TK]D-Fender | chris-lx141_: Your dial is wrong. "and in extensions.conf I'm testing with same => n,Dial(SIP/${EXTEN}@sip.voipfone.net)" <- this sends the call directly to that host. |
| 14:09.58 | [TK]D-Fender | chris-lx141_: instead of the peer you defined. You need to dial that peer properly. Dial(SIP/voipfone/NUMBER) |
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| 14:47.44 | plod | anybody done asterisk fast start course, any good? |
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| 15:03.46 | *** join/#asterisk kharwell (kharwell@nat/digium/x-bouuqcgiaqgvcenq) |
| 15:06.36 | [TK]D-Fender | if you don't have time and need a course where they can pin you down and feed it to you ... then that's why it's there. |
| 15:06.55 | [TK]D-Fender | If you're more directly motivated then you could do fine with just the book. |
| 15:10.44 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-oisiwzdairxwhfee) |
| 15:12.50 | chris-lx141_ | [TK]D-Fender: thanks for help, it works now |
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| 15:43.12 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:43.54 | *** join/#asterisk acidfoo (45a5a03c@gateway/web/freenode/ip.69.165.160.60) |
| 15:45.27 | acidfoo | does call parking require DAHDI ? like meetme require DAHDI |
| 15:45.47 | WIMPy | No |
| 15:47.31 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 15:47.31 | *** mode/#asterisk [+o file] by ChanServ |
| 15:48.13 | acidfoo | ok |
| 15:50.31 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 15:50.32 | *** mode/#asterisk [+o file] by ChanServ |
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| 16:20.53 | captain118 | I have an asterisk phone and a pbx running freepbx. When I dial a 10 digit phone number a pick up my phone it dials properly. When I pick up my phone and start to dial once I go past the number of digits that is our internal extensions I get "You call can not be completed..." |
| 16:21.04 | captain118 | Thoughts? |
| 16:21.32 | captain118 | Excuse me I have a digium phone. |
| 16:21.55 | [TK]D-Fender | That is your phone's dialplan cutting you short |
| 16:22.13 | cyford | hi i am having the hardest time installing flite, anyone familiar with these errors http://pastebin.com/5c6aVdwJ |
| 16:22.17 | *** join/#asterisk bmg505 (~leon@196.209.233.206) |
| 16:23.06 | captain118 | Where do I change the phone's dial plan? |
| 16:23.08 | *** join/#asterisk cyford33 (~cyford@96.80.63.53) |
| 16:23.26 | WIMPy | On the phone. |
| 16:23.49 | captain118 | I dont see the option in the config on the site |
| 16:23.52 | WIMPy | And if you don't know what to put there, the best option usually is to leave it blank. |
| 16:24.04 | WIMPy | What site? |
| 16:27.34 | captain118 | The web page for the phone. |
| 16:27.55 | cyford | phones web gui? |
| 16:28.04 | captain118 | Yea |
| 16:28.07 | captain118 | its nice |
| 16:28.33 | captain118 | Though I eventually want to just pull the config from a server. |
| 16:28.52 | WIMPy | Is it? For me it only worked propperly once I use a browser it called unsipported. |
| 16:29.35 | captain118 | Sounds like you have a really old phone. |
| 16:29.55 | captain118 | You should find the latest firmware and update it. |
| 16:30.07 | cyford | hi i am having the hardest time installing flite, anyone familiar with these errors http://pastebin.com/5c6aVdwJ |
| 16:30.14 | WIMPy | That was the latest. |
| 16:30.28 | drmessano | cyford: Which version of Asterisk? |
| 16:30.50 | cyford | 11.20 |
| 16:31.18 | cyford | on centos 6.5 |
| 16:31.52 | cyford33 | <PROTECTED> |
| 16:32.19 | captain118 | WIMPy: huh. I've got a D40 Digium phone and the web page lets you do all the configs. Interestingly enough the default Digium config in the phone wanted you to push 91 for an outside line. |
| 16:33.09 | WIMPy | It worked nicely with Opera even though it says it won't. |
| 16:34.05 | captain118 | I'm using Chrome |
| 16:36.50 | *** part/#asterisk mcargile (~mikec@rrcs-97-76-33-146.se.biz.rr.com) |
| 16:42.29 | drmessano | cyford33: I don't see at all where it will work with anything newer than 1.4 |
| 16:42.49 | drmessano | Maybe I am looking at the wrong page |
| 16:43.26 | WIMPy | There we have it again. 1.4 was probably the best supported version. |
| 16:43.27 | drmessano | Nevermind.. I am |
| 16:43.39 | drmessano | But you have a VERY old version of Asterisk Flite |
| 16:44.07 | drmessano | http://zaf.github.io/Asterisk-Flite/ |
| 16:47.40 | cyford33 | when i try to install that one i get error http://pastebin.com/XqxCXDet |
| 16:47.49 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 16:48.03 | cyford33 | what are Asterisk header files |
| 16:48.03 | cyford33 | Flite 1.4 libraries and header files |
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| 16:50.09 | cyford33 | i installed asterisk from source btw |
| 16:51.24 | cyford33 | make |
| 16:51.36 | cyford33 | opps wrong screen |
| 16:51.52 | drmessano | You didn't do an ldconfig after installing it |
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| 17:18.12 | cyford33 | i just added contrib/scripts/install_prereq install |
| 17:18.12 | cyford33 | and recompiled with make menuselect ; make && make install && make config && ldconfig ----- but still same error |
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| 18:52.56 | gmalsack | good day all. having problems with call file placed in /var/log/spool/asterisk. ast 13.1-cert2 on centos 6 |
| 18:53.25 | gmalsack | call file worked when testing in home office. |
| 18:53.41 | newtonr | what happens? |
| 18:53.45 | gmalsack | when I took phones to corporate office to demo, call asked for password |
| 18:53.51 | gmalsack | puzzled me |
| 18:53.57 | [TK]D-Fender | Unless you have a paid support contract with Digium using an outdated CERT like that is a supreme waste of time |
| 18:54.13 | gmalsack | so then I tried calling my cell again, worked fine. |
| 18:54.26 | gmalsack | tried calling bosses cell. asked for password again. |
| 18:54.28 | [TK]D-Fender | And a call asking for a password... is nott a callfile issue |
| 18:54.40 | [TK]D-Fender | Go look at what it's actually doing |
| 18:54.45 | gmalsack | no it's not. seems to happen when the calls go into the basic bridge |
| 18:55.21 | *** join/#asterisk bmg505 (~leon@196-209-225-196.dynamic.isadsl.co.za) |
| 18:55.41 | gmalsack | http://pastebin.com/wfBqdM0D -> asks for password |
| 18:56.19 | gmalsack | http://pastebin.com/QFkBHgvB -> works perfectly.... |
| 18:57.00 | WIMPy | I don't see anything there. Is it your ITSP asking for the password? |
| 18:59.12 | gmalsack | [TK]D-Fender: outdated CERT.... funny, downloaded from ast.org on Dec 28th 2015 |
| 18:59.12 | newtonr | gmalsack, there is a voice on the call requesting a password? |
| 18:59.20 | newtonr | There is nothing obvious in the debug |
| 18:59.32 | [TK]D-Fender | - Executing [s@macro-trunkdial:5] Dial("SIP/3006-00000013", "SIP/flowroute/50261305#12628758005,,r") in new stack |
| 18:59.37 | gmalsack | yes, alisons voice |
| 18:59.39 | [TK]D-Fender | <PROTECTED> |
| 18:59.54 | [TK]D-Fender | VERY obvious difference in yoru dial out to flowroute |
| 19:00.04 | [TK]D-Fender | And thqat format shows you needing AUTHENTICATION in the dial you pass |
| 19:00.11 | [TK]D-Fender | and you did NOT do it the same on your other attempt |
| 19:00.20 | [TK]D-Fender | ops |
| 19:00.22 | [TK]D-Fender | bad aim |
| 19:00.25 | [TK]D-Fender | hold tthat thought |
| 19:00.39 | [TK]D-Fender | is going cross-eyed today |
| 19:00.54 | newtonr | :D |
| 19:00.58 | [TK]D-Fender | gah |
| 19:01.47 | newtonr | I need food or I'll go cross-eyed |
| 19:01.55 | [TK]D-Fender | Have a Snickers. |
| 19:01.57 | gmalsack | right! |
| 19:02.00 | [TK]D-Fender | You're not yourself when you're hungrgy |
| 19:02.23 | newtonr | mmmm chocolate |
| 19:02.23 | gmalsack | I have no food left. ate it all yesterday. time to the the gs |
| 19:02.59 | [TK]D-Fender | gmalsack: You hear this prompt after the Dial() we see there? |
| 19:03.07 | [TK]D-Fender | gmalsack: If so ... then that's Flowroute. |
| 19:03.12 | gmalsack | after the connect to flowroute |
| 19:03.23 | [TK]D-Fender | The "why" would remain in question then. |
| 19:03.34 | [TK]D-Fender | Look at the SIP debug to see if there is anything else different |
| 19:03.48 | WIMPy | Then you should find out why or when they ask for a password. |
| 19:03.50 | [TK]D-Fender | WAIT |
| 19:03.53 | [TK]D-Fender | I think I see it |
| 19:03.58 | gmalsack | right..... why the heck would flowroute do that.... dorks probably an old DID in their system they are not using anymore that the telco's recycled..... |
| 19:03.59 | [TK]D-Fender | <PROTECTED> |
| 19:04.08 | [TK]D-Fender | <PROTECTED> |
| 19:04.14 | [TK]D-Fender | You are Calling your OWN number |
| 19:04.26 | [TK]D-Fender | This would be their VOICEMAIL system kicking in because you are calling yorurself |
| 19:04.49 | [TK]D-Fender | And is a mechanism for picking up voicemail. |
| 19:04.55 | WIMPy | Ugh. You can't call yourself? |
| 19:05.24 | newtonr | I'll bet that is it |
| 19:05.29 | gmalsack | Normally I'd agree, but it's the same context running the call that does work. and the one that does work sets the caller id as well. ;-) |
| 19:07.07 | [TK]D-Fender | or I'm going cross-eyed again |
| 19:08.27 | [TK]D-Fender | Is the failed one using a callerID you don't own? |
| 19:08.35 | [TK]D-Fender | because they may not allow that |
| 19:08.38 | gmalsack | they both are |
| 19:08.57 | [TK]D-Fender | hrm |
| 19:09.06 | [TK]D-Fender | needs more coffee |
| 19:09.12 | gmalsack | but you're right, I do need to update the script so it sets the correct ocid |
| 19:09.17 | [TK]D-Fender | I'd certainly call up Flowroute and ask them what's going on |
| 19:09.36 | WIMPy | That would roule out the VM. |
| 19:10.06 | gmalsack | so definitely leaning towards it being flowroute? |
| 19:10.34 | WIMPy | Right from the beginning. |
| 19:11.32 | gmalsack | guess with it being alison's voice asking for the pw, I just thought that was a little too close to home to be coincidence.... however I guess it's not far off to think flowroute might be using ast |
| 19:12.12 | newtonr | Alison has also done a lot of work for other products and companies :) |
| 19:19.20 | gmalsack | ah, yea.... i suppose. duh! |
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| 19:33.35 | *** join/#asterisk rtpman (~rtpman@82.118.236.34) |
| 19:33.36 | rtpman | hi all |
| 19:35.15 | rtpman | i am doing a school project. I have redirected my sdp audio IP to another server in my call initiation packet. I receive the the RTP call data on the correct server when looking in wireshark, but Asterisk is dropping the call after a few seconds. Why would this be? |
| 19:35.46 | WIMPy | rtptimeout? |
| 19:36.00 | rtpman | i do not have control of the asterisk that is killing my call |
| 19:36.10 | rtpman | but all i have done different than normal, is change the media IP |
| 19:36.19 | rtpman | and I do receive media for a few seconds |
| 19:37.02 | rtpman | do I need to reply to the RTP packets ? is that why the call drops? |
| 19:37.05 | WIMPy | Usually it's a good idea to kill calls with no RTP comming in. |
| 19:37.17 | rtpman | I am just viewing them in wireshark and seeing them start tthen seeing them stop |
| 19:37.46 | WIMPy | Hmm. What are you redirecting exactely and from where? |
| 19:38.12 | WIMPy | Might be a good idea for a little ASCII art showing SIP and RTP streams. |
| 19:38.59 | rtpman | i make a call to example: 1.2.3.4. i put 2.3.4.5 as the audio IP. i run wireshark on 2.3.4.5 . I see audio data flowing properly. a Few seconds later, the audio data stops. |
| 19:40.05 | WIMPy | What are the components involved? |
| 19:40.31 | rtpman | well this only happens when the media ip is changed |
| 19:40.41 | WIMPy | And as I already said: Usually it's a good idea to drop calls when no RTP is received. So that might give you a clue. |
| 19:40.54 | rtpman | ok, so i must handle the RTP connectoin |
| 19:41.02 | rtpman | i cannot just actively monitor it using wireshark or the cal will drop |
| 19:41.03 | rtpman | correct? |
| 19:41.26 | WIMPy | Depends on configuration. |
| 19:41.34 | WIMPy | And we still don't know the full setup. |
| 19:41.35 | rtpman | ok |
| 19:41.38 | rtpman | asterisk default configuration |
| 19:41.44 | rtpman | does it drop calls which don't handle RTP |
| 19:41.58 | rtpman | because some RTP data IS received on the media server |
| 19:42.03 | rtpman | then the call is suddenly lost |
| 19:42.29 | WIMPy | There isn't really a default configuration. But I think rtptimeout is off by default. |
| 19:43.38 | WIMPy | Tell us EXACTELY what your setup looks like and we might be able to give you better answers. It's too unclear ATM. |
| 19:44.20 | rtpman | ok well i think i figured out the problem thanks to your help |
| 19:44.29 | rtpman | do you know if RTP requires handshakes to avoid timeout? |
| 19:45.00 | WIMPy | Not by itself. |
| 19:45.11 | rtpman | ok |
| 19:45.17 | WIMPy | But then there are things like NAT support that can play a role. |
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| 19:50.44 | rtpman | is there a rtp server standalone? |
| 19:50.51 | rtpman | that just handles the rtp connections? so it doesn't timeout? |
| 19:51.37 | rtpman | can i direct my rtp audio toward a asterisk? and will it handle it? |
| 19:52.14 | WIMPy | You need to make clear what you're trying to do. |
| 19:56.47 | rtpman | ? |
| 19:57.05 | rtpman | I'm trying to receive audio on a different server than the place the call was initiated from. |
| 19:58.48 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 20:00.07 | WIMPy | pa [n] da man! |
| 20:00.25 | pa | :) |
| 20:00.39 | pa | just upgraded to nx5 :-) |
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| 20:17.30 | rtpman | I'm trying to receive audio on a different server than the place the call was initiated from. Can anyone help with this please |
| 20:34.55 | rtpman | Is audio data sent and received on 1 udp port in RTP protocol? |
| 20:36.10 | WIMPy | Not neccessarily. |
| 20:37.02 | gmalsack | hello all ~ Seems to have been a flowroute problem. I have my system updated so it sends the outgoing callerid of our corporate office if the ocid and number dialed match as they were. |
| 20:37.11 | gmalsack | now the calls go just fine. |
| 20:37.39 | gmalsack | doesn't really make sense though why it worked calling my cell phone..... ocid and number dialed matched there as well. |
| 20:38.00 | gmalsack | guess it's an inconsistency in flowroutes dial plan based on area code maybe.... |
| 20:38.16 | rtpman | ok and when it is (in most cases i believe) |
| 20:38.51 | WIMPy | When you implement it that way. |
| 20:40.12 | rtpman | ok, |
| 20:40.29 | rtpman | do i just send audio data back to the port it's sending me from? |
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| 20:41.17 | WIMPy | The ports and IPs are negotioated in te SDP via SIp. |
| 20:42.07 | rtpman | ok thanks |
| 20:42.12 | rtpman | can asterisk be used as a RTP server only? |
| 20:42.18 | rtpman | without using sip etc? |
| 20:42.23 | rtpman | and i will just point audio there |
| 20:42.53 | WIMPy | No. What would it do with that? |
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| 21:19.57 | rtpman | anyone conisder themselves a voip engineer? i need a consoltation |
| 21:23.23 | *** join/#asterisk jab416171 (~jab416171@c-76-27-96-12.hsd1.ut.comcast.net) |
| 21:28.36 | [TK]D-Fender | * is not a "media server", a "proxy", or much else in the sense of being a "SIP server". It is a.... |
| 21:28.39 | [TK]D-Fender | ~b2bua |
| 21:28.39 | infobot | [b2bua] a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 21:29.11 | [TK]D-Fender | If anything like this is what you're looking for... you're looking at the wrong tool for the job. |
| 21:29.44 | WIMPy | Good luck finding out what the job is. |
| 21:30.09 | [TK]D-Fender | So far we are hearing only bits of aspects... with no actual end-game goal in mind. |
| 21:30.25 | [TK]D-Fender | So yeah... best of luck for whoever wants to jump through all those hoops. |
| 21:30.43 | [TK]D-Fender | packs up to head home |
| 21:47.46 | rtpman | anyone conisder themselves a voip engineer? i need a consoltation |
| 21:49.50 | rtpman | I need to play ~1000 IVRs at once to alot of customers. I want the audio stream to be handled by a different server. How do I sert something lik,e this up? |
| 21:50.58 | WIMPy | Oh, the first pieace of the puzzle. |
| 21:51.12 | WIMPy | And what is the 1st server going to do then? |
| 21:51.28 | lvlinux | rtpman: why do you want the audio stream handled by a different server? |
| 21:51.53 | WIMPy | Don't ask. Just sell something *eg* |
| 21:52.10 | lvlinux | lol |
| 21:52.10 | rtpman | 1st server is handling calls. |
| 21:52.31 | rtpman | lvlinux, because the 2nd server is to be managed by a different company (the company who provide the ivr) |
| 21:52.36 | WIMPy | What is the difference between handling calls and handlin rtp? |
| 21:52.57 | lvlinux | yeah |
| 21:52.58 | WIMPy | Now it starts to make less sense again. |
| 21:53.00 | rtpman | the ivr is for a different company to manage |
| 21:53.08 | lvlinux | what company? |
| 21:53.21 | rtpman | for example lvlinux |
| 21:53.25 | lvlinux | what company does IVR, but doesn't "handle calls" |
| 21:53.43 | rtpman | the idea of the project is to handle rtp |
| 21:53.47 | rtpman | not the calls part |
| 21:53.56 | lvlinux | why??? |
| 21:54.01 | rtpman | thats the project |
| 21:54.04 | WIMPy | One doesn't go without the other. |
| 21:54.14 | lvlinux | yup |
| 21:54.23 | lvlinux | rtpman: Why is there such a project? |
| 21:54.44 | lvlinux | Is this something you just want to do or do you have someone else giving you these specifications? |
| 21:54.53 | rtpman | this is something i want to do for the project |
| 21:55.08 | rtpman | like serve audio |
| 21:55.11 | lvlinux | ok, and WHY do you want to separate RTP from the "calls" |
| 21:55.12 | rtpman | via rtp |
| 21:55.19 | rtpman | no i dont want to seperate it |
| 21:55.24 | rtpman | i want to provide it as an example service |
| 21:55.29 | lvlinux | huh? |
| 21:55.34 | lvlinux | an example of what? |
| 21:55.40 | rtpman | its for a school project |
| 21:55.45 | rtpman | its not a real life scenario |
| 21:55.46 | WIMPy | How do you know what audio to play where without calls? |
| 21:56.02 | rtpman | thats not the problem |
| 21:56.05 | WIMPy | You urgently need a sense of the basic concepts of handling calls. |
| 21:56.07 | lvlinux | rtpman, how long have you been messing with VoIP? |
| 21:56.13 | rtpman | lvlinux, not long |
| 21:56.36 | lvlinux | ok. I'm going to put it to you straight---you cannot do what you want to do because it does not make sense. |
| 21:56.47 | rtpman | ok |
| 21:56.58 | lvlinux | Now, here's a scenario that you may find does make sense that would fit into your project: |
| 21:57.41 | lvlinux | You have one server, it handles call "routing" and "press this for that" and such, |
| 21:58.00 | lvlinux | and it receives the incoming calls from wherever you are getting them from. |
| 21:58.44 | lvlinux | Then you have another server that all it does is talk with the first server and connect calls (with RTP audio) between endpoints, based on "instructions" from the first server. |
| 21:58.54 | lvlinux | Is that sortof what you are trying to accomplish? |
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| 21:59.24 | lvlinux | [bear with me WIMPy] :-D |
| 21:59.33 | rtpman | yes sort of |
| 21:59.44 | rtpman | im just trying to type out exactly what i am trying to do now |
| 22:00.15 | WIMPy | How does one server handle press this or that and another play the menus? |
| 22:00.22 | rtpman | can i use my voip phone , dial a number , having pre-set the rtp sdp ip to x.y.x.y , and recieve the audio not on my desktop where i made the call. but receive the audio on x.y.x.y server |
| 22:00.53 | lvlinux | WIMPy: I [purposefully] left out the mention of RTP that the first server is using, which it is of course. |
| 22:01.34 | rtpman | there are no menus it is just for a demoonstration for handling audio away from the client call initiator |
| 22:01.36 | WIMPy | Doesn't make sense to me, yet. |
| 22:02.08 | rtpman | make a call and put the audio data to another server - doesnt make sense ? |
| 22:02.29 | WIMPy | What's the use case? |
| 22:02.35 | lvlinux | it does, but you don't "preset" RTP really, unless you are writing your own signalling stack. |
| 22:02.46 | WIMPy | Earlier you said you want >100 IVRs. |
| 22:02.46 | rtpman | that part is done |
| 22:02.54 | rtpman | the rtp is heading to my server now |
| 22:02.58 | WIMPy | 1000 |
| 22:03.00 | rtpman | so i already pick it up on wire shark |
| 22:03.08 | rtpman | im using an asterisk to play those IVR WIMPy |
| 22:03.13 | rtpman | this is to RECEIVE audio |
| 22:03.17 | rtpman | not to PLAY AN IVR |
| 22:03.33 | lvlinux | rtpman: SIP is designed to be agnostic of where RTP comes from or is sent. You use SIP to tell each endpoint where to send RTP and where to expect to receive it back. |
| 22:03.56 | rtpman | ok. |
| 22:04.11 | rtpman | if RTP is not received from one side, the call will be disconneced in most default cases? |
| 22:04.15 | WIMPy | So you just want something to stress-test Asterisk? |
| 22:04.47 | rtpman | for example if i just use a c# app to send a invite packet |
| 22:04.55 | lvlinux | rtpman: nope, SIP could care less about whether RTP is received or not ,that's why you have the problem with one-way audio in a lot of misconfigs |
| 22:05.03 | rtpman | initating the call , sending the acks and cfollowing the call flow |
| 22:05.22 | rtpman | now i get the RTP audio on a different server |
| 22:05.34 | rtpman | but it seems after 2-4 seconds |
| 22:05.38 | rtpman | the RTP stops being received |
| 22:05.45 | rtpman | and the remote server i invied sent me a BYE |
| 22:05.51 | rtpman | which doesn't happen when i use a normal client |
| 22:05.54 | rtpman | it is not my signalling |
| 22:05.58 | rtpman | it is something to do with RTP |
| 22:06.01 | WIMPy | Sounds more like a SIP issue. |
| 22:06.01 | lvlinux | rtpman: that is probably because you are not following the sip specs in your c# app. |
| 22:06.13 | lvlinux | it is nothing to do with RTP and everything to do with your signalling |
| 22:06.15 | rtpman | lvlinux, i followed the wire shark 100% perfectly |
| 22:06.17 | rtpman | ok. |
| 22:06.23 | rtpman | i am listening to you |
| 22:06.39 | rtpman | so if no audio data is sent to the rmeote asterisk |
| 22:06.43 | rtpman | and we just receive receive |
| 22:06.47 | rtpman | that won't cause a disconnectright ? |
| 22:06.48 | WIMPy | 2-4 seconds sonds like a timeout at the SIP level. |
| 22:06.54 | lvlinux | rtpman: here's what yo uneed to do: go study about the SIP protocol. |
| 22:07.06 | rtpman | lvlinux, its not SIP im worried about and im not even using SIP |
| 22:07.14 | rtpman | not only |
| 22:07.21 | rtpman | we have several implementations of protocols with the same problem |
| 22:07.26 | rtpman | i am asking you about RTP |
| 22:07.29 | WIMPy | I doubt anyone would set rtptimeout to less than 10s. Often much higher. |
| 22:07.34 | lvlinux | buddy if you are sending an INVITE and getting a BYE you are using SIP whether you like it or not. |
| 22:07.52 | lvlinux | You have in your mind that the RTP is the problem, and it isn't. |
| 22:08.01 | lvlinux | RTP doesn't know or care if it is ever received. |
| 22:08.16 | lvlinux | that's exactly why you need a signalling protocol like SIP |
| 22:08.18 | rtpman | so asterisk doesnt disocnnect calls based on silence/lack of rtp transmission from one party? |
| 22:08.24 | lvlinux | nope |
| 22:08.36 | WIMPy | No, but Asterisk can care. But that is not likely the cause. |
| 22:08.53 | WIMPy | Depends on configuration. |
| 22:09.05 | lvlinux | but I doubt that is his problem |
| 22:09.34 | WIMPy | So do I |
| 22:10.12 | rtpman | so there are no requirements for rtp handshakes etc |
| 22:10.19 | rtpman | if i just send signalling and request rtp media to other IP |
| 22:10.25 | rtpman | it should just be flowing normally until i disocnnect the call |
| 22:10.28 | rtpman | fif signalling is correct |
| 22:10.29 | rtpman | yeah ? |
| 22:10.29 | WIMPy | Depends on configuration. |
| 22:11.40 | lvlinux | rtpman: You can set up a purposeful one-way audio "problem" scenario where neither end gets one bit of audio, RTP just sent into the bottomless pit somewhere out on the internet, and your call will stay alive indefinitely. RTP doesn't know or care, and by default SIP doesn't know or care. You have to _TELL_ Asterisk to know or care if it isn't getting RTP. By default you can just ave a silent call. |
| 22:11.59 | *** join/#asterisk spicyramen (~Adium@216.239.45.89) |
| 22:12.17 | rtpman | thanks lvlinux |
| 22:12.45 | lvlinux | rtpman: do yourself a favor---study about SIP, and go look at pjsip SIP stack (the code API is quite simple and easy to work with). |
| 22:13.21 | lvlinux | If you feel really brave, you can implement your own SIP stack like you have already started to do, but if you do that you'll probably end up with problems. |
| 22:13.59 | lvlinux | Use a tried and true one like pjsip or sofiasip and you can accomplish what you need. |
| 22:14.16 | lvlinux | ^^^^and you can do it quicker and easier than trying to roll your own. |
| 22:14.19 | WIMPy | doesnt think there will ever be such a thing as SIP without problems. |
| 22:14.33 | rtpman | me neither |
| 22:14.40 | lvlinux | unfortunately I agree with WIMPy. |
| 22:15.02 | lvlinux | but rtpman you haven't hit any problems with SIP yet, just your code that is trying to talk to it. |
| 22:15.23 | lvlinux | SIP will absolutely do what you are wanting---it doesn't care where you want the RTP sent, it will do it. |
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| 22:52.42 | *** join/#asterisk lambda79 (~lambda79@unaffiliated/lambda79) |
| 22:53.49 | lambda79 | hi guys, about webrtc do you think this websocket browser-oriented webrtc protocol could one day replace sip in entreprise ? |
| 22:54.04 | lambda79 | in business i mean |
| 22:54.14 | WIMPy | Any crap is possible. |
| 22:54.33 | WIMPy | But someone said there is a comeback for reason. |
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| 22:54.43 | WIMPy | won't hold his breath. |
| 22:55.49 | lambda79 | it is more a toy for webdeveloper ? no |
| 22:56.17 | lambda79 | no way telco business comes from this world |
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| 23:00.01 | lambda79 | today are the webRTC offer reliable or professional ? |
| 23:10.43 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 23:17.56 | lvlinux | lambda79: do you consider Google Hangouts to be professional? |
| 23:18.39 | lvlinux | lambda79: there are many professional big $$ applications using WebRTC. Now _replacing_ SIP is another story, but WebRTC is most certainly NOT a "toy." |
| 23:30.31 | lambda79 | ok thx you |
| 23:32.18 | lambda79 | didn't know for google hangouts |
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