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| 02:27.22 | bluefirecorp | I hate to ask, but does anyone have experience with a grandstream GXP1400 ? |
| 02:27.37 | bluefirecorp | I literally can't make it connect to a sip account -- web interface doesn't even allow you to choose an account. |
| 02:27.38 | bluefirecorp | >.> |
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| 02:40.17 | bluefirecorp | Finally got it -- wow, they need much better documentation. |
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| 04:03.36 | tom_thumb | Hi |
| 04:09.43 | ChannelZ | oh hai |
| 04:13.22 | tom_thumb | ChannelZ, Would be able to help me with Superfecta. |
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| 04:40.42 | ChannelZ | Maybe, if I had any idea what that was. |
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| 05:08.37 | tom_thumb | ChannelZ, It is a module that changes the CallerID based on a lookup. |
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| 05:14.55 | ChannelZ | try #freepbx |
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| 07:42.04 | Geek-Linux | Hi Every one, i am setting meetme over IAX2 ITSPs, but cant hear voice. can you please help me what could be the possible issue ? |
| 07:43.50 | gerritfromsa | try to add a directmedia=no to force all the rtp traffic via asterisk , this often fix your natting issue |
| 07:44.15 | Geek-Linux | in my ITSP settings ? |
| 07:45.40 | gerritfromsa | in iax.conf |
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| 07:46.10 | Geek-Linux | OK let me check thanks for replying |
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| 07:55.21 | gerritfromsa | Geek-Linux, let us know if it worked OK? |
| 07:55.28 | linocisco | hi all , do u have any experience with grandstream GXP1405 callpickup and call transfer? |
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| 07:57.08 | gerritfromsa | DTMF issues ? |
| 07:57.20 | Geek-Linux | gerritfromsa: same issue, i can only hear but cant talk |
| 07:57.35 | Geek-Linux | no DTMF is OK |
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| 07:58.57 | Geek-Linux | i have basically two scenarios, one calls landed through E1's that are ok, but calls landed through IAX2 have the mentioned issue. |
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| 08:02.32 | ChannelZ | Do regular calls have bidirectional audio? |
| 08:02.54 | ChannelZ | IE meetme not involved, over the same link |
| 08:03.00 | gerritfromsa | Natting done ? port 4569 |
| 08:14.57 | gerritfromsa | Geek-Linux, send us the output of the CLI and/or wireshark trace + contents of iax.conf |
| 08:15.17 | gerritfromsa | we need to see what you see ... |
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| 09:02.53 | zafu | hi, in 11.5.1 is it a know problem that MixMonitor produces zero length files? |
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| 09:08.21 | kaldemar | zafu: it also produces non-zero length files. |
| 09:08.29 | zafu | yes |
| 09:08.47 | zafu | is it a known bug? |
| 09:10.42 | kaldemar | under what circumstances? there are bugs related to MixMonitor in the issue tracker. |
| 09:13.34 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 09:18.18 | gerritfromsa | could be sox |
| 09:19.12 | zafu | it happened when "upgrading" from debian/1.8 to debian/11.5.1 |
| 09:20.23 | gerritfromsa | MIxMonitor() depends on aspecific version of sox - the upgrade could have caused the upset |
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| 10:12.52 | dar123 | hi guyz, i am trying to sort out an issue with asterisk 12 build on centos. i get Asterisk died with code 127. |
| 10:12.54 | dar123 | cat: /var/run/asterisk/asterisk.pid: No such file or directory |
| 10:13.04 | dar123 | any tips |
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| 10:15.11 | kaldemar | dar123: start it with asterisk -vvvc |
| 10:15.28 | kaldemar | dar123: you'll see what fails on start. |
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| 10:17.08 | *** join/#asterisk RadJackson (~RadJackso@fktel.cust.jaguar-network.net) |
| 10:21.11 | RadJackson | Hi , i am developing a module for my webstore , that notify the customers when theirs orders are being shipped, before , i was using auto dial (.call files in /var/spool/asterisk/outgoing/ folder), now i'm using Dial CMD, i would like to know if there is a possibility to know if the customer number is available or not , i mean , i would like to detect if there is a ringtone when i call the |
| 10:21.11 | RadJackson | number. I've tried ${DialStatus} variable , it doesnt distinguish between a number who didnt answer or an unavailable number in both cases it returns NO-ANSWER. i've also tried CDR variables, for both cases its UNANSWERED. My purpose is to detect numbers that are not unavailable and remove 'em from my database. thanks. |
| 10:25.34 | RadJackson | anyone ? |
| 10:31.09 | *** join/#asterisk gerritfromsa (~gerritfro@8ta-229-168-77.telkomadsl.co.za) |
| 10:32.31 | gerritfromsa | where can I upload my files (pcap etc) ? |
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| 10:52.14 | *** join/#asterisk bombev (~User@73.109.Globcom.Net) |
| 10:52.18 | bombev | hello all :) |
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| 10:59.24 | bombev | can somebody help me http://pastebin.com/eukaSQYK |
| 10:59.48 | bombev | the problem is that Asterisk cant receive incoming call from trunk which is registered |
| 11:02.33 | gerritfromsa | if this is asterisk 1.6> insecure=very does not work anymore change it to insecure=port,invite |
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| 11:04.27 | kaldemar | bombev: your paste is garbage. device names missing, type=frie, "Sending fake auth rejection for device 359885092216" is not related to the call you try to make. |
| 11:05.03 | bombev | kaldemar 359885092216 is my number |
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| 11:05.09 | bombev | I am calling from |
| 11:05.43 | kaldemar | your paste is just random broken snippets here and there. |
| 11:06.23 | bombev | this type=frie is mistkae in my trunk section is friend |
| 11:10.56 | bombev | ok insecure is changed to insecure=port,invite |
| 11:11.15 | bombev | but still i am getting this |
| 11:11.16 | bombev | Sending fake auth rejection for device 359885092216 <sip:359885092216@46.19.210.14>;tag=62c196b8da557d337c47e1f84eb074f7 |
| 11:33.56 | gerritfromsa | bombev, I believe it's because you have 2 accounts , one peer and the other friend(which means its also a peer) - i suggest making it 1 friend account and that all |
| 11:35.16 | gerritfromsa | obviously its not Peer: or User: but a [context] in your sip.conf correct? |
| 11:36.16 | gerritfromsa | Im assuming [VoicePlat] ? |
| 11:37.17 | *** join/#asterisk jhlavacek (~jirka@80.215.32.144) |
| 11:38.17 | gerritfromsa | the info on the paste does not all make sense ? |
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| 14:42.21 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 14:42.21 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.6.0 (2013/10/21), 10.12.3 (2013/08/27), 1.8.24.0 (2013/10/21), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 14:42.59 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.27) |
| 14:47.57 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 14:49.28 | *** join/#asterisk monsterco (~monsterco@bas6-toronto47-1176197537.dsl.bell.ca) |
| 14:49.51 | boom^time | gerritfromsa, the closest thing I can find is ManagerEvent_OriginateResponse.html but it just gives you a list of the returned arguments, one of which is reason, but no information on reason codes. |
| 14:50.22 | monsterco | 'Morning everyone - I am trying to register a SIP device from outside via public IP to my asterisk server. When I do "sip show peer 202" I get: sip:202@10.10.9.14:8417 |
| 14:50.40 | monsterco | and I get: Addr->IP : 65.44.55.66:8417 |
| 14:50.45 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 14:51.09 | monsterco | so ofcourse I get one way audio because all the RTP is sent to local IP of "10.10.9.14" - how can I fix this? |
| 14:51.27 | [TK]D-Fender | Show the actual registration attempt |
| 14:51.37 | [TK]D-Fender | And registration has nothing to do with RTP |
| 14:51.43 | [TK]D-Fender | that is an actual call |
| 14:51.52 | [TK]D-Fender | So show the debug for those |
| 14:53.42 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:53.43 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:54.44 | gerritfromsa | boom^time, I got a file called Application_Originate.html , to view it better create a symbloic link and view it via your webbrowser ... |
| 14:55.46 | monsterco | [TK]D-Fender - here is SIP registration: http://paste.debian.net/64505/ |
| 14:56.48 | RadJackson | Hi ? |
| 14:57.03 | boom^time | gerritfromsa, this shows originate_status. What I receive on a return is success, failure, and then a reason code for the failure. Which is just an integer. |
| 14:57.36 | gerritfromsa | asterisk version ? |
| 14:57.41 | *** join/#asterisk sp00kz (~ilubj00@unaffiliated/sp00kz) |
| 14:59.09 | [TK]D-Fender | monsterco: and the call? |
| 15:00.32 | monsterco | <[TK]D-Fender> And registration has nothing to do with RTP <<< I think I saw the local IP in the call still. I will make an echo test right now and will paste it |
| 15:01.12 | *** join/#asterisk serafie (~erin@nat/digium/x-mjvilwazcffpwsnl) |
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| 15:02.51 | *** join/#asterisk ghost75 (~quassel@dslb-088-064-052-221.pools.arcor-ip.net) |
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| 15:03.08 | ghost75 | is it possible to run a application on cli for testing purpose? |
| 15:03.26 | [TK]D-Fender | A channel has to call it |
| 15:03.31 | [TK]D-Fender | So go Originate one |
| 15:03.37 | monsterco | [TD]D-Fender - here is the call sip debug : http://paste.debian.net/64506/ |
| 15:04.35 | monsterco | here is the prob: Via: SIP/2.0/UDP 10.10.9.14:8417;branch=z9hG4bK-d87543-947127738-1--d87543-;rport |
| 15:04.46 | [TK]D-Fender | monsterco: Looks fine... although you should have provided VERBOSE 10 with that |
| 15:04.52 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 15:05.26 | [TK]D-Fender | monsterco: pastebin your peer as well |
| 15:05.51 | monsterco | [TK]D-Fender - you mean sip show peers? |
| 15:06.05 | [TK]D-Fender | no, the actual config |
| 15:06.09 | monsterco | also you mean "core set verbose 10"? |
| 15:06.11 | boom^time | gerritfromsa, 11.5 |
| 15:06.15 | [TK]D-Fender | monsterco: And yes |
| 15:07.08 | gerritfromsa | boom^time, looking on the net ... |
| 15:07.38 | gerritfromsa | ghost75, yes you can use console dial ... |
| 15:08.19 | monsterco | [TK]D-Fender - here is the peer info: http://paste.debian.net/64507/ |
| 15:09.01 | monsterco | So, why do I have retransmission many times? are you saying this is an issue on the client side? |
| 15:09.16 | RadJackson | Hi , i am developing a module for my webstore , that notify the customers when theirs orders are being shipped, before , i was using auto dial (.call files in /var/spool/asterisk/outgoing/ folder), now i'm using Dial CMD, i would like to know if there is a possibility to know if the customer number is available or not , i mean , i would like to detect if there is a ringtone when i call the |
| 15:09.16 | RadJackson | number. I've tried ${DialStatus} variable , it doesnt distinguish between a number who didnt answer or an unavailable number in both cases it returns NO-ANSWER. i've also tried CDR variables, for both cases its UNANSWERED. My purpose is to detect numbers that are not unavailable and remove 'em from my database. thanks. |
| 15:10.26 | [TK]D-Fender | RadJackson: No option to detect ringtone for that purpose |
| 15:10.42 | gerritfromsa | RadJackson, depends on your ITSP |
| 15:10.45 | [TK]D-Fender | RadJackson: there is no "number not registered" signalling |
| 15:10.58 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 15:11.01 | [TK]D-Fender | that * is going to get |
| 15:11.31 | RadJackson | So there is no way to distinguish between a number who didnt answer , or a not registred number ? |
| 15:11.34 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.231) |
| 15:11.51 | *** join/#asterisk asteriskmonkey (~Tardis@74-51-38-204.telnetcommunications.com) |
| 15:12.15 | asteriskmonkey | has anyone had issues before with asterisk ivr timeout timeing in vmware? |
| 15:12.33 | asteriskmonkey | it seems ive got an odditiy where i set a timeout of 1 etc.. and its not processing for 4-5 seconds |
| 15:12.34 | [TK]D-Fender | asteriskmonkey: VIR knows nothing about vmware.... |
| 15:12.39 | [TK]D-Fender | asteriskmonkey: You should know better on this front |
| 15:12.48 | [TK]D-Fender | asteriskmonkey: If it's voip, then packets are packets. |
| 15:12.58 | asteriskmonkey | yeah but timing is timing |
| 15:13.05 | asteriskmonkey | not a packet issue :/ |
| 15:13.07 | [TK]D-Fender | asteriskmonkey: Well if you massacre it that bad :p |
| 15:13.12 | asteriskmonkey | seems to be a timing isue |
| 15:13.28 | asteriskmonkey | just wondering is anyone else has reported that wierdness before in vm's |
| 15:13.55 | [TK]D-Fender | monsterco: new call with verbose please.... |
| 15:14.39 | gerritfromsa | RadJackson, what worked for me was |
| 15:14.51 | *** join/#asterisk gtjoseph (~gtj@unaffiliated/gtj) |
| 15:15.21 | gerritfromsa | ${EXTENSION_STATE} - I used this prior to Dia() |
| 15:15.28 | gerritfromsa | ${EXTENSION_STATE} - I used this prior to Dial() |
| 15:15.38 | boom^time | gerritfromsa, did you find out any details? |
| 15:15.54 | gerritfromsa | bombev, no luck yet |
| 15:16.01 | gerritfromsa | boom^time, no luck yet |
| 15:16.11 | boom^time | Yeah me neither. |
| 15:16.25 | gerritfromsa | open source documentation ? |
| 15:16.31 | boom^time | nope |
| 15:16.48 | monsterco | [TK]D-Fender - ok - coming in 1 minute |
| 15:17.13 | gerritfromsa | anyone have an idea what code 3 response from sipsak means ? |
| 15:17.40 | gerritfromsa | Im gonna wireshark it ... |
| 15:19.07 | gerritfromsa | Port closed ... |
| 15:19.40 | ghost75 | [16:05:27] <gerritfromsa> ghost75, yes you can use console dial ... <- is that not working in 1.8 ? |
| 15:20.02 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:20.07 | gerritfromsa | its works in 1.8 |
| 15:20.32 | ghost75 | then maybe i have it not compiled |
| 15:21.12 | ghost75 | looks like oss/alsa is needed |
| 15:21.22 | gerritfromsa | need a rtp engine |
| 15:21.45 | RadJackson | gerritfromsa i've tried to display ${EXTENSION_STATE} before and after Dial() using NoOp() , it is empty... |
| 15:22.15 | gerritfromsa | RadJackson, did you add @context ? |
| 15:22.41 | boom^time | [TK]D-Fender, sorry to bug you, but since you're an * dev I was hoping you could point me to the right source file for discovering what each OriginateReponse Failure Reason code means. I must be grepping for the wrong thing. |
| 15:23.06 | gerritfromsa | ${EXTENSION_STATE(${EXTEN}@outbound)} |
| 15:24.06 | *** join/#asterisk Pullphinger (~Pullphing@12.40.23.68) |
| 15:24.20 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 15:25.18 | [TK]D-Fender | that is not going to do much... |
| 15:26.19 | gerritfromsa | works for me |
| 15:27.15 | [TK]D-Fender | gerritfromsa: that won't get you a phone # availability |
| 15:28.00 | RadJackson | gerritfromsa I did that before and after the Dial CMD |
| 15:28.01 | RadJackson | http://pastebin.com/K9m9guwA |
| 15:28.11 | RadJackson | For Both numbers , and in both places , it displays Unknown |
| 15:28.26 | [TK]D-Fender | RadJackson: This will not work. |
| 15:29.12 | [TK]D-Fender | RadJackson: That looks for a HINT created against 1234 and that will in no way validate a number off a remote system |
| 15:29.53 | gerritfromsa | my service provider gives me either a RINGING or INVALID |
| 15:29.54 | gerritfromsa | this wau I know if the number exists or not |
| 15:30.42 | [TK]D-Fender | This is not all appropriate |
| 15:30.52 | [TK]D-Fender | that function only checks a fixed hint |
| 15:31.01 | [TK]D-Fender | And never contacts a device either way |
| 15:31.25 | [TK]D-Fender | States are not polled, Asterisk only tracks based on calls it makes itself. |
| 15:31.37 | gerritfromsa | dont have hints |
| 15:32.12 | [TK]D-Fender | core show function EXTENSION_STATE <- read the instructions |
| 15:35.00 | RadJackson | When you place a call file on the outgoing folder , and thru the CLI , asterisk is able to distinguish if its ringing or not. So there must be a way to check if the remote device is ringing or not , no? |
| 15:37.06 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 15:37.40 | [TK]D-Fender | RadJackson: * will not report it BACK in an way you can use |
| 15:37.47 | navaismo | sounds like an autodialer to annoy, i mean to validate phones |
| 15:39.43 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:74db:a6f0:b82b:df70) |
| 15:42.46 | *** join/#asterisk navaismo (~navaismo@189.191.206.207) |
| 15:44.07 | ghost75 | is there any mistake in this line: exten => _X.,n,SendFAX(/tmp/faxconvert-${EXTEN}.tiff,d) |
| 15:44.50 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 15:46.00 | [TK]D-Fender | ghost75: What happens when you use it? |
| 15:46.43 | ghost75 | nothing, its not showing up in verbose log, that is what boggles me |
| 15:47.34 | [TK]D-Fender | Then it isn't being executed... |
| 15:47.48 | [TK]D-Fender | So Where do we see the rest of the call that should be using that line? |
| 15:48.12 | [TK]D-Fender | And the rest of the dialplan around it? |
| 15:48.31 | ghost75 | http://pastebin.com/jD5eQNTu |
| 15:48.45 | ghost75 | its doing everything except the sendfax |
| 15:49.33 | [TK]D-Fender | show us the call |
| 15:49.35 | ghost75 | maybe the application is broken |
| 15:49.51 | [TK]D-Fender | ghost75: From what I see, NONE of that should work at all |
| 15:49.58 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 15:53.40 | ghost75 | http://pastebin.com/7gA6bu7s |
| 15:54.17 | [TK]D-Fender | ghost75: that does not match the dialplan you showed us at all |
| 15:54.34 | ghost75 | why |
| 15:54.45 | [TK]D-Fender | you tell me |
| 15:54.49 | [TK]D-Fender | You showed that code |
| 15:54.55 | [TK]D-Fender | that is NOT what is getting executed at all |
| 15:54.57 | [TK]D-Fender | et("SIP/arcor_out998774-00000048", "FAXOPT(localstationid)=08192998774") in new stack |
| 15:55.11 | [TK]D-Fender | That line sure isn't in the dialplan you showed |
| 15:55.30 | [TK]D-Fender | And as I said, what you showed was by itself, broken. |
| 15:55.33 | [TK]D-Fender | You need to look at what you're doing... |
| 15:56.02 | ghost75 | why in the log everything shows up except the sendfax application |
| 15:56.08 | [TK]D-Fender | ghost75: No. |
| 15:56.17 | [TK]D-Fender | ghost75: look at the APPLICATIONS it is call |
| 15:56.20 | [TK]D-Fender | ing* |
| 15:56.45 | [TK]D-Fender | http://pastebin.com/7gA6bu7s <- look at all the crap it is setting for FAXOPT. |
| 15:56.55 | [TK]D-Fender | http://pastebin.com/jD5eQNTu <-- where do you see that in here? |
| 15:56.59 | [TK]D-Fender | This is NOT the same at all |
| 15:58.04 | ghost75 | http://pastebin.com/jD5eQNTu <- i just didnt put set options in pastebin because it is not relevant |
| 15:58.12 | [TK]D-Fender | ghost75: You are showing junk |
| 15:58.25 | ghost75 | no its just 2 missing lines |
| 15:58.34 | [TK]D-Fender | pastebin actual dialplan |
| 15:58.41 | [TK]D-Fender | Not fractional bits |
| 15:58.50 | [TK]D-Fender | We have no reason to trust what you show us |
| 15:59.36 | ghost75 | http://pastebin.com/MsJ5ynnr |
| 15:59.39 | [TK]D-Fender | [Nov 7 15:08:28] VERBOSE[23773] pbx.c: -- Executing [08195998718@faxoutgoing:6] Wait("SIP/arcor_out998774-00000048", "60") in new stack |
| 15:59.40 | [TK]D-Fender | <PROTECTED> |
| 15:59.44 | ghost75 | that is just 2 lines more |
| 15:59.57 | [TK]D-Fender | And this answers why it isn't happening all by itself |
| 16:01.53 | [TK]D-Fender | [Nov 7 15:08:28] ERROR[23773] utils.c: write() returned error: Broken pipe <- while you're at it, your AGI is broken..... |
| 16:01.57 | ghost75 | is the wait command a problem or what? |
| 16:04.36 | [TK]D-Fender | ghost75: What do you see right there? |
| 16:05.07 | ghost75 | a wait and exit non zero |
| 16:05.21 | [TK]D-Fender | ghost75: And what about it? |
| 16:05.32 | ghost75 | tell me |
| 16:05.40 | [TK]D-Fender | ... |
| 16:06.15 | [TK]D-Fender | [Nov 7 15:08:28] ----> [Nov 7 15:09:18] Does this look like it survived 60 seconds? |
| 16:06.22 | [TK]D-Fender | It didn't |
| 16:06.37 | [TK]D-Fender | The other end killed the call |
| 16:06.46 | [TK]D-Fender | Noone wants to sit around waiting like that |
| 16:07.44 | ghost75 | so the dialplan doent hold after wait and continues? |
| 16:08.18 | [TK]D-Fender | the call DIED |
| 16:08.21 | [TK]D-Fender | They hung up |
| 16:08.28 | [TK]D-Fender | There is no more "dialplan" |
| 16:08.44 | *** join/#asterisk u0m3 (~u0m3@109.96.175.232) |
| 16:08.58 | [TK]D-Fender | Off to "h" you go. |
| 16:09.12 | ghost75 | i thought that the call starts with "sendfax" |
| 16:09.35 | [TK]D-Fender | You ORIGINATED a call. |
| 16:09.56 | [TK]D-Fender | Your dialplan started after they ANSWERED |
| 16:09.57 | ghost75 | let me try with removing this wait |
| 16:10.31 | *** join/#asterisk Pullphinger (~Pullphing@12.40.23.68) |
| 16:10.38 | ghost75 | so kind makes sense what you say |
| 16:11.28 | [TK]D-Fender | You phone out and make them wait 60 seconds and the other side say "forget it, I give up" and hangs up |
| 16:11.42 | [TK]D-Fender | It was all right there |
| 16:11.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.153) |
| 16:12.12 | ghost75 | so i was not aware about the originate actually already starting the call |
| 16:13.10 | [TK]D-Fender | ghost75: that is the entire definition of what it does |
| 16:13.25 | [TK]D-Fender | You don't dial out and start processing stuff right away |
| 16:15.58 | ghost75 | thx now it shows up |
| 16:17.15 | [TK]D-Fender | not "shows up". |
| 16:17.33 | [TK]D-Fender | "Other side doesn't get tired of waiting and hangs up" |
| 16:17.42 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 16:17.56 | ghost75 | i mean it shows up now in log |
| 16:18.18 | ghost75 | Status: SUCCESS Error: HANGUP |
| 16:19.42 | [TK]D-Fender | It isn't alogging issue or "shoing up" |
| 16:19.46 | [TK]D-Fender | the call DIED |
| 16:19.52 | [TK]D-Fender | it never made it that far |
| 16:20.07 | ghost75 | yes i know |
| 16:20.11 | [TK]D-Fender | Now at least you aren't making them wait so long that they hang up on you |
| 16:23.18 | *** join/#asterisk cmendes0101 (~cmendes01@pool-173-58-47-18.lsanca.fios.verizon.net) |
| 16:25.37 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 16:27.56 | ghost75 | does it make sense to lower transfer rate to make fax more reliable on g711 ? |
| 16:28.33 | Chainsaw | ghost75: Yes. 2400 bps has a better chance. |
| 16:28.40 | Chainsaw | ghost75: I hope you don't often send dozen of pages though... |
| 16:28.51 | ghost75 | nope |
| 16:29.10 | ghost75 | as long the remote side doesnt hang up because too slow lol |
| 16:29.35 | Chainsaw | It's well within spec. It will think you're faxing from the seventies, but it'll cope. |
| 16:29.55 | ghost75 | is it set via faxopt ? |
| 16:32.47 | TrevorH1 | I tried to use an extconfig.conf file with two lines of different priorities last night and got a segfault. did I do something wrong? extconfig like http://pastebin.centos.org/5576/ |
| 16:34.08 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
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| 16:47.50 | bitkill3r | Hi everyone, i have an asterisk (using trixbox) that's facing the internet, it's being secured by a firewall, but sip ports and rtp ports are open. It also has allow anonymous inboind sip calls set to yes. I know this might not be the best way to have it setup. But is there a way that I can prevent from getting these [501 "501" <501> s ANSWERED00:00? |
| 16:52.50 | navaismo | first drop that thing called trixsucks then set the allowguest to no |
| 16:54.00 | ghost75 | and alwaysauthreject=yes |
| 16:56.01 | *** join/#asterisk mitchrodrigues (~mitchrodr@204-16-155-82-static.ipnetworksinc.net) |
| 16:56.10 | bitkill3r | I am, actually testing other options at the moment |
| 16:56.58 | ghost75 | V.17 V.27 V.29 which one is the most reliable? |
| 16:57.19 | pabelanger | depends on a lot of factors |
| 16:57.52 | ghost75 | when g.711 with compression comes into play |
| 16:58.39 | pabelanger | If you are trying to do t.30 over IP, you are in for some pain |
| 16:58.45 | pabelanger | just avoid it and do T.38 |
| 16:58.57 | ghost75 | i cant :< |
| 16:59.01 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:59.10 | pabelanger | why |
| 16:59.22 | ghost75 | isp dont has it |
| 16:59.58 | pabelanger | find a company that does |
| 17:00.52 | ghost75 | the biggest provider doesnt even know if they have it |
| 17:02.52 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 17:07.18 | TrevorH1 | we use mysql realtime with a pair of multimaster mysql machines but when one of them is down I'd like asterisk to use the other one to send its updates. Is this possible? Asterisk 1.8 and already using readdb/writedb |
| 17:11.38 | bitkill3r | just tried what you guys suggested. I've been bombarded with those today. I'll keep an eye on it and report back if anything.. Thanks! |
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| 18:54.57 | zoid_ | anybody knows if asterisk spooler is single-threaded? |
| 19:00.23 | *** join/#asterisk v0lZy (~Thunderbi@84-255-194-41.static.t-2.net) |
| 19:01.11 | *** join/#asterisk HolTech (~IceChat77@pool-108-40-37-180.bltmmd.fios.verizon.net) |
| 19:01.50 | HolTech | is there a way via cli to set a user as having voicemail? would just creating a file in the user's mailbox folder as defined in voicemail.conf do the trick ? |
| 19:02.27 | [TK]D-Fender | voicemail.conf is what defines mailboxes |
| 19:02.34 | [TK]D-Fender | there is no such thing as a "user" really |
| 19:03.10 | HolTech | right, so if i stick a sound file in that mailbox folder, would it trigger the "hasvoicemail" tone so the phone lights up ? |
| 19:03.15 | Penguin | They should change "voicemail show users" then. |
| 19:03.23 | [TK]D-Fender | Or are you trying to trick it into sending MWI based on a mailbox that is actually defined? |
| 19:03.32 | [TK]D-Fender | Guess so |
| 19:03.46 | HolTech | not sure what MWI is, but I think so |
| 19:03.54 | Penguin | ~mwi |
| 19:03.54 | infobot | Message Waiting Indicator |
| 19:04.38 | HolTech | right |
| 19:04.51 | HolTech | then yes, that is exactly what I am trying to do. |
| 19:06.28 | Penguin | I'm considering trying your idea of putting a file in there just to see what happens. |
| 19:08.16 | Penguin | touching a file in the INBOX does not trigger it. |
| 19:09.18 | Penguin | But if you simulated the voice mail by creating a real voice mail message file, I bet it would work. |
| 19:09.30 | Penguin | The txt, not the audio file. |
| 19:11.15 | HolTech | conversely removing an unlisteded to file does not stop the MWI |
| 19:11.22 | HolTech | so where is this controlled in * ? |
| 19:11.44 | Penguin | If you delete the files from the INBOX, the mwi should turn off. |
| 19:11.55 | HolTech | I just tried and it did not |
| 19:12.04 | HolTech | v 1.8.18.0 |
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| 19:15.37 | [TK]D-Fender | HolTech: What is your goal with this? |
| 19:15.48 | [TK]D-Fender | HolTech: What are you trying to indicate? |
| 19:16.32 | HolTech | Looking to be able to add a voicemail for a user without calling the extension... |
| 19:16.53 | Penguin | I see the problem. You think extension means phone. |
| 19:16.54 | HolTech | to be able to trigger that MWI |
| 19:17.38 | Penguin | You should make an extension that does not Dial the device, but does run VoiceMail(). |
| 19:17.53 | HolTech | kindof, I recognize the extension is not the phone, and one user can have mulitiple extensions |
| 19:18.07 | Penguin | Then you can call the extension, leave voice mail, and then the light will come on. |
| 19:19.01 | [TK]D-Fender | HolTech: Why don't you just call VoiceMail()? |
| 19:19.02 | HolTech | That would work, I was hoping to do this outside of asterisk if possible. like creating files in the voicemail folder from some other application. |
| 19:19.21 | Penguin | How are you going to create the voice mail message? |
| 19:19.23 | HolTech | I guess that is what I will do. |
| 19:19.23 | *** join/#asterisk hardwire (~hardwire@222-83-237-24.gci.net) |
| 19:19.23 | [TK]D-Fender | HolTech: Ok, so the source of the audio is something external? |
| 19:19.37 | HolTech | right |
| 19:20.19 | [TK]D-Fender | Ok, that does get trickier. Guess you'd have to deposit the file & txt |
| 19:21.05 | HolTech | I will try that, but i would presume that because removing those files doesn't trigger the mwi to stop , that adding them would not trigger it on. |
| 19:21.28 | Penguin | I think it will trigger it. |
| 19:21.43 | HolTech | i will try and let you know. |
| 19:22.31 | Penguin | But you need to create the files, have them on the same file system, and then mv them into the INBOX. Just like call files need to be in place instantly, I have to assume the voice mail does as well since it is in the spool dir. |
| 19:23.17 | Penguin | I think it's the txt file that triggers the mwi. |
| 19:24.31 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:24.52 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:25.24 | Penguin | To test it, I might grab someone else's txt file from their INBOX, edit it accordingly for the testing mail box, then mv it into place. |
| 19:25.25 | HolTech | Did not work. Then doing a core shut down and start the light came on. |
| 19:25.28 | *** join/#asterisk dannyboy-work (~dan@64-60-100-58.static-ip.telepacific.net) |
| 19:26.13 | HolTech | * must poll that at some point, it would be nice if there was some way to kick that off. I will keep digging |
| 19:28.42 | Penguin | I tested it, too, and it did not trigger MWI. |
| 19:29.13 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 19:30.56 | Penguin | That's odd. I've never tried to see what triggered it. I just always assumed that since it was in the spool dir, asterisk was watching INBOXes for new files. |
| 19:31.46 | HolTech | that would be nice but I guess that would be resource consuming. reloading the dial plan did not trigger it |
| 19:32.16 | Penguin | It wouldn't. |
| 19:32.19 | Penguin | But reloading voicemail might. |
| 19:32.58 | Penguin | And if THAT doesn't trigger it, unload the voicemail module and load it back. |
| 19:33.38 | HolTech | voicemail reload , that command reloads config but does not trigger it |
| 19:33.51 | HolTech | whats command to unload and load module/ |
| 19:34.10 | Penguin | module unload app_voicemail.so |
| 19:34.16 | Penguin | module load app_voicemail.so |
| 19:35.01 | HolTech | that did work. Certainly not an elegant way, but good to know it does refresh |
| 19:35.31 | Penguin | That almost makes me thing the VoiceMail app is the only thing that triggers it. |
| 19:35.50 | Penguin | s/thing/think/ |
| 19:36.19 | *** join/#asterisk tears (~htc@host109-154-99-55.range109-154.btcentralplus.com) |
| 19:36.26 | tears | hello |
| 19:36.52 | tears | I would like to configure an astersik server to 'redirect' calls from a sip number to a mobile number |
| 19:36.57 | tears | it's possible? |
| 19:37.20 | Penguin | Do you really need to redirect the calls, or can you accept the call on asterisk and have asterisk call the mobile? |
| 19:37.24 | [TK]D-Fender | tears: there is no such thing as a "sip number" and "mobile number" is similarly vague |
| 19:37.33 | tears | voip to gsm |
| 19:37.42 | [TK]D-Fender | tears: Asterisk doesn't speak GSM |
| 19:37.55 | [TK]D-Fender | tears: special hardware you might by for it might. |
| 19:38.07 | tears | I have a raspberry pi running raspbx |
| 19:38.21 | tears | with an usb modem |
| 19:38.29 | [TK]D-Fender | tears: How do you intend to call that "cell number"? |
| 19:38.53 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:38.53 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:39.12 | tears | well, when someone calls me on the number from my sip account, the rpi calls me on my mobile number and .. |
| 19:39.23 | tears | transfer the audio from one to another maybe? |
| 19:39.38 | [TK]D-Fender | tears: Again, what are you using to call out? |
| 19:39.40 | tears | so both calls are active in the same time |
| 19:39.47 | tears | the usb modem |
| 19:39.52 | tears | voice activated |
| 19:40.00 | [TK]D-Fender | tears: And that is supported by Asterisk? |
| 19:40.04 | tears | yes it is |
| 19:40.05 | [TK]D-Fender | tears: Almost none are |
| 19:40.14 | [TK]D-Fender | tears: What model? |
| 19:40.21 | tears | huawei e180 |
| 19:40.40 | tears | I had to 'hack' it to activate the voice feature |
| 19:40.46 | elguero | HolTech: Did you look at the pollmailboxes option for voicemail? |
| 19:40.49 | tears | I tested it on windows xp and it works |
| 19:41.02 | Penguin | elguero: Tell us more about that. |
| 19:41.16 | [TK]D-Fender | Windows has nothing to do with Asterisk support |
| 19:41.25 | elguero | Penguin: It is documented in voicemail.conf |
| 19:41.29 | tears | raspbx supports it |
| 19:41.30 | [TK]D-Fender | http://code.google.com/p/datacard/wiki/Supported_devices |
| 19:41.48 | tears | somebody else did this but I can't find a tutorial on how to do it myself |
| 19:41.53 | tears | that's why I'm here |
| 19:41.55 | [TK]D-Fender | tears: Well if you have a channel driver that is functional with it... take your call in, Dial() out. The end. |
| 19:42.01 | Penguin | elguero: Very nice. I never looked for that. I'm sure this is _exactly_ what holtech needs. |
| 19:42.02 | elguero | Penguin: it is supposed to be for cases where mailboxes are changed outside of app_voicemail |
| 19:42.09 | Penguin | elguero: I see that. |
| 19:42.40 | HolTech | elguero: i have not, looking into it now. |
| 19:42.58 | Penguin | It looks like you basically need to set pollmailboxes=yes and if the default polling time of 30 intervals is good enough, save the file and run voicemail reload. |
| 19:43.07 | [TK]D-Fender | tears: Seems chan_dongle supports it... |
| 19:43.10 | Penguin | 30 sec intervales |
| 19:43.13 | tears | yes |
| 19:43.15 | tears | exactly |
| 19:43.16 | [TK]D-Fender | http://wiki.e1550.mobi/doku.php?id=requirements |
| 19:43.20 | HolTech | awesome |
| 19:43.34 | tears | e180 + + + ok by ramoncio |
| 19:43.45 | tears | but I dunno where to start.. |
| 19:44.03 | Penguin | Yeah, I should have looked in voicemail.conf when you asked about voice mail. That's my failure for the day. I have zero fails left. |
| 19:44.03 | tears | with the configuration I mean.. |
| 19:44.20 | [TK]D-Fender | tears: http://wiki.e1550.mobi/doku.php?id=usage |
| 19:44.23 | tears | it has freepbx installed also |
| 19:44.24 | [TK]D-Fender | tears: chan_dongle.conf |
| 19:44.56 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 19:45.06 | [TK]D-Fender | http://wiki.e1550.mobi/doku.php?id=configuration |
| 19:45.36 | tears | that's auto configured when I install the usb modem |
| 19:45.38 | HolTech | Just tested, that worked ! thanks elguero, and penguin, the default is 30 but it looks like the line pollfreq=XX will let you set it |
| 19:45.42 | tears | with the command instal-dongle |
| 19:45.59 | [TK]D-Fender | tears: We can't speak for what that custom-baked distro does... |
| 19:46.04 | tears | what I want to know is how I set the inbound and outbound rules |
| 19:46.11 | [TK]D-Fender | tears: We don't support it or the GUI it includes. |
| 19:46.14 | elguero | HolTech: Great! Yes, you should be able to set the frequency with that option. |
| 19:46.19 | tears | oh |
| 19:46.23 | [TK]D-Fender | tears: But we can at least comment on the * related bits |
| 19:47.03 | tears | http://www.raspberrypi.org/archives/3679 |
| 19:47.10 | tears | this is what I want |
| 19:47.11 | [TK]D-Fender | tears: the sample config page there shows you the format Dial() uses... so make a custom trunk to do that |
| 19:47.13 | tears | but just end |
| 19:47.38 | tears | dongle/dongle0/$OUTNUM$ |
| 19:47.40 | tears | i think |
| 19:48.30 | tears | yes |
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| 19:48.36 | tears | that's where I need to change it ? |
| 19:48.43 | [TK]D-Fender | for that part, yes |
| 19:49.52 | tears | so it will work |
| 19:50.04 | tears | like a bridge between voip and gsm |
| 19:50.17 | [TK]D-Fender | ~b2bua |
| 19:50.17 | infobot | i heard b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 19:50.22 | [TK]D-Fender | That is what * is |
| 19:50.36 | [TK]D-Fender | you have 2 ends that are functional, * processes your calls as you configure it to |
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| 19:50.40 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:51.01 | jblack | [TK]D-Fender: You came up in a borderline random conversation in meatspace the other day. Someone said very good things about ya |
| 19:51.58 | [TK]D-Fender | jblack: cool I guess... never heard of "meatspace" ... and wonder what you mean by "borderline"\ |
| 19:53.26 | jblack | meatspace = in real life. borderline = from network administration to a tangent on asterisk to you |
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| 19:57.20 | [TK]D-Fender | Ah, who was this with? |
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| 20:08.05 | *** part/#asterisk HolTech (~IceChat77@pool-108-40-37-180.bltmmd.fios.verizon.net) |
| 20:08.43 | iulhk | Server Info: Hp Proliant DL380 G7, E5605 cpu / 4 cores, 32gb ram, OS centos6.2, how many live radio stream i can get at this server and how many concurrent calls can be connected ? any idea or how do i measure these things? |
| 20:09.10 | navaismo | sipp |
| 20:12.30 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 20:13.11 | danfromuk | Hi, How does UniqueID work? IE what format is it in? I'm trying to match up the incoming call leg with the outgoing call leg using data provided by the AMI |
| 20:13.40 | danfromuk | The only thing I can see is that the beginning of the UniqueID is the same for both legs, but after the '.' its different. |
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| 20:19.25 | *** join/#asterisk saxa (~sasa@84-255-205-164.static.t-2.net) |
| 20:20.09 | saxa | hi I'm trying to configure my te133 card to work with asterisk and openr2, compilled all stuff and seems everything in place. |
| 20:20.11 | tears | "the number you have dialed is not in service" |
| 20:20.14 | tears | so it's working |
| 20:20.35 | navaismo | danfromuk: for me is an epoch number |
| 20:21.06 | saxa | now the thing is when I load the driver this is what I get: http://pastebin.com/SUk5UQ45 |
| 20:21.24 | danfromuk | navaismo: yep. youre right. Back to the drawing board. |
| 20:21.58 | saxa | Running dahdi_cfg: |
| 20:21.58 | saxa | Something went wrong. Try running dahdi_cfg -vvv. |
| 20:23.08 | saxa | DAHDI_CHANCONFIG failed on channel 16: Invalid argument (22) |
| 20:23.19 | saxa | ok it seems that I have to rebuild dahdi |
| 20:23.21 | saxa | thx |
| 20:23.28 | saxa | sorry for spamming |
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| 20:29.25 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:21b:63ff:fe31:8426) |
| 20:29.29 | gusto | i have run into problems |
| 20:30.23 | gusto | i have two asterisks, one is locally bridging and the other is remote bridging, now when i hang up, the locally bridging one is noticing it, but the remotely bridging is getting a timeout |
| 20:32.47 | gusto | http://ur1.ca/fzyqa |
| 20:34.22 | [TK]D-Fender | gusto: that doesn't help as you're not actually looking at the SIP debug |
| 20:35.33 | gusto | [TK]D-Fender: OK, i am going to make a few tests now, and then i am going to enable sip debuging |
| 20:38.26 | gusto | i may have found the reason, i had externaddr instead of externhost for a DNS name |
| 20:38.35 | gusto | maybe that was the problem, i am going to try that first |
| 20:40.06 | gusto | no, it was not the case and it is even worse, with local bridging i am not even getting a RTP stream |
| 20:40.15 | gusto | and the problem is still the same |
| 20:40.45 | Penguin | If the one you are using for local bridging never touches the nat, configure it for no nat. |
| 20:41.50 | Penguin | If it never communicates with anything on the outside of your nat on the public internet, configure it for nat=no, directmedia=yes, no externhost/externaddr, and no localnet. |
| 20:45.40 | gusto | interesting approach |
| 20:45.43 | gusto | but may work |
| 20:47.56 | gusto | eh, does not help |
| 20:48.08 | gusto | i have to turn on sip debug to see what is going on |
| 20:48.36 | Penguin | That was suggested 15 minutes ago. |
| 20:48.41 | gusto | LOL, now even the other one is complaining about: |
| 20:48.43 | gusto | [Nov 7 20:47:48] NOTICE[17406]: chan_sip.c:22976 handle_request_invite: Unable to create/find SIP channel for this INVITE |
| 20:49.15 | gusto | i know, but i wanted to do some testing first to rule out some other possibilities, for example that externhost misconfiguration, because this sip debug takes far longer |
| 20:51.08 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.140) |
| 20:51.12 | gusto | hm ... i am starting to think that the problem is on my new ATA adapter |
| 20:51.35 | Penguin | ~ata |
| 20:51.36 | infobot | hmm... ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 20:51.37 | gusto | i bought a new SPA112 and that one may be misconfigured |
| 20:51.42 | gusto | yes |
| 20:51.54 | *** join/#asterisk jpoz (~jpoz@184.169.152.1) |
| 20:51.57 | gusto | because now i turned on that SIP debug and i see similar messages on both of them |
| 20:51.59 | Penguin | ATA adapter = analog terminal adapter adapter |
| 20:52.17 | gusto | so i am beginning to suspect a third party being the problem |
| 20:52.45 | Penguin | echo test |
| 20:55.14 | anonymouz666 | adds adapter adapter to my list of redundancy |
| 20:55.26 | anonymouz666 | he he |
| 20:56.13 | Penguin | redundancy list of redundancy |
| 20:58.16 | [TK]D-Fender | tears: inbound route to match your inbound call -> Custom Destination (1234567890,from-internal,1), and make an outbound route that can match that #. |
| 20:59.43 | gusto | yes, that echo test may be a good way how to test this |
| 21:00.02 | gusto | however, i searched the sip debug for problems and i may have found one |
| 21:00.31 | gusto | he is trying to sell him the internal IPv4 address of his jail as the address on which he waits for connection |
| 21:01.59 | gusto | http://ur1.ca/fzyu8 |
| 21:02.28 | Penguin | Is that the asterisk that talks to the internet? |
| 21:03.53 | gusto | 192.168.10.3 is the address that makes me worry |
| 21:03.58 | gusto | yes, that is the outside asterisk |
| 21:04.09 | Penguin | Then you need to configure it to work behind nat. |
| 21:04.22 | Penguin | nat=yes, directmedia=no, set externhost/externaddr, and localnet. |
| 21:05.03 | gusto | however, that makes no sense |
| 21:05.22 | Penguin | Sure it does. If the asterisk is behind a NAT (has a private address) and talks to things on the public internet, it must be configured for nat. |
| 21:05.46 | gusto | yes, i ll do that in a moment, but that is the communitation between the outside asterisk and the VoIP provider, so I do not see where there could be a problem, because generally I can make calls, so there is no problem |
| 21:05.58 | Penguin | Both criteria must be met. It has to be behind a nat AND talk to things on the outside of the nat. |
| 21:06.23 | Penguin | If it is behind a nat but never talks to things on the public internet, it does not need to be configured for nat. |
| 21:06.47 | gusto | localnet |
| 21:06.54 | gusto | that may be the solution, i did not set localnet |
| 21:07.01 | Penguin | If it talks to things on the public internet but is not behind nat, it does not need to be configured for nat. |
| 21:08.09 | Penguin | You said you have two asterisks. Both are behind nat. One talks to the public internet -- it must be configured to work behind nat. The other does not talk to things on the public internet -- it does not need to be configured to work behind nat. |
| 21:08.13 | Penguin | Very simple. |
| 21:08.45 | gusto | hey |
| 21:08.51 | gusto | it seems to work |
| 21:08.54 | gusto | LOL |
| 21:08.59 | gusto | it was LOCALNET!!! missing |
| 21:09.12 | Penguin | (1504.08) <Penguin> Then you need to configure it to work behind nat. |
| 21:09.12 | Penguin | (1504.20) <Penguin> nat=yes, directmedia=no, set externhost/externaddr, and localnet. |
| 21:09.13 | gusto | i am going to try that once again, but i think we solved it |
| 21:09.37 | gusto | lol, i know, but i had it before and it did not help, because i forgot localnet |
| 21:10.04 | gusto | i did think that externhost does replace all addresses, but now i know that it replaces only the localnet ones |
| 21:10.12 | gusto | that's why NAT config did not help |
| 21:10.40 | Penguin | NAT config = localnet, externhost or externaddr, directmedia=no, and nat=yes |
| 21:10.52 | Penguin | If you don't set all of those correctly, it isn't configured to work behind NAT. |
| 21:10.53 | gusto | OK, it works |
| 21:11.11 | gusto | well, nat- i do not need |
| 21:11.26 | gusto | and directmedia is set to yes, because i want it to, and it works now |
| 21:11.46 | gusto | and i could try directmedia=no too, let's see if that works as well |
| 21:12.03 | gusto | that's only for trying to route RTP packts through, nothing more |
| 21:12.52 | Penguin | If you set directmedia=yes, the call can be reinvited and RTP will go around that asterisk. |
| 21:13.27 | gusto | OK, that seems to work too |
| 21:13.32 | Penguin | directmedia=no forces it to go through that asterisk. And since the devices are behind a NAT, you most certainly don't want reinviting. |
| 21:13.51 | gusto | yes, and that is what i want, and now i tried w/o directmedia and yes, it also works with RTP |
| 21:14.15 | gusto | but this asterisk is on the net, so it does not need to have RTP routed over it, it should only do SIP |
| 21:14.28 | Penguin | But the other asterisk, the one that does local bridging only, can use directmedia=yes since it does not talk to things on both sides of a NAT. |
| 21:14.44 | gusto | and that's why i have the second localy bridging asterisk on my router, that is doing w/o directmedia |
| 21:15.28 | Penguin | It's apparent to me that you either have no idea how directmedia works with asterisk and nat, or you simply are not being clear with your descriptions and goals. |
| 21:16.05 | gusto | and i think that you got it the other way round, but basically right what i have done here |
| 21:16.13 | gusto | but i can explain if you wish |
| 21:16.24 | gusto | or better, i put it in a paste, w8 |
| 21:17.00 | Penguin | Either way, I'm not going to keep wasting my bandwidth trying to help until after you rectify whichever one of those is true. |
| 21:17.06 | *** join/#asterisk jsjc (~Adium@250.219.221.87.dynamic.jazztel.es) |
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| 21:18.32 | gusto | wait, i have a nice picture for you in a moment |
| 21:22.05 | gusto | Penguin: http://ur1.ca/fzyx0 |
| 21:23.22 | Penguin | That's not clear to me, either. |
| 21:23.42 | gusto | ok |
| 21:24.08 | Penguin | I only care about your device and your two asterisk systems. |
| 21:24.11 | gusto | the reason i have that second directmedia asterisk out there is because i do not want to have that extra distance for the RTP packets to travve |
| 21:24.14 | gusto | travel |
| 21:24.15 | Penguin | And where they are located. |
| 21:25.47 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 21:26.12 | gusto | however, it works nwo |
| 21:26.13 | gusto | now |
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| 21:48.36 | boom^time | Hey guys, I've gotten almost every ORiginateReponse Failure reason codes figured out except for 0. I was getting it during testing and I'm not sure why and now I can't reproduce it. Is anyone familiar with it? |
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| 22:43.48 | talntid | if I want to play a background sound 3 times (or repeat indefinately), is there a better way than just running background() 3 times? |
| 22:44.09 | jblack | you could loop, but i'd jut do it 3 time |
| 22:44.18 | WIMPy | Runit one with the file listed 3 times. |
| 22:45.02 | talntid | hm, ok, i didn't see a flag to run it X times or anything. wasn't sure if i missed it |
| 22:47.18 | jblack | I mean an actual loop |
| 22:47.35 | jblack | cmd while |
| 22:55.30 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 22:55.31 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 22:58.49 | *** join/#asterisk dar123 (~DAR@94.96.40.249) |
| 23:00.04 | dar123 | hey guyz, i was looking for a case study or tutorial to build a very scalable asterisk solution. Few million subscribers, i am unable to find anything yet |
| 23:01.47 | pabelanger | not much out their |
| 23:01.52 | pabelanger | there* |
| 23:02.10 | pabelanger | also, how do you want to scale |
| 23:04.57 | dar123 | i have no idea at the moment |
| 23:05.25 | pabelanger | well, the basic concept is drop proxies in front of asterisk |
| 23:05.33 | pabelanger | that will let you scale |
| 23:06.43 | dar123 | sounds good |
| 23:07.13 | dar123 | let me keep searching will let you know if i find something really intersting |
| 23:07.23 | dar123 | interesting* |
| 23:08.09 | pabelanger | dar123, really all depends on what you want to do |
| 23:08.37 | pabelanger | Right now, we are trying to build out a scalable queue system. For us, we are building the queue on top of asterisk and just using asterisk to plan media |
| 23:11.14 | dar123 | hmmm |
| 23:12.05 | dar123 | i am very new to asterisk, total 10 days of experience |
| 23:13.38 | pabelanger | okay, you likely want to sit down with a consultant or something then. Aiming for a few million customer is good and all, but might be hard to get to with 10days XP |
| 23:13.51 | pabelanger | but, lots of info, just need to know where to look |
| 23:15.18 | *** join/#asterisk GameGamer43 (sid5533@gateway/web/irccloud.com/x-gtzoyldypqdzgscl) |
| 23:17.03 | *** join/#asterisk kemmler (tully@r74-192-178-52.htvlcmta01.hnvitx.tl.dh.suddenlink.net) |
| 23:17.25 | kemmler | How can I increase the SIP retransmit timeout? |
| 23:18.36 | pabelanger | sip.conf |
| 23:18.40 | pabelanger | timert1 |
| 23:18.56 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:19.16 | pabelanger | but if you need to play with them, sounds like you have network issues |
| 23:19.50 | kemmler | don't have network issues atm, just looking to monitor them. |
| 23:19.53 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 23:20.11 | dar123 | i have 10 yrs voip experience |
| 23:20.54 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 23:20.55 | dar123 | i also know how to build SP networks, it's just matter of finding the right information and guidance |
| 23:21.13 | SeRi | it's a trap |
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