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| 03:17.09 | smirker | Hi, I'm changing the systemname in asterisk.conf (Asterisk 1.6) to change the way Unique IDs are generated and stored in CDRs. I want to apply this change without restarting (it's a production server). I've tried 'reload' and 'module cdr* reload' but the Unique IDs haven't changed. Any suggestions? :) |
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| 03:31.41 | smirker | I'm assuming I'll need to restart. |
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| 03:45.24 | MaliutaLap | core restart gracefully, although you might want to check there are no active calls |
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| 05:34.48 | swarmhost | exten => s,n,GotoIf($["${CONTEXT}" != ""]?${CONTEXT},s,1}) |
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| 05:35.04 | swarmhost | this is sending to it's parent context. can someone give me advice on that? |
| 05:36.00 | swarmhost | ${CONTEXT} = OPEN, but instead of sending it to OPEN,s,1, it sends to mainmenu,s,1. |
| 05:39.04 | swarmhost | as if it isn't evaluating the variable on the gotoif |
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| 06:16.18 | lnb | anyone know what causes fast busy signal? |
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| 06:23.04 | ziz212 | Hi, |
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| 06:45.43 | ChannelZ | lnb: congestion or some other failure |
| 06:46.33 | lnb | hi |
| 06:46.44 | b7 | Hello everyone. Please, can anyone explain, what does dialing timeout depends on? I mean, after I did input the number there is some time to wait before the dialing starts. Is this the phone's setting or should I setup this in my Asterisk server? |
| 06:47.39 | lnb | well i was told by people trying to call me they heard fast busy signals... some gave me time called and thier number, doesn't show in cdrs on our server nor on itsp |
| 06:47.41 | Penguin | It could be the phone's setting. |
| 06:47.48 | Penguin | b7: It could be the phone's setting. |
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| 06:48.01 | lnb | these are different people in different places |
| 06:48.11 | lnb | and none are stupid either |
| 06:48.16 | Penguin | b7: Some phones have a digit map or dial plan that controls the timeout after a pattern is entered. |
| 06:48.40 | lnb | i can understand a regular'busy signal' if on phone and call holding |
| 06:49.13 | lnb | odd |
| 06:49.22 | Penguin | The "fast busy" is temp fail or congestion. |
| 06:49.39 | Penguin | It is a failure, not a busy indicator. |
| 06:50.11 | lnb | Penguin: but shouldn't the call show up on itsp cdrs as failure or congestion? |
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| 06:50.38 | lnb | i should say calls as there were more than 10 |
| 06:51.00 | lnb | people were calling from different NPA's |
| 06:51.29 | Penguin | If the call failed before it was processed in their system, why should it show up in CDR? |
| 06:52.24 | Penguin | The only thing that will help determine where the failure is will be to debug the call. |
| 06:52.25 | lnb | agreed |
| 06:52.41 | lnb | but 10 people from 10 different NPA's ? |
| 06:52.58 | Penguin | I'm not going to try to guess. |
| 06:53.03 | lnb | how do debug the call that shows up nowhere |
| 06:53.11 | Penguin | sip set debug on |
| 06:53.26 | lnb | i know that :) |
| 06:53.53 | lnb | but if the call doesn't reach my server and according to ITSP, not thier server.. then what |
| 06:54.06 | Penguin | Network failure? |
| 06:54.11 | lnb | well according the CDRS I looked at |
| 06:54.29 | lnb | its a mystery |
| 06:54.42 | lnb | i guess call Dr. House |
| 06:56.27 | MaliutaLap | lnb: it never existed |
| 06:56.51 | MaliutaLap | it's like the sound from the tree that fell in the woods when nobody was around :) |
| 06:59.16 | lnb | MaliutaLap: i wish you were right, but the people that called, like I said before, are intelligent and are all in different NPANXXXX's. How is it that 10 people fell off the tree? |
| 07:00.04 | b7 | Penguin, thanks, seems to be that. |
| 07:00.27 | lnb | the ONLY thing i can think of is, my NPA was borked at some point in the pstn world |
| 07:00.58 | lnb | which means no one in the world would have been able to call my NPA |
| 07:01.03 | lnb | if that is even possible |
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| 07:03.13 | lnb | i know one of our ITSPs sent out email about 2 weeks ago regarding an incident with a larger carrier which killed off several NPAs for a couple of days. |
| 07:04.55 | m0sphere | i've just installed festival on my server via yum, tested with phpagi and it works but I want to change the voice. apparently this is done in /usr/share/festival/lib/voices.scm and i see in the config it says "define current-voice nil" |
| 07:05.09 | m0sphere | now am i supposed to replace the "nil" part with the name of a voice?? |
| 07:09.04 | swarmhost | exten => s,n,GotoIf($["${CONTEXT}" != ""]?${CONTEXT},s,1}) |
| 07:09.06 | swarmhost | this is sending to it's parent context. can someone give me advice on that? |
| 07:09.11 | swarmhost | ${CONTEXT} = OPEN, but instead of sending it to OPEN,s,1, it sends to mainmenu,s,1. |
| 07:09.14 | swarmhost | as if it isn't evaluating the variable on the gotoif |
| 07:10.23 | kaldemar | swarmhost: show CLI output for that |
| 07:11.22 | kaldemar | you have an extra curly brace in there after 1 btw. |
| 07:11.31 | swarmhost | yeah I have since removed that one |
| 07:12.35 | swarmhost | http://pastebin.com/Va40Cv20 |
| 07:13.06 | swarmhost | dialplan: http://pastebin.com/yA1Ed5gZ |
| 07:13.12 | kaldemar | and that line really should equal Goto(s,1) |
| 07:13.21 | lnb | trying to get encryption between sip phone and pbx server. Have openvpn working and phone shows VPN. However, doing a tcpdump on the gateway server (between sip phone and pbx server) , it sure doesn't look encrypted |
| 07:13.43 | kaldemar | Executing [s@mainmenu:5] GotoIf("IAX2/north-ella-2596", "1?mainmenu,s,1") in new stack <-- your context is "mainmenu". |
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| 07:14.05 | swarmhost | but the variable ${CONTEXT} is opennoivr |
| 07:14.29 | kaldemar | swarmhost: no, it's not. :) CONTEXT is a reserved variable that holds the current context. use something else for your own purposes. |
| 07:14.46 | swarmhost | do'h! |
| 07:14.47 | swarmhost | lol. |
| 07:15.08 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables |
| 07:15.54 | swarmhost | damn i'm an idiot. |
| 07:15.59 | swarmhost | high five, kaldemar. it works. |
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| 08:36.43 | ziz212 | hi, |
| 08:39.19 | ziz212 | I am trying TLS, SRTP.. server side settings are ok and certificate is accepted whcih show in sip relaod ... TLS address and port is also taken. But soft phone try to register it is now happening. Even sip debug doesnt show any information about connection. Do we have any options to check this .. I mean some debugging or any other log files ? |
| 08:39.21 | ziz212 | pls help |
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| 08:57.50 | rox | Morning, I have a question regarding hinting, is it possible to have phones display arbitrary data on display using hints? |
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| 09:44.39 | b7 | It's me again. :) Does anybody know, which version is right: Dial(SIP/1001&SIP/1002,,tT) or Dial(SIP/1001,,tT&SIP/1002,,tT)? I want to let any of the SIP users to transfer thier calls. |
| 09:45.35 | skrusty | options should be at the end - it's not destination independant |
| 09:45.41 | skrusty | as far as i know |
| 09:48.32 | kaldemar | Dial(Technology/Resource[&Technology2/Resource2[&...]][,timeout[,options[,URL]]]) |
| 09:48.53 | b7 | Thank you. |
| 09:51.00 | *** join/#asterisk atha (~atha@unaffiliated/athayde) |
| 09:54.58 | sjs205 | Morning all, can anyone help my debug a problem whereby I'm getting an "unable to create/find SIP channel" seems to work for all but one sip client |
| 09:55.16 | ziz212 | I am trying TLS, SRTP.. server side settings are ok and certificate is accepted whcih show in sip relaod ... TLS address and port is also taken. But soft phone try to register it is now happening. Even sip debug doesnt show any information about connection. Do we have any options to check this .. I mean some debugging or any other log files ? |
| 09:55.29 | ziz212 | all have questions ... |
| 10:12.14 | mirela666 | sjs205: what does the CLI output says exactly? |
| 10:13.29 | sjs205 | mirela666, shall I send you a pastebin of my asterisk output? |
| 10:13.43 | mirela666 | sjs205: sure, paste it here |
| 10:13.50 | mirela666 | the link |
| 10:16.36 | sjs205 | mirela666, the full asterisk dialog is here: http://pastebin.com/P6skqsSN |
| 10:16.43 | sjs205 | Thank you for looking... |
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| 10:23.19 | sjs205 | mirela666, I think it could be due to "lack of codecs" but I'm not 100% sure, especially since I seem to have the correct codecs in place whin initilising the call... |
| 10:24.04 | mirela666 | sjs205, and sip show peers show hcm-thread as registered? |
| 10:25.41 | mirela666 | <PROTECTED> |
| 10:25.57 | sjs205 | mirela666, I just ran the application again and while it was dialing did sip show peers... and no, it isn't regestered |
| 10:26.32 | mirela666 | sjs205: that is why channel can't be created |
| 10:26.37 | sjs205 | The thing is that the application should perform registration just before making the call... |
| 10:27.08 | sjs205 | mirela666, hmmm... that makes sense... although the other end does ring, the call just ends when 0104 answers |
| 10:29.20 | mirela666 | sjs205: hm I had the same biheviour with soft phone, not registering but could ring ,. ... |
| 10:29.52 | sjs205 | :q |
| 10:29.57 | sjs205 | oops... |
| 10:30.09 | mirela666 | or couldn't ring but could dial in |
| 10:31.12 | sjs205 | mirela666, yeah, that must be the problem... although my other peer which doesn't show in sip show peers is able to make the call sucessfully... is registered |
| 10:31.46 | sjs205 | sorry, the other peer isn't registered is able to sucessfullyy make the call... |
| 10:32.49 | mirela666 | sjs205, and sip show registry shows registration? |
| 10:33.00 | sjs205 | mirela666, nope... |
| 10:33.06 | mirela666 | hmm |
| 10:33.51 | sjs205 | Could it be that these users are all realtime |
| 10:33.52 | sjs205 | ? |
| 10:35.22 | sjs205 | mirela666, the only users that are showing in sip show peers are the ones defined in my sip.conf.... non of my realtime ones are... |
| 10:35.56 | sjs205 | although I know realtime is working otherwise the wouldn't authenticate... |
| 10:36.07 | mirela666 | oh ok, but they should be seen in registry i guess, hmm |
| 10:36.38 | sjs205 | the only thing to show in registery is my outbound line... |
| 10:36.42 | sjs205 | grrr... |
| 10:41.42 | *** join/#asterisk gaps (79f28b33@gateway/web/freenode/ip.121.242.139.51) |
| 10:42.54 | gaps | attended transfer channel is not released in Asterisk 11.1.2 |
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| 12:06.51 | *** join/#asterisk JamKo (~JamKo@unaffiliated/jamko) |
| 12:08.40 | JamKo | Greetings. I have recently upgraded to asterisk 11.6.0. In 1.8 I was able to update the udptl.conf file to use alternate ports outside of the reserved 4000 range. In 11.6 asterisk is ignoring an |
| 12:08.50 | JamKo | alternate port range. Anyone heard of this as an issue? |
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| 12:15.37 | *** join/#asterisk BKhan (ca7d90e3@gateway/web/freenode/ip.202.125.144.227) |
| 12:15.43 | BKhan | Hi |
| 12:16.06 | BKhan | I want to change my DNID |
| 12:16.14 | BKhan | Need some help regarding to it |
| 12:26.01 | *** join/#asterisk gryphon (~gryphon@82.140.120.164) |
| 12:27.25 | slicknick5181 | BKhan, I appologize I can't assist you but this room comes alive over the next hour or 2 |
| 12:27.54 | slicknick5181 | BKhan, After more people join you may want to post again |
| 12:30.51 | BKhan | ok thanks. I will do it after 2 hours |
| 12:35.32 | WIMPy | BKhan: What do y want to change? |
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| 12:36.21 | ziz212 | How can I find my asterisk support srtp and tls ? |
| 12:37.04 | ziz212 | does it complied with srtp enabled some thing like that? |
| 12:37.51 | WIMPy | module show like srtp |
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| 12:38.58 | ziz212 | is it .so file |
| 12:39.00 | ziz212 | ? |
| 12:39.18 | WIMPy | just enter that in the *CLI. |
| 12:39.22 | ziz212 | in /user/lib/astersik |
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| 12:40.35 | ziz212 | thanks |
| 12:40.38 | ziz212 | I got it |
| 12:41.14 | ziz212 | thansks |
| 12:41.28 | ziz212 | And how can I sure that tls is working with the server |
| 12:41.54 | ziz212 | it shows that sip show settings and sip show peer <extension> |
| 12:41.58 | ziz212 | tls something |
| 12:42.37 | WIMPy | If you have a port in 'sip show settings' it's active. |
| 12:45.30 | ziz212 | yes |
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| 12:46.53 | ziz212 | TLS SIP Bindaddress: |
| 12:46.56 | ziz212 | it is there |
| 12:47.31 | ziz212 | where I have configured that in sip.conf the settings in the asterisk wiki related to srtp and tls |
| 12:48.20 | ziz212 | What I need to know is ... dose it show any information like sip-tls or some thing in netstat -ntlp command in bash ? |
| 12:48.40 | ziz212 | something related to tls ? |
| 12:48.46 | ziz212 | or srtp ? |
| 12:57.24 | BKhan | WIMPy: I want to change soure address of a call when call comes in our server |
| 13:03.21 | ziz212 | Dear friends, Why it is not showing the incoming TLS SRTP connections when sip debug on ? |
| 13:04.07 | slicknick5181 | BKhan, Why? What are you trying to accomplish? |
| 13:05.28 | BKhan | Actually our requirment is that when an incoming call hit to agent we want to append company name with its DNID number |
| 13:09.52 | BKhan | slicknick5181: Actually our requirment is that when an incoming call hit to agent we want to append company name with its DNID number |
| 13:10.43 | slicknick5181 | BKhan, is this an asterisk distro or a base install |
| 13:17.26 | WIMPy | ziz212: rtp and srtp use the same port range. And otherwise netstat will tell you the same as 'sip show settings'. |
| 13:18.31 | WIMPy | BKhan: If you don;t want to use multiple accounts, setting CALLERID(name) is usually a good way to transport extra information. |
| 13:19.26 | ziz212 | Thank Wimpy .. Does it show any revelant information in "netstat-ntlp" ? |
| 13:19.37 | ziz212 | regarding srtp tls |
| 13:19.59 | WIMPy | no |
| 13:24.19 | ziz212 | Thanks; |
| 13:24.26 | BKhan | slicknick5181: Thanks but we need to set it on incoming call |
| 13:24.58 | BKhan | slicknick5181:CALLERID(name) sets for outgoing call. |
| 13:25.49 | slicknick5181 | BKhan, This varible is not dirction specific. Can be set either way |
| 13:26.21 | WIMPy | You want to set it on an outgoing call. |
| 13:26.50 | WIMPy | Call from Asterisk = outgoing |
| 13:27.58 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
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| 13:33.54 | BKhan | hmm thanks let me try it |
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| 13:44.20 | gaps | channels are not releasing for attended transfer... in incoming queues... |
| 13:45.22 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 13:51.19 | slicknick5181 | I am extremely confused by the dahdi.conf file. I have a TDM400 card 1fxo 1 fxs. dahdi_chnnels.conf has the card configured correctly but I have no clue what to do with dahdi.conf to complete the process........... Asterisk 1.8 |
| 13:51.28 | slicknick5181 | I'm sorryAsterisk 11 |
| 13:52.53 | WIMPy | there's no dahdi.conf. |
| 13:53.22 | WIMPy | You can just #include (or copy the contents of) dahdi_channels.conf to your chan_dahdi.conf. |
| 13:54.39 | slicknick5181 | My apologies chan_dahdi.conf. Thank you I will try |
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| 14:13.44 | ghost75 | in cdr_custom.conf only ${CDR... variables can be used? |
| 14:17.06 | *** join/#asterisk linjan (~linjan@176.195.237.203) |
| 14:17.55 | linjan | Hello! Maybe someone knows where to buy a number (DID) in Belize with access via SIP? |
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| 14:27.02 | ghost75 | exten => h,n,set(CDR(rjitt)=${CHANNEL(rtpqos,audio,remote_jitter)}) <- that is working? |
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| 14:28.32 | volga629 | Hello Everyone is anybody so that kind of warning |
| 14:28.35 | volga629 | [2013-11-22 15:26:37] WARNING[3147]: res_corosync.c:316 ast_event_cb: CPG mcast failed (6) |
| 14:28.37 | volga629 | [2013-11-22 15:26:37] WARNING[3147]: res_corosync.c:316 ast_event_cb: CPG mcast failed (6) |
| 14:31.53 | slicknick5181 | WIMpy, I included the file with #include "dahdi-channels" and I put that at the top of the file. no dial tone and dahdi show channels returns no channels |
| 14:31.59 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.44) |
| 14:32.28 | WIMPy | You need to #include that at the end. And use the full file name. |
| 14:33.06 | slicknick5181 | the full path? /etc/asterisk/ |
| 14:33.17 | WIMPy | .conf |
| 14:34.32 | slicknick5181 | ok |
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| 14:53.38 | slicknick5181 | DAHDI working correctly but everytime I make or recive a call after I have hung up the caller ID shows a missed call from the dahdi extension |
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| 14:56.16 | ghost75 | what is difference between jitter, normdevjitter and stdevjitter ? |
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| 15:09.05 | WIMPy | What do you mean by "dahdi extension"? |
| 15:09.12 | WIMPy | Or the "from"? |
| 15:10.00 | Penguin | Calls are not from extensions. |
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| 15:11.04 | Penguin | "from the dahdi extension" must mean from a phone connected to a dahdi port/channel. |
| 15:11.34 | WIMPy | Possible, but I don't want to guess. |
| 15:11.50 | Penguin | I would prefer that as well. |
| 15:13.28 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 15:14.45 | slicknick5181 | YI'm soory, Yes a phone connected to the card |
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| 15:15.36 | slicknick5181 | extension is 5005 and when extension 5005 hangs up for some reason the Caller ID displays Showroom 5005 which is the caller ID info I have set for that DAHDI port |
| 15:16.27 | WIMPy | I'm still not sure, what displays where and especially not WHEN. |
| 15:16.36 | Penguin | When you hang up the phone, that's when the callerid shows the name/number on the display? |
| 15:16.48 | *** part/#asterisk volga629 (~bendersky@CPE085b0e07d3f2-CM7cb21b15b251.cpe.net.cable.rogers.com) |
| 15:16.56 | Penguin | And it shows the name/number of itself? |
| 15:18.07 | slicknick5181 | Correct |
| 15:18.44 | Penguin | Does the phone also ring when you hang it up? |
| 15:18.48 | slicknick5181 | and then the phone registers a missed call - also the phone does not ring when this occurs |
| 15:19.13 | *** join/#asterisk dxd828 (~dxd828@195.191.107.205) |
| 15:20.00 | slicknick5181 | noteworthy I do have Hangup in my dialplan |
| 15:20.29 | Penguin | Do you have an extension pattern of _. in the dialplan? |
| 15:20.44 | *** part/#asterisk tomdm (~tomdemoor@ip-188-118-0-245.reverse.destiny.be) |
| 15:21.00 | slicknick5181 | no I have XXXXXXXXXX and 1XXXXXXXXXX for outbound calls |
| 15:21.15 | slicknick5181 | I realize there should be a few N's in there |
| 15:21.39 | Penguin | NXXNXXXXXX for North America |
| 15:21.41 | WIMPy | Extensions are for inbound calls. Asterisk is not a PBX so the terminology is somewhat different. |
| 15:22.42 | slicknick5181 | cect, Could these letters create this call back? It also does not happen on my SIP extensions |
| 15:23.35 | Penguin | The only extensions you told me about so far are ones you use to make calls outbound. |
| 15:24.13 | slicknick5181 | I have 4 SIP extensions and 1 DAHDI extension |
| 15:24.28 | Penguin | Extensions do not have technologies. |
| 15:24.29 | WIMPy | Extensions are destinations defined in your extensions.conf. The SIP stuff are peers or devices. |
| 15:24.36 | slicknick5181 | I have 1 DAHDI outbound line and SIP failover in DAHDI is in use |
| 15:25.24 | slicknick5181 | My dialplan is working it's just the caller id issue on the DAHDI phone |
| 15:25.51 | Penguin | My guess was a pattern of _. in dialplan, but you said that doesn't exist. |
| 15:26.33 | slicknick5181 | Correct I refuse to use that |
| 15:26.39 | Penguin | Good. |
| 15:26.50 | slicknick5181 | Agreed |
| 15:28.16 | Penguin | My experience excludes the TDM stuff, so if it isn't caused by dialplan, it must be a dahdi configuration issue. Maybe someone else will be willing to go over that part. |
| 15:29.12 | WIMPy | My last experience with analog stuff was before caller ID display existed. |
| 15:29.13 | slicknick5181 | Mine as well :/ I'm relatively handy with SIP but the DAHDI is kicking me in the rear |
| 15:29.54 | Penguin | I do all VoIP stuff, so that's my scope. |
| 15:30.38 | slicknick5181 | Thats what I want to do but this is a Small buisness so going straight VoIP is way more xpensive |
| 15:31.40 | Penguin | More expensive? Interesting. |
| 15:32.31 | slicknick5181 | very much so |
| 15:32.32 | WIMPy | That's what I've been saying for ages. |
| 15:33.01 | ghost75 | oh wow 5% packet loss on zoiper |
| 15:33.02 | Penguin | I don't see how it's more expensive. |
| 15:33.27 | slicknick5181 | In this small office we only need a max of 2 outgoing lines and maybe 3 incoming |
| 15:33.53 | WIMPy | Administrative costs, electricity and high call rates with ITSPs. |
| 15:34.08 | slicknick5181 | if you search around you can find the right provider for cheap or even free incoming lines. I have our PSTN Call forward busy to a free voip number |
| 15:34.32 | WIMPy | But it's good for lots of cheap inbound channels if you don't need reliability. |
| 15:35.25 | slicknick5181 | And I barely use the VoIP lines there are spurts like any buisness |
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| 15:39.03 | slicknick5181 | but my provider also has voicemail so in event of a fail I'm covered |
| 15:40.19 | Penguin | I have mine configured so that in the event of a failure, there is an auto attendant on the provider's side and then it forwards to my mobile. |
| 15:41.10 | slicknick5181 | I could set that up but I'm alright with voicemail lol |
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| 15:41.33 | gaps | channels are not releasing for attended transfer... in incoming queues... |
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| 16:51.16 | As001 | WIMPy I can block unwanted events in manager in version 1.8 of Asterisk with help of event filter like this eventfilter=!Event: QueueMemberStatus |
| 16:53.54 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.44) |
| 16:56.30 | As001 | I wonder why I didn't found it in manager.conf documentation in file. There is no trace of that option in comments of manager.conf |
| 16:58.49 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/11/configs/manager.conf.sample?revision=371121&view=markup |
| 16:59.05 | [TK]D-Fender | line 96 of the provided sample configs which are "documentation" |
| 16:59.49 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/1.8/configs/manager.conf.sample?revision=353770&view=markup |
| 16:59.54 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 16:59.55 | [TK]D-Fender | 89 in the 1.8 branch |
| 17:00.29 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 17:00.50 | Katty | hi kids. |
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| 17:01.29 | As001 | I guess I had older configuration file then... I will check from samples config folder. |
| 17:02.19 | As001 | lol you are right [TK]D-Fender |
| 17:02.34 | As001 | I need other glasses definitely. |
| 17:03.26 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/1.8/configs/manager.conf.sample?revision=279056&view=markup |
| 17:03.34 | [TK]D-Fender | 3 years ago, oldest revision at the bottom. |
| 17:03.36 | [TK]D-Fender | yup. |
| 17:05.00 | Katty | hi fender bender. |
| 17:05.07 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 17:08.41 | As001 | I had that in config files and I didn't look in samples I looked in my modified manager.conf file and now I am complaining... I am unbelievable. Sorry. |
| 17:09.22 | As001 | is going to punish himself |
| 17:09.25 | *** part/#asterisk As001 (~uros@82.117.198.142) |
| 17:10.38 | [TK]D-Fender | Katty: Mew. |
| 17:10.51 | Katty | how're you dear. |
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| 17:24.20 | [TK]D-Fender | Katty: Getting by ... took the day off so was able to stay out late last night and am doing sound for a friend's show early this evening... |
| 17:24.26 | [TK]D-Fender | Katty: Packing things upp now |
| 17:24.30 | Katty | nods |
| 17:24.44 | Katty | your collarbone still doing ok? |
| 17:26.02 | [TK]D-Fender | long since finished doing what it's doing... is the way it'll be until something else happens... |
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| 17:33.46 | lnb | what causes call dropping? |
| 17:35.39 | Penguin | The unplugging of cables would cause call dropping. |
| 17:35.45 | *** join/#asterisk MaliutaLap (~nobusines@eth637.qld.adsl.internode.on.net) |
| 17:39.38 | carrar | only if the cable is black or gray |
| 17:39.59 | Penguin | The blue ones keep working after they are cut off? |
| 17:40.03 | carrar | yes |
| 17:40.06 | Penguin | I think I heard something about that before. I never tested it. |
| 17:40.39 | carrar | Something about residual color energy keeps the path there |
| 17:41.13 | Penguin | Residual capacitance in the pigment or something. |
| 17:41.41 | carrar | yeah it complicated stuff |
| 17:41.45 | carrar | its |
| 17:41.48 | carrar | I can't explain it |
| 17:42.57 | lnb | really? |
| 17:43.22 | lnb | wow i had no idea that removing a cable would have any effect whatsoever. |
| 17:43.41 | lnb | gese, you learn something new every day! |
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| 17:53.53 | CrashSys | anyone noticed an significant different between 32 or 64 bit with Asterisk v.1.8 as far as load goes? |
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| 18:04.53 | Joel_re | hey, if I want to invoke a script on an incomming call, what should I be looking at? |
| 18:05.09 | navaismo | agi or system |
| 18:05.49 | Joel_re | navaismo: ok, and I should be able to pass the script details like the caller id and keys pressed? |
| 18:06.53 | Joel_re | ok seems like thats possible |
| 18:06.54 | Joel_re | thanks |
| 18:09.09 | navaismo | yep |
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| 18:25.15 | xytis | Hi, I am setting up asterisk to work with WebRTC client based on SipMl5 library. Currently asterisk (versions 11 and 12) are experiencing crashes each time client requests HOLD on the connection. |
| 18:26.57 | xytis | I traced the error to PJICE library, which is used in 12 version. Crashes are created by assertion in pj_ice_sess_create_check_list function. |
| 18:28.10 | navaismo | hold in sipml5 is an unstable function, you can saw the uissues reported on that and there is no fix yet |
| 18:28.43 | navaismo | s/saw/see/ |
| 18:28.55 | xytis | I do understand that. |
| 18:29.28 | xytis | Yet the problem appears to be in PJSIP and asterisk, not in sipml5. |
| 18:30.13 | navaismo | well, if you are shure about that you can raise a bug in the JIRA page with the method to replicate and pointing to the issue |
| 18:30.30 | xytis | Ok, I'll do that. |
| 18:31.13 | xytis | Yet I am here because I need to fix that thingy as soon as possible AND it appears that the fix is not so complicated. |
| 18:31.50 | Katty | i too, fix thingies that appear to not be so complicated |
| 18:32.20 | Katty | like the water line running to the fridge :< |
| 18:32.49 | xytis | Mostly because I think the assertion is incorrectly formed. Straight from source: PJ_ASSERT_RETURN(rcand_cnt + ice->rcand_cnt <= PJ_ICE_MAX_CAND, PJ_ETOOMANY); |
| 18:33.18 | navaismo | xytis if you can fix it create a patch and submit in the bug report |
| 18:34.14 | xytis | Well, I would like to consult someone with extensive knowledge about this area before I do that. |
| 18:35.02 | xytis | Since it is always a tricky thing to alter an assertion condition without knowledge about specifics... |
| 18:36.30 | xytis | If patching is the way to do that, then I do have even more questions about where the patch should be submitted. Since the same line exists in both svn trunk of PJ libs and github.com/asterisk/pjproject |
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| 18:44.24 | navaismo | wait for the experts |
| 18:48.13 | Joel_re | hey Katty, heh after ages |
| 18:48.20 | Joel_re | <- smooth_penguin :p |
| 18:54.58 | *** join/#asterisk jdoe (jdoe@falseprophet.ca) |
| 18:56.04 | Penguin | <---- ruffled |
| 18:56.07 | jdoe | how can I make asterisk silently reject an extension? |
| 18:56.23 | jdoe | NOTICE[30815]: chan_sip.c:23272 handle_request_invite: Call from '1234567890' (1.2.3.4:5060) to extension '9995551212' rejected because extension not found in context 'whatever'. |
| 18:56.29 | jdoe | ^ basically I want to suppress that under some circumstances. |
| 18:56.47 | Penguin | You'll have to create the extension and execute Hangup(). |
| 18:56.58 | Penguin | If it doesn't exist, you'll get the NOTICE. |
| 19:02.49 | *** join/#asterisk vlad_starkov (~vlad_star@77.50.56.176) |
| 19:03.07 | jdoe | hmm... still get logs for the call, but it's better I guess. Thanks. |
| 19:03.23 | Penguin | What do you mean by logs? |
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| 19:04.54 | jdoe | Penguin: just that I still see it as it executes the dialplan. This is working around a provider's inability to forward a number without trying my pbx first :/ |
| 19:05.45 | paulc | jdoe: why not forward it in your PBX? All numbers in, you control where they go to next? |
| 19:06.05 | Penguin | Or don't even forward it at all. |
| 19:06.16 | Penguin | Just Dial() the place you want the call to go. |
| 19:06.29 | Joel_re | hey Penguin :> |
| 19:06.32 | jdoe | because that would leave me double-billed for call time? |
| 19:06.52 | Penguin | Yeah it would. |
| 19:06.59 | jdoe | that's why ;) |
| 19:07.14 | Penguin | That's a pretty good reason. |
| 19:07.52 | jdoe | I'm not sure why they can't forward the number blindly, only as a fallback. |
| 19:08.02 | *** join/#asterisk dxd828 (~dxd828@85.255.232.211) |
| 19:09.11 | Penguin | What is the default skipms value for ControlPlayback()? |
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| 19:09.39 | pabelanger | 15 rings a bell |
| 19:10.06 | Joel_re | so I have asterisk and the sip client runnig on the same box, I end up with "retrans_pkt: Retransmission timeout reached on transmission' |
| 19:10.13 | Joel_re | midway through the demo context |
| 19:10.15 | Penguin | I know I saw it written somewhere, but I can't find it now that I wanted to know. |
| 19:10.35 | Katty | Joel_re: SMOOTH OPERATOR! |
| 19:10.37 | Katty | hugs Joel_re |
| 19:10.59 | Joel_re | heylo :> |
| 19:11.18 | Joel_re | Katty: did you ever start critter cam again? |
| 19:11.47 | Joel_re | now has a squirrel feeder with them feeding all day, need to find a good cam |
| 19:12.17 | Katty | i did! |
| 19:12.27 | Katty | infobot: crittercam |
| 19:12.27 | infobot | i heard crittercam is Katty's Critter Cam http://tinyurl.com/b5k3lt4 |
| 19:12.36 | Joel_re | woot |
| 19:12.37 | Joel_re | thanks |
| 19:13.06 | Katty | i had a scarecrow in the front yard for awhile as decor, and they stopped showing up |
| 19:13.14 | Katty | regardless of the fact that i put feeders on the scarecrow |
| 19:13.20 | Katty | oh well, i'll know for next year ;) |
| 19:13.57 | Joel_re | heh |
| 19:14.07 | navaismo | Its normal when you use ${PJSIP_DIAL_CONTACTS()} and one of the contact are busy the other never rings? |
| 19:15.18 | Katty | Joel_re: are you going to eat the squirrels? |
| 19:15.21 | Katty | Joel_re: or just watch them? |
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| 19:19.13 | ghost75 | wow lot of birdies |
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| 19:25.02 | Joel_re | Katty: :p eat the squirrels?! |
| 19:25.31 | *** part/#asterisk jdoe (jdoe@falseprophet.ca) |
| 19:25.40 | Penguin | I'd eat 'em. |
| 19:25.53 | Joel_re | I'd want to put them on the internet |
| 19:26.02 | Joel_re | make them famous and fat |
| 19:26.14 | Penguin | But they are so tasty. |
| 19:26.27 | Penguin | If you put them on the internet, where's the noms in that? |
| 19:26.35 | Joel_re | heh |
| 19:26.55 | Katty | lol |
| 19:26.57 | Katty | i was just asking. some people eat squirrel ;) |
| 19:27.06 | Penguin | Heck yeah we do. |
| 19:27.20 | Penguin | They're made of meat. |
| 19:27.38 | Joel_re | Penguin: the ones we have here are too tiny |
| 19:27.59 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:27.59 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:28.07 | Penguin | I've seen some so small that it would take a dozen to make a meal for one person. |
| 19:28.51 | *** join/#asterisk dxd828 (~dxd828@85.255.232.211) |
| 19:35.54 | Katty | i'm kind of curious how one eats a squirrel |
| 19:36.42 | Penguin | Eat it like a chicken leg. |
| 19:37.26 | Penguin | You can cut it up and fry the smaller pieces and eat them like you eat anything. |
| 19:37.55 | *** join/#asterisk user0 (~user0@195-132-180-226.rev.numericable.fr) |
| 19:38.47 | Katty | no mean, how you clean it |
| 19:38.52 | Katty | and the recommended way of preparing it |
| 19:38.57 | user0 | Hello, i can't log in to the Voicemail & Recordings (ARI) http interface, can someone help ? |
| 19:39.17 | Katty | like if you need to put it in a crockpot |
| 19:39.27 | user0 | Can't find any documentation ... |
| 19:39.29 | Katty | like sirloin ;) |
| 19:43.13 | ghost75 | chinese would eat them for sure |
| 19:45.35 | Penguin | Cut the skin around the rear ankles and peel the whole skin toward the head. When the skin is peeled all the way to around the neck, cut off the head taking the skin with it. Soak the meat (body) in some salt water for a while to draw out excess blood. You can then butcher it to your liking. Take the meat pieces, dredge in plain flour, dip through buttermilk or egg wash, then dredge through seasoned flour. Pan fry until ... |
| 19:45.39 | navaismo | ate grasshopper & maguey worms |
| 19:45.41 | Penguin | ... fully cooked. |
| 19:46.35 | Joel_re | ugh |
| 19:47.48 | Penguin | Oh, I didn't explicitly mention cutting open the body to take out the guts. I probably should have mentioned that after skinning and before soaking. |
| 19:48.38 | Penguin | Kill, skin, gut, soak, cut, cook. |
| 19:50.21 | ghost75 | skin alive |
| 19:50.43 | Penguin | I doubt they will hold still while you nail them to the tree. |
| 19:51.11 | ghost75 | saw a video where they did on bigger animals :< |
| 19:52.02 | Penguin | That's how we always skinned them after hunting. We'd hang them on the tree by the hind legs, back against the tree, on nails. |
| 19:52.17 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-220-20.socal.res.rr.com) |
| 19:52.38 | Penguin | Cut the skin between the leg bone and the tendon and put that over the nails, which were always left in the tree. |
| 19:53.12 | Penguin | When you cut the skin around the legs, then you just have to pull the whole skin downward until it reaches the head. |
| 19:53.44 | Penguin | If you don't have it nailed to the tree, it can be easier to have someone hold the back legs while you pull the skin. |
| 19:53.58 | file | user0, huh? |
| 19:54.34 | navaismo | user0, try #freepbx |
| 19:55.12 | Penguin | Rabbits are done the same way. |
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| 20:14.49 | Joel_re | :< |
| 20:15.28 | Joel_re | a netstat -tulnp should list the AMI port |
| 20:15.31 | Joel_re | if its listening? |
| 20:15.37 | drmessano | I tried that with a human once, but he kept hitting me with his stapler and sending "Help" emails to the support desk |
| 20:15.45 | drmessano | Guess I should have invited him to lunch first |
| 20:16.45 | Joel_re | my application says Connection refused, but I dont see anything in manager debug |
| 20:16.54 | Joel_re | the app is running on the same host |
| 20:17.19 | *** join/#asterisk dar123 (~DAR@216.131.72.35) |
| 20:18.32 | Joel_re | enabled = no |
| 20:18.35 | Joel_re | slaps forehead |
| 20:20.52 | drmessano | You'll want to change that to yes |
| 20:20.59 | drmessano | </obvious> |
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| 20:34.14 | Joel_re | <PROTECTED> |
| 20:34.20 | Joel_re | oops |
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| 20:37.39 | Vendigroth | So I have a strange agi mystery. If a if a call comes in to dahdi channel, and is answered in extensions.conf everything works fine, but if it's passed to an agi script and it calls answer() I get dropped frame notice. any idea why that happens? Pri debug messages are identical. and why would the native format be different between those 2. |
| 20:50.18 | user0 | navaismo: trying ... |
| 20:54.03 | *** join/#asterisk saint_ (~saint@c-50-166-85-78.hsd1.nj.comcast.net) |
| 20:54.27 | saint_ | hi all - What is the best way to switch from a 3 way conference into a 4 parties ? |
| 20:55.11 | Penguin | How are the first three people conferencing? |
| 20:55.42 | saint_ | Penguin: well.. one called me... then I called the 2nd one ... i put the 3 of us in conf... |
| 20:55.54 | Penguin | Define "put in conf" |
| 20:55.56 | saint_ | Penguin: en then a 4th one called ne. |
| 20:56.09 | saint_ | Penguin: when the 1st one called me, I pushed the CONFERENCE button and called someone else |
| 20:56.20 | Penguin | What does your conference button do? |
| 20:56.57 | saint_ | Penguin: I m still in a 3 ways right now so I can't try. but from what I saw, it gives me a new dialtone.. I can call someone, then push CONFERENCE again and it put the 3 of us in a 3 way |
| 20:57.24 | Penguin | I don't need to know how you use it. I need to know what it does. |
| 20:57.51 | saint_ | Penguin: well.. I use DPMA with those asterisk phone, so whatever it is programmed to do by default. unless if you tell me that there is a file I should look into. |
| 20:57.56 | Penguin | If this is going to be a one-time thing, and if you have access to the CLI or AMI, you can redirect the channels of the people you are talking to now into a confbridge or meetme. |
| 20:58.32 | Penguin | Then you can call the 4th person and blind transfer him to the confbridge too. |
| 20:59.04 | saint_ | Penguin: so for example setup a meetme conf file, and if this case happens again, just throw them all in the meetme conf ? |
| 20:59.35 | Penguin | You don't need to configure a meetme room in the meetme.conf file. You can create it dynamically in dialplan. |
| 20:59.46 | Penguin | meetme or confbridge |
| 20:59.49 | saint_ | Penguin: would you have an example by any change ? |
| 21:01.07 | saint_ | Penguin: I'm using 1.8. Looks like confbridge is not a part of it.. |
| 21:01.12 | saint_ | i might have to upgrade. |
| 21:01.12 | Penguin | Yes it is. |
| 21:01.33 | Penguin | Maybe you didn't include it in your build, but confbridge is available in 1.8. |
| 21:02.02 | saint_ | i don t see anything containing the word confbridge in src/configs/ whether it's a file name or within a file. |
| 21:03.15 | file | it has no configuration file in 1.8, it was extremely basic |
| 21:03.47 | saint_ | I can tell you I have 1.8.15 cert 3, and its not here. |
| 21:03.53 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 21:03.53 | *** mode/#asterisk [+o angler] by ChanServ |
| 21:03.58 | Penguin | core show application ConfBridge |
| 21:03.58 | saint_ | I'll check out the meetme stuff .. |
| 21:04.15 | saint_ | Penguin: not here. |
| 21:05.52 | Penguin | http://pastebin.com/p4AZdCKT |
| 21:06.02 | Penguin | two examples |
| 21:06.45 | saint_ | Penguin: thank you sir. |
| 21:07.17 | Penguin | The example options may not be exactly what you want/need, but you can get the idea. |
| 21:07.28 | saint_ | Penguin: yup, got it. thanks a ton. |
| 21:07.38 | saint_ | Penguin: i think i m going to upgrade my 1.8 to 11.x |
| 21:07.45 | saint_ | does it make sense ? |
| 21:08.01 | Penguin | Do you need some feature of 11 that you don't have in 1.8? |
| 21:08.18 | saint_ | nah. i don t think so. |
| 21:08.22 | Penguin | Then it makes no sense. |
| 21:08.42 | saint_ | i just want to make sure that whatever i m doing now is going to work on the next release, when 1.8 will not be supported anymore.. |
| 21:08.52 | Penguin | You've got several years. |
| 21:08.59 | Penguin | ~versions |
| 21:08.59 | infobot | Asterisk versions and their support levels are documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 21:09.23 | saint_ | and i have other systems running 11.x ... i'd like to have them all at the same release, so i don t go through one version that has an option which would not be present in the other system.. |
| 21:09.29 | *** join/#asterisk hardwire (~hardwire@cl-101.anc-01.us.sixxs.net) |
| 21:10.44 | Penguin | You've got just under two years until 1.8 goes EOL, so you have time to work on the upgrade. |
| 21:12.00 | saint_ | i guess.. thanksa again for the examples. i ll try it now., |
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| 21:15.24 | ghost75 | whats the benefit of using confbridge against meetme? |
| 21:16.23 | mjordan | better performance, basic video conference, wideband audio, better AMI integration, more flexible options in user/bridge/menu profiles, configurable DTMF menus, generally awesome |
| 21:16.40 | Penguin | but not in 1.8 |
| 21:16.41 | ghost75 | k |
| 21:16.50 | Penguin | In 1.8, it isn't quite so awesome. |
| 21:17.19 | mjordan | Tjere |
| 21:17.24 | mjordan | There's no ConfBridge in 1.8. |
| 21:17.26 | ghost75 | wish i could tell the phone that a extension is dialed when conference is pressed |
| 21:17.31 | mjordan | Just like they never made two sequels to the Matrix. |
| 21:17.46 | Penguin | Then how do we use ConfBridge in 1.8 so easily? |
| 21:18.04 | mjordan | (it was a joke) |
| 21:18.27 | mjordan | but yes, it is very limited in 1.8. It was a proof of concept at the time for the Bridging Framework. |
| 21:18.34 | Penguin | Oh it's a joke alright. |
| 21:18.52 | mjordan | sense pessimism and goes back to JIRA |
| 21:20.32 | Penguin | Despite the very basic capabilities of ConfBridge in 1.8, I still use it sometimes. It does work, and that's what I care about. |
| 21:25.27 | Katty | drmessano: YOU |
| 21:30.04 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 21:33.28 | drmessano | Katty, YOU |
| 21:43.59 | Katty | :> |
| 21:44.03 | Katty | hugs drmessano |
| 21:44.19 | drmessano | hugs Katty |
| 21:44.21 | drmessano | Hows u? |
| 21:44.36 | Katty | bored, restless, and antsy. |
| 21:44.39 | Katty | how about you? |
| 21:49.28 | mjordan | Katty: want to test some Asterisk 12 hotness? |
| 21:49.34 | mjordan | you wouldn't be bored! |
| 21:50.22 | *** join/#asterisk Wiktor (wiktor@2001:470:9a06::1) |
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| 21:51.08 | Wiktor | is it possible to change some debug stuff or verbose options to show password that is sent to asterisk from voip phone ? Im getting 401s all the time |
| 21:51.38 | Wiktor | and register requests are being destructed |
| 21:53.04 | Katty | mjordan: maybe when i get done watching thise old family home video from the 1940s |
| 21:55.57 | newtonr | Wiktor, Not that I know of. Have you tried a really simple password, just for testing briefly? |
| 22:00.31 | Wiktor | its "cow" now |
| 22:00.37 | Wiktor | its drastically simple |
| 22:00.41 | Wiktor | newtonr: |
| 22:00.54 | *** join/#asterisk serafie1 (~erin@nat/digium/x-vqllqhgtkxgmnnor) |
| 22:01.57 | Wiktor | newtonr: AD-320 crap which could handle asterisk, then somebody ripped the firmware out of it, i put it back and it cannot connect. Owner of this phone doesn't remember how he configured that |
| 22:02.08 | newtonr | Wiktor, are you sure it's matching against the correct peer? Pastebin a log showing the attempted registration, turn up verbose and debug to 5 and turn on "sip set debug on" |
| 22:03.18 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 22:03.43 | drmessano | Katty, i'm ready to go home |
| 22:05.41 | Katty | drmessano: me too |
| 22:05.59 | drmessano | I'm going to see Trans-Siberian Orchestra |
| 22:06.05 | drmessano | I'm pretty excited |
| 22:06.56 | Katty | wow. jealous :> |
| 22:07.21 | drmessano | I love TSO... I saw Mannheim Steamroller last year. Pretty good show, but TSO is the bomb |
| 22:08.10 | drmessano | Speaking of which.. I'm out. Have a weekend! |
| 22:12.12 | Wiktor | newtonr: http://pastebin.com/q8EZvJkX |
| 22:12.23 | newtonr | Wiktor, thanks i'll check it out in a minute |
| 22:12.33 | Wiktor | newtonr: 1.8 - asterisk ip |
| 22:12.38 | Wiktor | 1.7 - phone ip |
| 22:23.23 | newtonr | Wiktor, so, looks like the phone isn't even trying to authenticate. It just repeats the register back to asterisk after the challenge. Another thing, possibly not affecting your issue is that you have NAT settings enabled for that peer, which shouldn't be since it is local; you may not have localnet= configured appropriately. |
| 22:24.06 | newtonr | Wiktor, re: the authentication, the phone should reply to the 401 with an "Authorization:" header, filled with the important bits |
| 22:24.23 | newtonr | I'll show you an example, one minute |
| 22:25.33 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 22:26.10 | newtonr | Wiktor, http://pastebin.com/7gLqYssq an example successful registration |
| 22:28.13 | Wiktor | could it be because bad registration port ? |
| 22:28.48 | Wiktor | because in the control panel of this phone there is no other place to put password than the PASSWORD place |
| 22:28.57 | Wiktor | and it is filled with "cow" |
| 22:29.18 | newtonr | is there an option for authentication? i mean despite a password setting, perhaps that is something that tells it to just try registering without authentication |
| 22:29.29 | newtonr | that=there |
| 22:29.46 | newtonr | i wouldn't think its a bad port, since we see the register attempts in your debug |
| 22:30.16 | newtonr | and looks like its going to 5060 both ways |
| 22:31.03 | Wiktor | may i give you telnet dump ? |
| 22:31.37 | newtonr | a telnet dump of what? |
| 22:31.48 | Wiktor | of phone configuration |
| 22:32.08 | newtonr | i don't know that particular phone, but if its written in somewhat plain english i'll take a quick look |
| 22:32.32 | *** join/#asterisk serafie (~erin@nat/digium/x-nlytqcjrxteeuvgy) |
| 22:34.30 | Wiktor | newtonr: http://pastebin.com/RBD4FWPQ |
| 22:40.31 | newtonr | looks fine to me. I don't see a specific option that sounds like it would disable authentication |
| 22:40.45 | Wiktor | meh sure |
| 22:41.21 | newtonr | have you disabled nat settings for that peer, or adjusted the localnet setting and tried again? |
| 22:41.31 | mjordan | just to double check, are you sure that the phone is receiving the 401 responses? |
| 22:43.09 | Wiktor | it gets transmission errors |
| 22:43.15 | Wiktor | but cant see any 401ws |
| 22:43.17 | Wiktor | *401s |
| 22:43.22 | newtonr | mjordan, the CSeq isn't incrementing on the phones second register, would that indicate that it's just trying the same registration over and over, and like you said, probably not receiving, or else ignoring the 401? |
| 22:45.06 | mjordan | not sure, but it sounds like the 401 isn't reaching the phone itself. Wireshark would probably show what is actually occurring. |
| 22:48.53 | ghost75 | is this not valid pattern? _x-cisco-serviceuri-cfwdall-. |
| 22:50.52 | ghost75 | <PROTECTED> |
| 22:51.14 | newtonr | Wiktor, i'd compare the wireshark pcap of working registration from another phone of the same model if you have that option |
| 22:51.45 | newtonr | and of course both of the phones configs, and their peers configs in Asterisk |
| 22:55.13 | Wiktor | only thing i have is csipsimple app |
| 22:56.05 | newtonr | ooh, yeah that won't be much of a useful comparison except for troubleshooting possible networking issues |
| 22:56.37 | Wiktor | its upseting that in this app only thing i have to enter is login and password |
| 22:56.39 | Wiktor | and ip |
| 22:56.41 | Wiktor | ;/ |
| 22:56.57 | Wiktor | and this voip phone is piece of crap. |
| 22:57.29 | newtonr | in CSipSimple if you hit menu button and go to settings there is a lot more |
| 22:57.35 | newtonr | i have it on my android as well |
| 22:58.34 | newtonr | Wiktor, pastebin your sip.conf config |
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| 23:24.01 | ghost75 | what is [ ] in pattern doing when there is just one letter in it? |
| 23:24.49 | WIMPy | Make sure it's not interpreted as a wildcard. |
| 23:26.15 | ghost75 | ah lol |
| 23:26.24 | ghost75 | should prolly go to bed now |
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