| 00:03.40 | dwayne | when using SIP TLS / SRTP with Asterisk 1.8, shouldn't the SDP m= line use SRTP instead of RTP ? |
| 00:04.23 | dwayne | I have a a=crypto attribute, but I thought the media line would be different as well |
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| 02:42.11 | n8ideas | does anyone know if it's possible to kill the outbound NOTIFY when realtime peers are reloaded? |
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| 04:13.38 | stanlly | hi |
| 04:15.09 | ChannelZ | AHOY! |
| 04:15.16 | Penguin | Cookies? |
| 04:15.31 | pabelanger | Hello, yes? This is dog |
| 04:16.37 | stanlly | I need some help regarding Digium Wildcard TE110P T1/E1, card and cable are ok. I don't see any troubles with interupts. But the link never gooes up. |
| 04:16.55 | ChannelZ | pabelanger wins |
| 04:19.22 | stanlly | the card itself is detected and configured correctly but I insert an isdn cable it goes green and then goes to yellow alarm and up and down |
| 04:21.08 | stanlly | using dahdi-linux-complete-2.6.2+2.6.2, libpri-1.4.14 and asterisk-1.6.2.24 |
| 04:23.08 | stanlly | if I enable bri debug I only see outgoing frames. any ideas? I was thinking that it's a card issue but clearly it's not I just verified it in another box and it's working. |
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| 04:28.15 | ChannelZ | sorry I don't know ISDN. But I'd take a wild guess something _isn't_ configured right, that the control channel is wrong or something.. |
| 04:29.05 | stanlly | in dmesg I see dahdi: HDLC Receiver overrun on channel WCT1/0/16 (master=WCT1/0/16) |
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| 06:49.57 | lnb | in process of sending fax, why would cli> fax show sessions show 0 fax sessions when its sending? |
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| 07:25.29 | ironoxid | Hi, I've a problem with DID, it won't work with common phones (but it works from mobile phones)... overlapdial=yes, immediate=no, certified-asterisk-11.2-cert2, dahdi-linux-complete-2.7.0.1+2.7.0.1, libpri-1.4.14 |
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| 08:45.17 | _omer | I use "ConfBridge" , Calls are getting recorded (According to settings in confbridge.conf) |
| 08:45.53 | _omer | I want to use my own file_name for recorded calls. |
| 08:49.17 | kaldemar | _omer: what stops you? |
| 08:50.39 | _omer | tried some configs that I found on web. but nothing works. |
| 08:50.51 | kaldemar | _omer: you can define it with the record_file option. as the sample config says, it's better done in dialplan instead of a static definition in confbridge.conf. |
| 08:52.05 | _omer | confbridge.conf has mentioned it but I dont see any sytanx example |
| 08:53.01 | kaldemar | it tells you the function name. did you see what "core show function CONFBRIDGE" says? |
| 08:54.03 | _omer | I have already used |
| 08:54.41 | _omer | exten => _x.,1,set(CONFBRIDGE(user,announce_join_leave)=no) |
| 08:54.59 | _omer | ok let me check core show funtion CONFBRIDGE |
| 09:12.26 | _omer | kaldemar: set(CONFBRIDGE(bridge,record_file=/var/spool/asterisk/monitor/test.wav) does not work |
| 09:13.00 | _omer | let me try again |
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| 12:25.58 | _omer | I used ... Set(CONFBRIDGE(bridge,record_file)=testing.wav) but file saved as testing-1381833944.wav .. May I know how to fix this record_file appending thing? |
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| 13:23.23 | Vann | Does anyone know what could cause "continuous ringing"? I call an extension and it picks up fine, but every so often it doesn't pick up and it just rings and rings. |
| 13:27.29 | phix | Vann: misconfiguration could cause that |
| 13:27.55 | Vann | in the dialplan? |
| 13:29.22 | phix | possible |
| 13:29.28 | phix | do you want to pastebin it? |
| 13:29.35 | phix | redact it first of course |
| 13:29.50 | phix | if it contains sensitive information |
| 13:33.20 | Greenlight | Perhaps if no one was there to answer it? |
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| 13:41.50 | mirela666 | Hello, does GHotoif(condition) supports multiple conditions separated with AND/OR? |
| 13:42.28 | mirela666 | like Gotoif($[a>1 || a<0]?true:false) |
| 13:42.44 | Faustov | I guess you could nest another if inside "true" |
| 13:43.00 | [TK]D-Fender | mirela666: yes, with a SINGLE | |
| 13:43.07 | mirela666 | yeah that is always a solution but to |
| 13:43.17 | mirela666 | thx TKD |
| 13:43.24 | [TK]D-Fender | mirela666: & / | |
| 13:44.30 | mirela666 | [TK]D-Fender: thx |
| 13:44.33 | mirela666 | Faustov: thx |
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| 14:38.04 | nunne | Using Grandstream GXP2124 with Direct Pickup marked as *9 on the phone (and having logic for it in the devices context) doesn't seem to work using GS BLF. I see no dial logic coming in on *9 (but it DOES pickup the channel)... But dialing *9EXTEN will work as intended (i see the dial logic). |
| 14:38.16 | nunne | Anyone have any idea what might be going on here?! o.O |
| 14:39.51 | [TK]D-Fender | Show us the actual debug. Also I see no relationship with BLF for this |
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| 14:43.53 | nunne | I have notifycid = ignore-context |
| 14:45.48 | nunne | [TK]D-Fender: since i have *9 as call pickup in the phone config you might think I should see the same verbose messages as dialing just *9202 for example |
| 14:46.23 | [TK]D-Fender | waits for the requested backup |
| 14:46.28 | nunne | but no. the only verbose I see when using the BLF button for this is Channel SIP/asp-XXXXX answered SIP/43243243223 |
| 14:46.48 | nunne | so it seems to go directly and answering the Channel. Not passing any of the call logic |
| 14:46.56 | nunne | dialplan logic i mean |
| 14:47.41 | nunne | so manually dialing *9202 works flawlessly.. but using the BLF button which should dial just *9202 (as other phones does) answeres the SIP-channel directly without using the dialplan logic |
| 14:48.02 | nunne | which messes up my custom devstates for the extensions |
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| 14:54.08 | asteriskmonkey | im running asterisk 10 and having issues with queue bailing :/ i set timeout but its not observing it, and queue status is stating timeout on exit |
| 14:54.19 | asteriskmonkey | Queue(sales,nrR,,,120) |
| 14:54.21 | asteriskmonkey | is example |
| 14:54.28 | asteriskmonkey | tried with ,, too |
| 14:56.41 | asteriskmonkey | lol n option :) |
| 14:56.44 | asteriskmonkey | shoot in foot |
| 14:58.27 | Penguin | Check your queues.conf for timeout settings. |
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| 15:34.24 | zFleshMissile | I have an asterisk server acting as a call recorder in between a pbx and isdn30 line and when i dial out of the site the asterisk server only picks up 4 digits of the number and causes the call to fail. Does anyone know why that could be happening? |
| 15:38.37 | rrittgarn | not enough info. need to see the dial plan and or failure messages in a PB |
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| 15:50.53 | zFleshMissile | I can't get on the server at the moment to grab the dialplan sorry. If we assume the dialplan is fine, can you think of a configuration scenario that might be causing only the first 4 digits of a number being displayed when an outgoing call is attempted? |
| 15:51.10 | zFleshMissile | just even a stab in the dark could be helpful |
| 15:51.19 | Greenlight | It used to work? |
| 15:51.40 | din3sh | hey all |
| 15:53.54 | [TK]D-Fender | grabs one of his swords and turns out the lights... |
| 15:54.02 | zFleshMissile | Well originally I had a 3ish year old server there, it was working fine and hadn't been changed configuration wise in the last few years. The drive died and I replaced it with a new server, up to date asterisk and dahdi etc and now it doesn't work. |
| 15:54.34 | zFleshMissile | incoming works fine |
| 15:54.51 | din3sh | how you want to debug with no access on the server |
| 15:54.56 | Greenlight | RIght, so your assumption that the dialplan is okay, is based on, what exactly ? |
| 15:55.16 | zFleshMissile | the dialplan is exactly the same |
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| 15:55.21 | zFleshMissile | that never changes |
| 15:55.32 | din3sh | you might as well use one of [TK]D-Fender's swords |
| 15:55.51 | Greenlight | I've some wonderful chocolate tea pots... selling them cheap... intereseted ? |
| 15:56.23 | zFleshMissile | it's a simple pass through, doesn't do anything fancy with the numbers |
| 15:56.24 | din3sh | are the tea pots SIP based? |
| 15:57.00 | Greenlight | din3sh: No, silly, the SIP features are for the chocolate mugs! |
| 15:57.11 | Greenlight | SIP teapots, whatever next.. |
| 15:57.26 | din3sh | my bad :P |
| 15:57.45 | zFleshMissile | din3sh: I'm waiting for the server to be turned back on |
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| 15:58.27 | Greenlight | zFleshMissile: Presuming your running asterisk 11 on it now? And before the change, it was running 1.6? 1.4? |
| 15:59.36 | din3sh | i still have 2,3 prod servers running on 1.4 o.O |
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| 16:00.13 | rbd_ | hey guys... running 11.4 ... I have a System() command in my dialplan that takes a channel variable and saves it to a database (by running the 'mongo' commandline utility with an insert command specified). This works fine, but that channel variable is being truncated down to 237 characters...it's around 330 characters total. any ideas why? (the variable is set via a .call file Set() command...and |
| 16:00.13 | rbd_ | by examining the call file, the full string is specified there) |
| 16:00.28 | zFleshMissile | sec, looking now |
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| 16:00.54 | [TK]D-Fender | rbd_: * itself has a var length limit |
| 16:01.36 | rbd_ | [TK]D-Fender: ok...so with Set(), it internally truncates the variable down to that length. ok... is that hardcoded or is it changable somewhere? |
| 16:01.52 | din3sh | zFleshMissile: your box is back from the dead? |
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| 16:02.27 | zFleshMissile | Greenlight: Old was using 1.4.26.2 I believe. New is using 1.8.20.0 |
| 16:04.25 | din3sh | have you converted/optimised the dialplan for * 1.8.x? |
| 16:05.55 | zFleshMissile | Yeah, its currently in use in a number of other sites, the exact same build and everything and have never encountered this issue |
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| 16:06.28 | Greenlight | [04:55pm] <zFleshMissile> the dialplan is exactly the same <-- So that's not true? |
| 16:06.47 | zFleshMissile | hmm |
| 16:06.52 | [TK]D-Fender | rdhard |
| 16:06.58 | [TK]D-Fender | rbd_: Hard |
| 16:07.14 | zFleshMissile | you've got me there |
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| 16:08.23 | din3sh | meh |
| 16:09.20 | zFleshMissile | sec |
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| 16:10.48 | lnb | Vann: i had that happen with an ITSP, they were not keeping call state. I would hang up my SIP phone (dialing my cell phone) and the cell phone would keep ringing and ringing. |
| 16:11.59 | lnb | anyone know why in CLI> fax show sessions , displays 0 when there is a fax being transmitted ? |
| 16:12.01 | Greenlight | That sounds more like a NAT type issue |
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| 16:13.09 | lnb | Greenlight: are you referring to what I just said |
| 16:13.10 | lnb | ? |
| 16:14.01 | din3sh | yeah the keep ringing thing might be a NAT issue |
| 16:15.10 | lnb | din3sh: might be, but as soon as i left that ITSP, the issue stopped. |
| 16:15.27 | lnb | and nothing was changed on my end here. |
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| 16:33.00 | lnb | what would have to be done with Asterisk 10, when fax goes through it is detected and turns on t.38 udptl ? |
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| 19:13.45 | sawgood | Hi, for CentOS 6.x (is there) RPMs of Asterisk 1.8.x available (and if so) do you know the latest versions supported with RPM/yum? |
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| 19:18.31 | malcolmd | /join #freepbx |
| 19:18.34 | malcolmd | hah |
| 19:18.35 | malcolmd | sorry ;) |
| 19:18.58 | skrusty | fail ;) |
| 19:19.04 | malcolmd | indeed |
| 19:21.55 | sawgood | Yeah, I've got this Asterisk only box (for voicemail) on CentOS 6.4 (reinstalled) from CentOS 32-bit 5.9 (Final) (And) it will only ever have Asterisk only (and) I wanted to keep it managed via yum updates |
| 19:22.29 | sawgood | adding the repos to test now (to see what is available) from Digium/Asterisk for 6.x 64-bit builds |
| 19:24.34 | ayan | hehe. |
| 19:24.41 | ayan | <PROTECTED> |
| 19:24.43 | ayan | OOPS! |
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| 19:32.11 | przerull | Hello, how do you force native mixing in a confbridge conference? |
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| 20:20.52 | ibercom | Hi, I have two realtime asterisk servers (Active-Active). How can I pickup a call from a phone A in Asterisk1 and the call from Asterisk2 in phone B ? |
| 20:25.14 | ibercom | Anybody know ? |
| 20:29.14 | rrittgarn | curious how you have active:active... or are you trying to create an active:active scenario? |
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| 20:31.11 | ibercom | The same db and 2 server with phones...The phones have register server and backup server. |
| 20:33.08 | ibercom | The phones can register on any server. The servers have PRI card. |
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| 20:34.47 | rrittgarn | ok so the phones then register to both, but when they try to say Box1,Phone1 tries to call Box1,Phone2 and phone two is registered to box2, you want it to go through like its registered right? |
| 20:37.33 | ibercom | The problem is when the call comes from a different server to the registration server. Box1 and Box2 can call Phone1 because both know the address (AOR) of phone. |
| 20:39.19 | ibercom | I need pickup the call from Box2, Phone1 dialplan is in Box1. |
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| 20:57.59 | rrittgarn | how are you updating registrations between the boxes? or are you dual registering the phones? |
| 20:58.14 | rrittgarn | pub/sub for registration and state changes? |
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| 21:08.07 | ibercom | Phones register in Box1 or Box2. Box1 or Box2 update the DB. Still any pub/sub. |
| 21:09.09 | ibercom | DB is mysql Master-Master. |
| 21:13.12 | rrittgarn | they update the realtime DB suchas MySQL or the actual system internal database? |
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| 21:35.43 | AlexForster | running v1.8.20.0 (from EPEL), and I'm finding it extremely slow to issue ~6 Getvar actions for each channel - is it possible to batch them together? or, does Getvar get significantly faster in asterisk 11.x? |
| 21:36.53 | WIMPy | Are you sure it's Asterisk? |
| 21:37.50 | AlexForster | whoops actually i'm running cisco call manager |
| 21:38.05 | WIMPy | o.O |
| 21:38.36 | AlexForster | should Getvar be fast? on a ~100 channel system i'm seeing upwards of 5 seconds between the request and the response |
| 21:39.12 | AlexForster | i'd be willing to double check my concurrency strategy if Getvar is supposed to be a lightweight operation |
| 21:39.53 | WIMPy | No, I wouldn;t expect noticable delays. |
| 21:40.18 | AlexForster | but i'm imagining issuing 600 Getvar's (100 channels * 6 variables) causing channel lock contention or whatever |
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| 21:57.58 | jpcansa | hello, is it possible to do vlan tagging with a softphone? does anybody know if thats viable? |
| 21:58.19 | WIMPy | it's called routing |
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| 22:15.52 | phix | jpcansa: Yeah it is called routing :) would be easy on a Linux and Mac OSX box, not so much on a WIndows box (unless you are using a server edition or have a 3rd party routing software installed) |
| 22:18.11 | jpcansa | phix: how is it done on windows xp or 7? |
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| 22:18.53 | jpcansa | phix: or how is it done on linux so i can understand better? |
| 22:23.26 | jpcansa | WIMPy: do you know how to? |
| 22:24.03 | WIMPy | You route te destination to the right device. Just as always. |
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| 22:25.05 | jpcansa | i dont know how to do it on windows |
| 22:27.15 | phix | jpcansa: well within Linux you setup a VLAN interface then use iproute2 (ip command) to specify that VoIP packets or packets to a particular network are routed via the VLAN interface |
| 22:28.40 | phix | if the network address of your phone VLAN is different from that of your standard LAN then you won't need to us iproute2 |
| 22:28.49 | phix | us = use* |
| 22:31.05 | jpcansa | yeah, in linux i can doit with multiple interfaces in the same nic |
| 22:31.08 | jpcansa | but in windows? |
| 22:31.31 | WIMPy | Read the manual that came with Windows. |
| 22:31.42 | phix | and in WIndows 7 use this --> http://stackoverflow.com/questions/47854/how-do-you-create-a-virtual-network-interface-on-windows |
| 22:31.48 | phix | you need to setup a virtual interface |
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| 22:37.00 | jpcansa | thanks phix, you pointed me in the direction i needed |
| 22:38.23 | jpcansa | WIMPy if you have the chapter of the win manual that have that let me know, thanks in advance |
| 22:39.00 | WIMPy | I have never seen a manual. |
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| 22:58.21 | jpcansa | WIMPy: LOL |
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| 23:00.17 | drkat | howdy |
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