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| 07:56.38 | juned | Hi all |
| 08:00.06 | ChannelZ | o/ |
| 08:00.46 | juned | How do i set caller_id name in asterisk orginate command |
| 08:01.00 | juned | its showing unknow in softphone |
| 08:01.15 | juned | ChannelZ: Hii |
| 08:04.45 | ChannelZ | not positive, you can try setting it prior to your Originate application.. Set(CALLERID(name)=Foo) |
| 08:06.01 | juned | Actually I am originating call from asterisk manager |
| 08:06.10 | ChannelZ | how is it the extension doing the Originate being launched? |
| 08:06.16 | ChannelZ | oh.. |
| 08:06.34 | robmal | Uhm, Originate has CallerID parameter. |
| 08:06.46 | juned | yes that i am setting |
| 08:06.53 | juned | but want to set caller name as well |
| 08:07.03 | robmal | blabla <1231311> |
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| 08:10.05 | juned | robmal: but how do i set it in asterisk manager ? |
| 08:10.43 | robmal | The same way you set other parameters? |
| 08:13.00 | juned | <PROTECTED> |
| 08:13.03 | juned | did this |
| 08:13.12 | juned | and its working |
| 08:13.25 | robmal | Awesome. |
| 08:15.37 | juned | robmal: thanks man,, when i tried same it didn't work but now it did |
| 08:15.40 | juned | :) |
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| 08:45.56 | stefan27 | On a sip channel called SIP_XYZ_123 I want to find the sip.conf peer codec-settings for XYZ through dialplan - is the best way to this is to extract XYZ from channel name and then call the function SIPPEER(XYZ,codecs)? |
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| 09:17.12 | jamesc | I cannot find the older versions < 9 of the Intel PP libs any ideas? |
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| 10:38.59 | defsdoor | Hi - I'm unable to orignate a call outbound via a Sangoma E1 card (inbound is all ok) The call gets a hangup request, cause 1 |
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| 11:17.04 | *** join/#asterisk modesto916 (bd5ac048@gateway/web/freenode/ip.189.90.192.72) |
| 11:17.33 | modesto916 | Hi everyone |
| 11:17.55 | modesto916 | I have a situation with asterisk regarding SDP negotiation |
| 11:18.45 | modesto916 | I have a user configured in users.conf with the following codec options |
| 11:18.49 | modesto916 | disallow=all |
| 11:19.01 | modesto916 | allow=ilbc |
| 11:19.06 | modesto916 | allow=ulaw |
| 11:19.09 | modesto916 | allow=alaw |
| 11:19.53 | modesto916 | I also have a GoIP gateway which supports ulaw, alaw, g723, g729, etc |
| 11:20.57 | modesto916 | when I try to use the SIP phone registered as the user previously mentioned to place a call through the gateway |
| 11:21.17 | modesto916 | asterisk tries to transcode ulaw (gateway) to ilbc (sip phone) |
| 11:22.03 | modesto916 | Wouldn't it be better if asterisk agreed in using ulaw or alaw? |
| 11:24.46 | modesto916 | If I set directmedia and directrtpsetup to yes, I have the desired behaviour. But I can't do this because I need call transfer for this client |
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| 11:42.37 | Gugge | modesto916: the problem is that codec between caller and asterisk is negotiated before anything is know about supported codecs to the callee |
| 11:51.11 | modesto916 | Gugge. That's sad. I just read a topic about it. |
| 11:51.49 | modesto916 | I know freeswitch has a way to delay codec negotiation, doesn't asterisk have anything similar? |
| 11:54.36 | file | Not currently, no. |
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| 12:02.18 | modesto916 | It gets even worse, because asterisk is not managing to transcode ilbc to ulaw correctly |
| 12:03.55 | modesto916 | On the phone side I just hear some robotic not-even-close-to-understandable sound |
| 12:04.03 | modesto916 | On the other side I hear nothing |
| 12:04.40 | modesto916 | If I use just ulaw on my user configuration, it works fine |
| 12:05.14 | mrfrenzy | maybe focus on fixing the transcoding problem then? |
| 12:09.10 | modesto916 | mrfrenzy I'm not confortable with transcoding. I'm using ESXi to host virtual pbxs for my customers |
| 12:09.32 | mrfrenzy | I understand, every cpu cycle costs |
| 12:09.59 | modesto916 | That's correct =) |
| 12:10.26 | modesto916 | If I use directmedia and directrtpsetup, it works fine |
| 12:10.26 | mrfrenzy | maybe you could have ulaw as default and somehow build up a database of targets who do support libc |
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| 12:12.34 | modesto916 | We're considering using just ulaw |
| 12:13.08 | modesto916 | The problem is that some clients are in remote locations. In fact, not so remote, they use the same ISP provider where our server is hosted |
| 12:13.45 | modesto916 | icmp latency is less than 30ms if the customer's network is operating correctly |
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| 12:51.15 | davlefouAMD | hi, how i made an mail like that : mailcmd=/usr/sbin/sendmail -t -f asterisk.pbx@domain -cc 'contact@domaine,webmaster@domaine.com' -A 'Content-type:charset="utf-8"'? |
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| 13:07.48 | stefan27 | if sip peer A calls asterisk, and asterisk calls Dial(SIP/peerB...) will the generated SDP offer to peer B by default only be based on peerB's sip.conf-settings, so that even if peer A's offer did not include video as a media-type, the SDP offer sent to peer B *will* include the video media type (if peer B has that configured)? |
| 13:10.38 | newtonr | stefan27, yes, it is two separate call legs. |
| 13:11.31 | newtonr | davlefouAMD, Your question isn't clear. What are you trying to do? |
| 13:14.35 | *** join/#asterisk Cuzner (~ccuzner@198.41.29.45) |
| 13:15.15 | stefan27 | OK thanks, i was just confused because i have two asterisks and one of them does not behave that way, but a colleague has put a lot of custom patches in the other one so i wanted to know default behaviour |
| 13:17.28 | stefan27 | the clean asterisk seems to behave as i described above -- if I don't want that behaviour, I can do some dialplan logic based on SIP_A's channel's data and set SIP_CODEC_OUTBOUND before dialing B? |
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| 13:52.04 | stefan27 | 'rtcp set debug on' does not seem to display rtcp packets with packet type 206 (RTCP Feedback message type: FIR) which asterisk has sent to my video softphone client according to wireshark, is it not supposed to? it does display a lot of Sender Reports though |
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| 14:05.57 | WIMPy | defsdoor: Are you in North America? |
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| 14:39.03 | pa | is there any way to catch all calls coming from one trunk? |
| 14:39.09 | pa | in extension.conf i mean |
| 14:39.37 | robmal | _XXX. |
| 14:39.42 | pa | because it seems that when i call the number i just registered with one provider, sometimes asterisk gets the callerid as that number, sometimes it doesnt |
| 14:40.09 | pa | robmal, but can i somehow specify i want to catch all calls only from a specific SIP trunk? |
| 14:42.14 | pa | ah okay i need to specify the context |
| 14:43.41 | pa | but can i have multiple contexts in the [general] block of sip? |
| 14:43.43 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 14:43.54 | [TK]D-Fender | robmal_XXX. <- that's already 4 charater minimum, 3 of those forcibly digits |
| 14:44.21 | [TK]D-Fender | pabut can i have multiple contexts in the [general] block of sip? <- no |
| 14:44.46 | [TK]D-Fender | parobmal, but can i somehow specify i want to catch all calls only from a specific SIP trunk? <- make a peer that actually matches the sender |
| 14:45.30 | pa | [TK]D-Fender, you mean the CID of the sender, right? |
| 14:45.59 | newtonr | Anyone have a grandstream GXP 2000 or 2020 in their possession who would be willing to help with a bit of Asterisk triage? |
| 14:46.21 | edong23 | what is the best way to handle fax from fax gateway out to a dahdi pri ? |
| 14:46.29 | edong23 | just dial dahdi/g0? |
| 14:46.33 | WIMPy | pa: Either you use a pattern ot the i extension. |
| 14:47.02 | [TK]D-Fender | pa: no, to matcht the PROVIDER who is sending you the call |
| 14:47.41 | [TK]D-Fender | WIMPy: "i" doesn't work for SIP |
| 14:47.50 | [TK]D-Fender | WIMPy: it'll 404 |
| 14:49.29 | pa | practical example: i registered an account with callcentric |
| 14:49.39 | pa | then i got some number associated with it |
| 14:50.01 | pa | and got my asterisk reigster with the callcentric account |
| 14:50.45 | [TK]D-Fender | Make a peer to match them |
| 14:51.50 | pa | ok, i try |
| 14:54.31 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-frxvzxjxwesixmfb) |
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| 15:03.25 | stefan27 | newtonr - not GXP2200? |
| 15:03.52 | *** join/#asterisk pjensen00 (~per@ip-69-178-218-71.far.ideaone.net) |
| 15:04.06 | stefan27 | I can ask the office people on monday if it's an issue still then, it's 5 pm on a friday now so everyone left but me |
| 15:04.07 | pa | so i havent managed yet. what i see is incoming calls like the following: Executing [s@default:1] wait("SIP/callcentric.com-00000010", "1") |
| 15:04.24 | pa | should i just match provider: callcentric then? |
| 15:05.29 | [TK]D-Fender | They are getting matched against your peer |
| 15:06.12 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 15:10.52 | pa | also, it somehow spawns this: Executing [s@default:1] background("SIP/callcentric.com-00000010", "demo-congrats") , although i have demo-congrats specified nowhere |
| 15:12.02 | [TK]D-Fender | pa: [s@default:1] <- OH YES YOU DO |
| 15:12.27 | [TK]D-Fender | pa: it is TELLING you where the call is landing. |
| 15:12.53 | *** join/#asterisk Pegasus_RPG (~Icedove@pool-71-102-242-103.snloca.dsl-w.verizon.net) |
| 15:14.11 | Pegasus_RPG | Hello. Does anyone recommend a VoIP softphone for Linux to replace Twinkle? |
| 15:15.52 | Pegasus_RPG | (With an * server naturally) |
| 15:16.12 | *** join/#asterisk kharwell (kharwell@nat/digium/x-balxyxmumhiqqgdk) |
| 15:20.10 | [red] | also interested in that question |
| 15:20.16 | [red] | i've yet to find a softphone i like |
| 15:20.18 | [red] | they all suck |
| 15:22.08 | [TK]D-Fender | Yup. |
| 15:22.18 | [TK]D-Fender | All softphones suck. Some only slightly less than others |
| 15:22.28 | pa | ok it was in extensions.ael |
| 15:24.31 | igcewieling | hugs his Polycom phone |
| 15:26.02 | [red] | polycom is pretty good. strangely a really like grandstream for hardware even though their stuff is made out of the cheapest chinese plastic i've ever seen |
| 15:26.30 | [red] | but honestly for softphones, i've been pretty happy with the free android apps |
| 15:26.36 | [red] | like csipsimple and zoiper |
| 15:26.49 | jamesc | What is it finding versions of IPP libs < 9 like pulling teeth! |
| 15:26.54 | mjordan | [TK]D-Fender: I should add a patch that auto-creates a default,s,1 extension if it doesn't exist that just plays back a "Don't configure your peers with the [default] context!" sound. |
| 15:27.04 | mjordan | goes back to other things. |
| 15:28.17 | [red] | the sip vicious botnets that hit servers love the default context |
| 15:28.20 | [red] | mine is pretty special |
| 15:28.28 | pa | ok, one step forward. i removed extensions.ael and extensions.lua, and now i get something different |
| 15:28.33 | pa | that is: [Sep 11 17:23:45] NOTICE[30654][C-00000001]: chan_sip.c:25450 handle_request_invite: Call from '' (204.11.192.161:5060) to extension 's' rejected because extension not found in context 'default'. |
| 15:28.46 | [red] | pa: are you writing your dialplan in lua? |
| 15:29.03 | pa | no, thats why i removed it. it seemed it got autoloaded |
| 15:29.10 | [red] | it indeed does |
| 15:29.29 | pa | this strange extension was landing on demo, and i dont have demo in my extensions.conf |
| 15:29.43 | pjensen00 | I'm currently using JITSI which is fine.... except there's no hotkey to dial. *grumbles about having to click for each test call* |
| 15:31.40 | [red] | i'm actually fairly surprised how little i hear about people writing dialplans in lua |
| 15:31.46 | [red] | i'm absolutely in love with it |
| 15:33.48 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 15:34.59 | davlefouAMD | newtonr, i try to send message to servarles users with an mailcmd. |
| 15:36.13 | Synthase_ | Lua appears to have fallen out of favor, quite some time ago. |
| 15:36.37 | newtonr | stefan27, nah, something in the 20XX series preferably 2000 or 2020 , it is for reproducing https://issues.asterisk.org/jira/browse/ASTERISK-25169 |
| 15:37.00 | newtonr | davlefouAMD, okay I didn't understand the issue |
| 15:38.57 | ChkDigit | Anyone know how I can force a hint to reload? Hints worked up to a power surge yesterday for years... I've restarted asterisk, rebooted phones, and still all my SIP hints say "State:Unavailable" |
| 15:39.10 | [red] | coming from a developer background, lua makes much more sense to me than .conf and .ael |
| 15:39.37 | *** join/#asterisk kharwell (kharwell@nat/digium/x-lwbmqnlyetwjyuue) |
| 15:39.37 | [red] | and i'd rather not monkey around with the core code |
| 15:45.05 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 15:50.24 | ChkDigit | I band-aid patched my own problem. Doing a dialplan reload, now allows the SIP devices to provide state info. |
| 15:50.30 | ChkDigit | Weird. |
| 15:51.52 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 15:57.13 | pjensen00 | Does this mean that the ARI module itself crashed? "asterisk[671]: segfault at 154 ip 00007feeb934acbc sp 00007fee96b22db0 error 4 in res_stasis.so[7feeb933d000+16000]" |
| 15:59.44 | *** join/#asterisk karelk (~karel@31.10.145.202) |
| 16:01.17 | WIMPy | Yes, Asterisk crashed. |
| 16:06.32 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 16:07.14 | pjensen00 | Yeah, I'm just trying to narrow down why |
| 16:07.36 | pjensen00 | it hadn't crashed in a while so I took out my debug build in this environment |
| 16:08.31 | pjensen00 | it was on a dial command so I'm sure I screwed something up |
| 16:09.01 | WIMPy | Someone else did as well :-) |
| 16:11.48 | Pegasus_RPG | loves his Snom 300. |
| 16:12.08 | Pegasus_RPG | But having a phone in my laptop is mobile business bliss |
| 16:20.57 | pjensen00 | Hrm, I think I can reproduce this. I am hanging up a call at approximately the same time I'm issuing a dial command. |
| 16:21.20 | Synthase_ | @Pegasus_RPG Desktop Linux or Windows on that Laptop? Need to setup one with my VPS Asterisk gateway today. |
| 16:25.18 | igcewieling | pjensen00: upgrade your asterisk version |
| 16:28.54 | [TK]D-Fender | pathis strange extension was landing on demo, and i dont have demo in my extensions.conf <- you had it in AEL |
| 16:29.12 | [TK]D-Fender | And you are not paying attention to where you are even pointing your peers to |
| 16:30.58 | pjensen00 | ah, found the issue. I think you're right wielding. Guessing it's the "hangup while bridging" ARI bug that was fixed. I have two instances and the crashy one is 13.4 not 13.5 |
| 16:38.27 | Pegasus_RPG | Synthase_: Linux, Debian Jessie |
| 16:50.36 | Pegasus_RPG | just got Linphone working |
| 16:50.44 | Pegasus_RPG | issues with TLS though |
| 16:51.06 | Pegasus_RPG | "Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca" |
| 16:58.26 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 17:13.31 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 17:17.18 | Pegasus_RPG | Ah fixed it. Needed the CA's chain file in my sip.conf |
| 17:32.53 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 17:32.53 | *** mode/#asterisk [+o file] by ChanServ |
| 17:44.27 | Synthase_ | @Pegasus_RPG Thanks, same. Giving it a go. |
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| 18:15.12 | roler | Our PBX has been up for 40+ days. No configuration changes have been made. All of a sudden, outbound calls are working intermittently. The error I am seeing is: dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion. Any ideas? Could this be on our end or carrier end? |
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| 18:59.43 | cmendes0101 | roler: dahdi show status |
| 19:00.22 | cmendes0101 | actually I can't remember if that is it. I haven't used dahdi in a while. I think you can do dahdi_cfg -vvvvv outside of asterisk also |
| 19:01.56 | igcewieling1 | dahdi_cfg will drop all active calls. |
| 19:02.35 | cmendes0101 | oh really? lol good to know thanks |
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| 19:07.18 | igcewieling1 | cmendes0101: you'd have figured it out pretty quick.ly 8-| |
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| 20:09.29 | *** join/#asterisk moe` (~moe@node-be028882.scarlet.cw) |
| 20:10.09 | moe` | hey guys, question for you... if I enable TLS on asterisk (sip.conf) does it use SSL over UDP connections, or only TCP? |
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| 20:24.49 | *** join/#asterisk Welog (~Balthazar@80.12.51.60) |
| 20:26.13 | Welog | hello everybody. may somebody could help me please ? all my peers and trunk are unreachable and asterisk is very slow to reload. have you any idea ? some peers get registering but few seconds later become unreachable.. |
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| 20:45.41 | pjensen00 | Welog: you'll have to be more specific on the issue and throw some kind of log into pastebin for peopel to look at |
| 20:46.24 | igcewieling1 | moe`: I'm not *sure* but I believe TLS is TCP only. |
| 20:46.37 | moe` | that's unfortunate |
| 20:46.54 | moe` | suppose I could do some ngrep'ing or tcpdump'ing |
| 20:47.17 | igcewieling1 | moe`: encrypted udp might be called something else, It has been so long since I worked with that stuff. |
| 20:47.29 | moe` | it'd be nice if it does use SSL over UDP (like OpenVPN, for example) |
| 20:49.29 | igcewieling1 | search for DTLS |
| 20:52.09 | moe` | cool, thanks |
| 20:52.19 | moe` | just set it up, did sip reload, it didn't complain |
| 20:53.02 | Welog | okay pjensen00 i can provide log without problem :), thank for the answer, i try a debug log : http://pastebin.com/9rQBfj9L |
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| 21:00.17 | Welog | and when i typing on the asterisk cli "reload", the command is executed 10 seconds later, before it was done immediately.. how can i see if all asterisk modules are loaded ? |
| 21:00.33 | pjensen00 | what's the server's load average? |
| 21:01.02 | moe` | ngrep shows its not using crypto |
| 21:01.07 | moe` | dammit |
| 21:01.15 | Welog | 23:00:11 up 3:16, 3 users, load average: 1,24, 1,33, 1,25 |
| 21:03.01 | pjensen00 | Hrm. |
| 21:03.31 | pjensen00 | Do you have debug/verbose turned up? If so, paste the output between the command sent/executed |
| 21:07.30 | Welog | http://pastebin.com/L5eq48F2 :) |
| 21:21.42 | Welog | do you have any idea pjensen00 ? i will leave and come tomorrow afternoon. that doesn't matter, i can wait to solve it |
| 21:22.04 | pjensen00 | I looked at it and didn't see anything that'd cause it to lag. |
| 21:22.12 | WIMPy | Are you sure it's Asterisk and not something else? |
| 21:22.14 | pjensen00 | There are many external influences that could cause the delay |
| 21:23.07 | Welog | yes i'm unfortunately :( |
| 21:23.38 | Welog | hum, i'm not but i suspect |
| 21:29.30 | Welog | i come tomorrow. thank to have search :) (i will thinking of this problem during the night haha) |
| 21:40.59 | *** join/#asterisk jlnt (~jl@2602:306:b8f0:9e0:695e:94ff:8bff:83da) |
| 21:47.04 | jlnt | So how many of yall like the new way that FreePBX is moving things? |
| 21:48.06 | cmendes0101 | #freepbx |
| 21:48.13 | jlnt | I know the channel :) |
| 21:48.23 | WIMPy | Nit sure how many in here have an ide what you're talking about. |
| 21:48.29 | jlnt | just wanted to talk to my buddies here and see what yall think |
| 21:48.35 | WIMPy | Not... |
| 21:49.37 | igcewieling1 | FreePBX is terrible, but it is far far less terrible than all the others. |
| 21:49.40 | jlnt | well I have been installing Asterisk on systems since 2004 and all of a sudden FreePBX starts to become the Web Manager and for a long time it was great but I am not liking the idea |
| 21:50.09 | cmendes0101 | The idea of a web manager or just freepbx? |
| 21:50.13 | jlnt | they are commercializing everything and making it to where some of the files cannot be manipulated |
| 21:50.19 | jlnt | freepbx in gen |
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| 21:50.45 | WIMPy | I second the doubts that this is the right channel, but if you want to try, you will probably have to explain what you're talking about. |
| 21:50.55 | cmendes0101 | good thing asterisk is opensource so you can make your own modifications |
| 21:51.00 | jlnt | lol I am just talking :) |
| 21:51.06 | jlnt | been a while since I have been in the chan |
| 21:51.06 | WIMPy | That's what they all do, isn't it? |
| 21:51.15 | jlnt | it was so quiet in here just wanted to start a topic |
| 21:51.22 | igcewieling1 | What a wonderful error message. ERROR 1064 (42000): You have an error in your SQL syntax [...] near '' at line 1 |
| 21:52.50 | DanQuinney | Post the query igcewieling1 and I'm sure one of us will be able to help |
| 21:54.32 | igcewieling1 | DanQuinney: I'll figure it out quickly enough, just thought the error was a tad funny. |
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| 21:57.04 | DanQuinney | <PROTECTED> |
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| 22:25.39 | kchehab | i have multi outgoing trunks ins the same colocation and these is no problem with ring back but , i have a ringback problem with a trunk in another colocation , please do you have any idea |
| 22:40.44 | kchehab | i set progressinband=yes |
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