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| 03:52.27 | AliRezaTaleghani | what is the meaning of: SIP Options : (none) |
| 03:52.48 | AliRezaTaleghani | what are are values it can accepts? |
| 03:52.48 | AliRezaTaleghani | in sip show peer xxxx |
| 03:52.49 | AliRezaTaleghani | ? |
| 04:01.28 | ChannelZ | I believe that comes from an OPTIONS message, the 'Supported:' header. But I'm not sure if it's only filled in during a call or what |
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| 07:06.30 | FluxiFlax2022 | Hi I do have a problem with early media, sometmes a device does send traffic to asterisk before asterisk has allocated a port for rtp media, and one way audio happens, the main reason being that the Linux/Asterisk server returns ICMP PORT Unreachable...any input or ideas ? |
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| 07:51.30 | Ice_Strike | Hello |
| 07:51.49 | Ice_Strike | Sometime the recorded files are over lappping - what is causing this? |
| 07:52.28 | Ice_Strike | Like like one side is delayed so you hear the two parties talking on top of each other |
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| 10:42.17 | Adrian_Spark | Hello, could someone please help me with the following problem: I need to get the real IP address from an AGI script. Using SIPCHANINFO(peerip) or SIPCHANINFO(recvip) I receive an IP address from the SIP message, but I need the real public IP. If using Xlite dialer I see the local asterisk IP instead of the xdialer IP. If using pap2t or other phones I see the real IP. I am using Asterisk 1.4.29. |
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| 13:57.15 | jmetro | whenever i see "serafie" i think "selfie" |
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| 15:05.14 | jmetro | im trying to put asterisk on a citrix server so it can be run from a web client, but its having trouble finding their extensions.conf |
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| 15:12.40 | *** join/#asterisk killown (~killown@pdpc/supporter/student/killown) |
| 15:15.06 | Greenlight | Hmm... getting the following AMI hangup event, from a DAHDI ISDN30. "Cause: 41 Cause-txt: Temporary failure". Does that suggest a line problem ? |
| 15:15.27 | igcewieling | no, but you can verify by looking at PRI debug |
| 15:15.51 | WIMPy | Well, not at your end. |
| 15:17.19 | Greenlight | Oh, possibly an issue at the other end, or en route ? |
| 15:17.32 | WIMPy | Yes |
| 15:17.49 | WIMPy | And PRI debug will tell you the location. |
| 15:17.50 | igcewieling | does the number work when called from a cell |
| 15:17.50 | Greenlight | Seems to be happening on a few calls.... they're speaking away fine then, bam.. disconnected |
| 15:18.03 | Greenlight | Yes, the number is good. |
| 15:18.26 | Greenlight | Well it's a mobile phone, I guess if they went out of signal or something ? |
| 15:18.33 | Greenlight | Would I get that message back ? |
| 15:18.34 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:18.44 | WIMPy | I can't remember having seen that on the PSTN. I suspect there's a VOIP gateway somewhere that's giving it. |
| 15:18.57 | WIMPy | Possible. |
| 15:19.11 | Greenlight | This is a Virgin Media 4 span ISDN30... |
| 15:19.20 | Greenlight | No VOIP gateway |
| 15:19.49 | WIMPy | I'd expect subscriber absent or destination out of order, but who knows... |
| 15:19.56 | Greenlight | Yes |
| 15:19.58 | WIMPy | Well, not at your end. |
| 15:20.02 | Greenlight | I've not seen these before |
| 15:20.33 | Greenlight | Ok, will see if I can tell how frequently it's occuring then take a look at the pri debug |
| 15:21.15 | WIMPy | Someone should add a variable containing the location. |
| 15:21.34 | Chainsaw | Greenlight: It was Virgin Media that merged their landline & cellular networks isn't it? It would make sense that they have fairly custom events to say "GSM signal lost". |
| 15:22.07 | Greenlight | yes, however, in this case the mobile number is on the EE network, not the VM mobile network |
| 15:23.34 | igcewieling | Greenlight: "the PSTN" does randomly fail calls. We always redial after a 1 second delay if we get "weird" hangupcauses |
| 15:24.59 | Greenlight | I'm sure this has just recently started though - calls certainly shouldn't be getting cut off mid conversation |
| 15:25.15 | igcewieling | In mid conversation, no they should not |
| 15:25.21 | *** join/#asterisk killown (~killown@pdpc/supporter/student/killown) |
| 15:25.29 | igcewieling | start opening tickets with your telco |
| 15:25.34 | Greenlight | if it was just as we dialled it, then that'd not be so much of an issue |
| 15:25.38 | igcewieling | send them pri debug too |
| 15:26.01 | Greenlight | Yea, I've got the AMI output logged, so gonna work out how many times it's occuring then grab PRI output. |
| 15:26.06 | Greenlight | Sounds like a telco issue though |
| 15:26.12 | Greenlight | My customers telco. Great. |
| 15:26.18 | Greenlight | Thanks :) |
| 15:26.32 | igcewieling | They will tell you "it is a cell phone we don't open a ticket on that" and close the ticket. |
| 15:27.16 | Greenlight | I'll hope I find a landline example :) |
| 15:27.54 | WIMPy | You really want to know the location. |
| 15:27.58 | Greenlight | I wonder if I'll see alarms on the circuit |
| 15:28.11 | Greenlight | WIMPy: I'd see that in the PRI debug ? |
| 15:28.15 | WIMPy | Yes |
| 15:29.21 | WIMPy | And I've been arguing that we really need it in the dialplan as well. Some cause have very different meanings depending on the location they come from. |
| 15:29.52 | *** join/#asterisk ChadAragorn (~ChadArago@63-235-131-194.dia.static.qwest.net) |
| 15:30.20 | Greenlight | Does ${HANGUPCAUSE(${CDR(dstchannel)},tech)} perhaps work? |
| 15:31.08 | WIMPy | To do what? |
| 15:31.25 | Greenlight | Was thinking that might perhaps tell the location |
| 15:31.44 | Greenlight | I used it on SIP channels to grab extra detail of the hangup |
| 15:31.51 | WIMPy | Nope. You'd need another variable for that, like e.g. HANGUPLOCATION. |
| 15:31.55 | *** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh) |
| 15:31.57 | wasanzy | hi |
| 15:32.19 | wasanzy | how do u monitor asterisk service in nagios or zabbix? |
| 15:32.20 | Greenlight | And, we'd see, what Local/Remtoe ? |
| 15:32.27 | Greenlight | wasanzy: I use zabbix |
| 15:32.32 | WIMPy | Or you just shove both bytes in to one like CAPI applications displayed 4 digit hex codes. |
| 15:33.07 | wasanzy | Greenlight: how are u doing the monitoring and what and what are u monitoring? |
| 15:33.25 | WIMPy | local user / local private network / local public network / transit network / and back with remote instead of local. |
| 15:33.31 | Greenlight | SNMP. Channel useage mainly. |
| 15:34.55 | Greenlight | I play with DAHDI so in frequently can't even remember where to check for alarms |
| 15:35.07 | wasanzy | Greenlight: I will b happy if you can help me with the OIDs |
| 15:35.37 | *** join/#asterisk bytemaster (~ewr@host81-150-217-168.in-addr.btopenworld.com) |
| 15:38.17 | *** join/#asterisk freeedrich| (friedrich@perplexa.be) |
| 15:40.31 | Greenlight | DAHDI/i1 means span 1 - yes? |
| 15:40.46 | WIMPy | yes |
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| 15:48.58 | Greenlight | Log file's gonna grow quickly that's for sure |
| 15:50.03 | wasanzy | Greenlight: any help? |
| 15:51.41 | Greenlight | We just followed a tutorial online..., |
| 15:52.10 | Greenlight | YOu do an SNMP walk iirc |
| 15:52.13 | Greenlight | And it gives the OID's |
| 15:52.39 | wasanzy | Greenlight: can you give me link to the tutorial? |
| 15:53.32 | WIMPy | They are also hidden in the docs somewhere. Or in the contributiuons? |
| 15:53.33 | Greenlight | Think it was http://www.voip-info.org/wiki/view/Asterisk+SNMP although some of it is slightly dated |
| 15:54.19 | wasanzy | ok |
| 16:04.30 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 16:05.30 | Greenlight | http://pastebin.com/uN0ZsH50 thoughts? |
| 16:05.40 | Greenlight | (PRI debug of failed call) |
| 16:10.43 | igcewieling | I think you should send it to your telco |
| 16:14.20 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 16:16.42 | WIMPy | Errrrr |
| 16:17.06 | WIMPy | That's all transmit direction. |
| 16:17.42 | WIMPy | And that call has not been answered. |
| 16:17.56 | Greenlight | Yea - I'm hunting for one that's been connected just now |
| 16:18.06 | WIMPy | But it seems like your Asterisk decided that it didn't want the call to go on. |
| 16:18.15 | Greenlight | Hmm... how odd why would it do that |
| 16:18.25 | WIMPy | NFI. |
| 16:18.49 | WIMPy | Dis I already mention today that I don't trust libpri? |
| 16:19.02 | Greenlight | ^^ |
| 16:19.03 | WIMPy | Must be a timeout. |
| 16:19.15 | igcewieling | Greenlight: Asteisk libpri and dahdi versions? |
| 16:19.33 | Greenlight | I updated dahdi last night, let me check exact version numbers |
| 16:19.37 | WIMPy | Not getting any form of reply for 13 seconds surely is not normal. |
| 16:20.17 | Greenlight | DAHDI Version: 2.7.0 Echo Canceller: HWEC. libpri version: 1.4.14 |
| 16:20.41 | Greenlight | Oh - so in that PRI we get no reply after trying to initiate the call? |
| 16:20.45 | Greenlight | *PRI debug |
| 16:20.53 | WIMPy | Correct. |
| 16:21.15 | Greenlight | How can you tell the direction ? |
| 16:21.18 | igcewieling | Huh? You said this happens in the MIDDLE of the call. |
| 16:21.37 | WIMPy | So canelling the call fron your side would seem like an option, But what it tells you (and the remote party) is bullshit off course. |
| 16:21.42 | Greenlight | Yes - it does, certainly the ones we had it reported on and I looked at initially |
| 16:22.02 | WIMPy | There's a > or a < at the beginning of the debug lines. |
| 16:22.04 | Greenlight | However, the first one's I had a look for in the trace were not mid-call it seels |
| 16:22.07 | Greenlight | *seems |
| 16:22.12 | igcewieling | then it is unlikely you are getting no reply when trying to initiate the call. |
| 16:22.13 | Greenlight | Ahh, yes, the arrow |
| 16:22.34 | igcewieling | Greenlight: Are you SURE you don't have a SIP issue which is causing Asterisk to send that? |
| 16:23.08 | WIMPy | That doesn't change the fact that his telco or line has an issue. |
| 16:23.31 | Greenlight | My internal exensions are SIP, but that side of the call continues fine after the error |
| 16:23.45 | Greenlight | Let me try and find one where the call actually cuts off |
| 16:24.21 | igcewieling | I suspect your users are lieing to you and the call is failing when being setup. |
| 16:24.50 | WIMPy | Unfortunately I'm not sure if we see everything there, but as I don't see *anything* being reveiced at all, I'd expect the message to be retransmit. |
| 16:25.14 | Greenlight | igcewieling: The users I don't trust. However the AMI logs, I do :) |
| 16:25.36 | Greenlight | And the AMI logs show the call being connected for some minutes before the hangup with that reason |
| 16:26.00 | Greenlight | However, since I enabled PRI debug, the only calls being hungup with that reason don't seem to have connected first |
| 16:27.13 | WIMPy | wonders if that grep thing will hide things from us. |
| 16:27.31 | Greenlight | goes to find out... |
| 16:28.03 | WIMPy | But if your pastebin is correct, there's somethign VERY fishy. |
| 16:29.46 | Greenlight | Perhaps I can't use the thread number to grep like that... |
| 16:29.57 | jmetro | apparently wi-pipes break by being turned on |
| 16:30.14 | Greenlight | Now, how to I see which messages are associated with what channel |
| 16:31.04 | WIMPy | That's what the call referecnce is for. |
| 16:31.15 | WIMPy | That IS the call. |
| 16:31.34 | Greenlight | Hmm actually looking at other calls... I see ">" and "<" under the same ref. |
| 16:33.41 | Greenlight | TEI=0 Call Ref: len= 2 (reference 6313/0x18A9) (Sent from originator) <---- that ref? |
| 16:34.06 | WIMPy | yes |
| 16:34.24 | Greenlight | Going to be a royal pain to piece that together -- is a busy log ;/ |
| 16:35.35 | WIMPy | you could turn on intense debug and grep for the call ref in the hex dump. |
| 16:35.59 | Greenlight | Each line will have the ref? |
| 16:36.19 | WIMPy | No, but you only need the hex dump and that will contain it. |
| 16:36.34 | Greenlight | Ahh okay |
| 16:36.42 | WIMPy | But you don;t get the full hex dump with normal debug. |
| 16:37.04 | WIMPy | Are the bits documented somewhere? |
| 16:37.13 | WIMPy | can't remember :-( |
| 16:37.56 | WIMPy | usually only uses hex debug. |
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| 16:41.34 | jmetro | i hex debug my calls, dont you? |
| 16:41.35 | Greenlight | Hmm.. so this is more of a picture of the disconnection: http://pastebin.com/dvZgDc4v |
| 16:43.19 | Greenlight | Ahh.... there's that Location you mentioned |
| 16:43.33 | Greenlight | Location: Network beyond the interworking point |
| 16:44.28 | WIMPy | Yes, so that call seems to have left the PSTN as I suspected. |
| 16:44.55 | Greenlight | So, time to start arguing with telco |
| 16:45.44 | WIMPy | I guess they found a cheap chinese ITSP to route their calls *eg* |
| 16:45.57 | Greenlight | ^^ |
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| 17:05.56 | przerull | is it possible to execute a dialplan function from fastagi? |
| 17:06.26 | igcewieling | przerull: I can think of no reason you could not. |
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| 17:08.22 | przerull | igcewieling: I ask because there wasn't an agi command documented for that. exec would have seemed plausable but the docs https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+AGICommand_exec |
| 17:08.33 | przerull | claim it's for applications (not dialplan functions) |
| 17:09.01 | igcewieling | Dialplan functions are dialplan variables for almost any usage |
| 17:09.15 | igcewieling | hint: Dialplan VARIABLES |
| 17:09.52 | navaismo | What does asterisk 1.8 t the SIP MESSAGE, ignore it? |
| 17:12.51 | Penguin | przerull: You don't really "execute" functions. |
| 17:13.12 | przerull | hmmm thanks. so i should think of "CHANNEL(hangup_handler_wipe)" as a single variable whose name contains parentheses. correct? I had been thinking of them as keys to a dictionary object. |
| 17:15.09 | Penguin | That isn't an unreasonable way to think of it. |
| 17:17.02 | igcewieling | example: $agi->set_variable("ARRAY(CDR(account_sid),CDR(route_sid),CDR(omni_id),CDR(ani),CDR(dnis))", "{$route["account_sid"]},{$route["sid"]},{$route["omni_id"]},$ani,$dnis"); |
| 17:17.15 | Greenlight | Just caught a call which had connected, and then failed. Seems for sure this must be a telco issue: http://pastebin.com/iKsHa78A |
| 17:20.23 | igcewieling | Are you SURE your switchtype is correct? If it isn't much stuff will work, some won't. |
| 17:21.13 | Greenlight | Yea, pretty sure |
| 17:21.19 | Greenlight | It's been working okay for like 18 months |
| 17:22.28 | Greenlight | euroisdn i have it set to |
| 17:22.41 | przerull | thanks everyone |
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| 17:39.54 | navaismo | This sip debug output can be interpreted like a loop inside asterisk. A SIP MESSAGE is received, asterisk want to authenticate, The sip message is received with the challenge then asterisk accept the msg and try to send but when it try to send it to the 5005 peer seems like asterisk its sending to itself because the unknown peer at line 134 of the pastebin |
| 17:39.58 | navaismo | http://pastebin.com/AiV0J1i8 |
| 17:41.11 | navaismo | I can't take down this issue for sending sip messages :S |
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| 17:59.31 | jmetro | Verizon are huge wee-wee's for disallowing VOIP |
| 18:00.30 | igcewieling | jmetro: I noticed that on 4g |
| 18:00.37 | igcewieling | Maybe on 3g too, not sure. |
| 18:01.40 | jmetro | igcewieling: its on both networks |
| 18:05.48 | igcewieling | jmetro: have you tried VPN to get around it? I could not get it to work with tls |
| 18:07.02 | jmetro | I didnt try that actually |
| 18:08.14 | igcewieling | jmetro: a VPN was not the first thought? Surrender your geek card, please. |
| 18:08.30 | jmetro | I cant make changes to our VPN during business hours \o_o/ |
| 18:09.20 | jmetro | i just thought it was odd, considering i was hooked in via Wi-Fi and it wasnt working off though either |
| 18:09.31 | jmetro | maybe linphone isnt good anymore |
| 18:11.17 | igcewieling | they can't block VPN or they won't have any customers left |
| 18:11.36 | jmetro | nah theyd have plenty of desparate Iphone people forking over 300$ a month |
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| 18:53.02 | saxa | hello, i was looking at what is the minimum version of sqlite3 for asterisk 11.5.0 ? |
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| 18:54.44 | saxa | i have sqlite 3.6.23.1 on my system and its complaining that i have not sqlite |
| 18:55.10 | saxa | tried also to compile it --with-sqlite=/usr/bin |
| 18:55.25 | saxa | but its still errors with the same error message |
| 18:55.29 | navaismo | you need the devel packages |
| 18:55.38 | saxa | i have the includes in |
| 18:56.14 | saxa | navaismo: i use slackware, sqlite package have all includes in |
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| 18:59.37 | saxa | /usr/include/sqlite3.h |
| 19:00.00 | MauriMant74 | hello guys and gals... I have asterisk 1.8 runnign in a rt-n16.. I was wondering if this hardwre is good for asterisk 11 |
| 19:01.12 | MauriMant74 | I guess many people uses this same Asus router to run asterisk |
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| 19:06.19 | MauriMant74 | bump |
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| 19:09.54 | fakhir | ½$wü |
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| 20:00.11 | soFh | hello all |
| 20:00.51 | soFh | is it possible to playback a file while dialing the sip peer untill answered by the SIP PEER ? |
| 20:01.06 | soFh | what i am doing is first it plays file then dials , |
| 20:01.16 | soFh | while i want to play the file whilte its dialing the sip peer |
| 20:01.55 | [TK]D-Fender | use it as then only file in a folder used by a MoH class and specify it in your dial |
| 20:02.29 | jmetro | anyone had luck getting asterisk running on citrix? I've got everything except g722 working |
| 20:03.25 | [TK]D-Fender | and what are you running on it? |
| 20:04.15 | soFh | Moh...ok in that case i should disable sip-183 by dialed Peer ? |
| 20:04.27 | soFh | cause i want my own file announcement till its answered |
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| 20:05.27 | [TK]D-Fender | sof I just told you what to do |
| 20:05.51 | soFh | can you help me by a little example in dialplan |
| 20:06.02 | soFh | as i tried with StartMusiconHold() but that didn't worked for me |
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| 20:13.34 | igcewieling | soFh: put the CLI output of a failed call on a pastebin, unless you are using FreePBX. |
| 20:13.36 | igcewieling | ~pb |
| 20:13.37 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 20:13.51 | soFh | i am directly on asterisk |
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| 20:16.37 | navaismo | soFh, option m in dial cmd |
| 20:16.49 | navaismo | more help: core show applicatyon dial |
| 20:22.15 | soFh | thanks a lot navaismo and others |
| 20:22.20 | soFh | m worked perfectly as i wanted |
| 20:22.21 | soFh | Thanks again |
| 20:22.33 | soFh | Dial(SIP/peer/${EXTEN},32,m(default)) |
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| 20:28.13 | markusl | hi |
| 20:29.39 | markusl | a quick question regarding AMI: can i / should i send a new action while waiting for the result of another action? |
| 20:30.01 | markusl | i think the docs on voip-info contradict: |
| 20:30.38 | markusl | " Only one action may be outstanding at a time. " |
| 20:30.40 | markusl | but: |
| 20:31.07 | markusl | That way the client can easily match Action and Response packets while sending Actions at any desired rate without having to wait for outstanding Response packets before sending the next action. |
| 20:31.19 | markusl | ? |
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| 20:49.29 | WIMPy | saxa: You need the package from -current. |
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| 20:51.22 | WIMPy | markusl: Some actions will first only give you an acknowledgement and later a result. |
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| 20:56.59 | markusl | ah ok. |
| 20:57.06 | markusl | and how do those differ? |
| 20:58.22 | WIMPy | i think that's the naming used in the replies. I'd have to look at the wiki myself for details. |
| 21:05.24 | markusl | WIMPy: ok, thanks. i'll find it if it's on the wiki. |
| 21:12.52 | WIMPy | Err, wiki.asterisk.org, that is. |
| 21:17.32 | markusl | yep |
| 21:18.42 | markusl | woah. slow chat. (: |
| 21:18.55 | markusl | kthxbye |
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| 21:57.58 | WIMPy | So Polycom sold the DECT department again? |
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