| 00:00.00 | Samot | Now with some verbosity |
| 00:00.38 | Samot | Because the failed call was due to it not being found in the from-inside context |
| 00:01.08 | Samot | Lets see what * is doing with the call |
| 00:01.27 | file | sip:422 parses as a SIP URI with hostname 422 |
| 00:01.41 | file | since it has no user portion chan_sip would consider the dialed extension as 's' |
| 00:01.43 | justdave | it's all in there, it's already on verbose |
| 00:01.50 | justdave | the one that completes shows it getting processed |
| 00:02.04 | justdave | and what file says was my guess as well |
| 00:02.17 | Samot | Yup. |
| 00:02.26 | justdave | hence my initial question being polycom configuration because I think it's the phone doing it wrong |
| 00:02.30 | file | I have no touched Polycom phones in, er, many many years so I'm unaware of how to bend it to your will |
| 00:02.50 | Samot | What model? |
| 00:02.55 | justdave | IP550 |
| 00:06.32 | Samot | I would have to look at the 550s config files. |
| 00:07.33 | Samot | But i dont set anything really in them but user sip settings and theres been no complaints from end users about not beimg able to dial from the missed call list |
| 00:07.44 | Samot | Or recent calls, etc. |
| 00:07.59 | justdave | nothing even related in our config, so it must be something they have a bad default on that we need to undo. Looking through their example files now to see if they specified something related in their example |
| 00:08.32 | justdave | I ran our exising 3.3.1 config files through their upgrade/converter app, there's probably something they want specified in the config that's not in it. |
| 00:09.26 | Samot | Im not sure. Ive got polycoms on 4.1.1 |
| 00:13.12 | justdave | sees some callList related stuff in the example configs that isn't in ours, tries adding that to it |
| 00:15.50 | shido6 | are you using a hostname or ip in the polycom config - ? |
| 00:16.30 | shido6 | did I miss the polycom configs? |
| 00:26.56 | justdave | it's a hostname in the configs. |
| 00:27.13 | justdave | and I know that's correct because calls placed by dialing on the keypad work |
| 00:27.19 | justdave | (so dns is working and whatnot) |
| 00:27.42 | justdave | and it does actually connect to asterisk, it's just not sending the correct data in the INVITE header |
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| 01:17.35 | justdave | ok, so shido6 figured out that if you use an IP address to connect it works fine, the polycom is just getting goofy if you use a domain name. And only affects the call lists, not the rest of the phone operation (which is very weird) |
| 01:18.11 | justdave | so it's either a bug or a feature in the polycom firmware and figured out how to work around it |
| 01:27.38 | scv | a polycom bug? shocking |
| 01:50.32 | Samot | Does the polycom have DNS set? |
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| 02:17.34 | wyoung | wind 20 |
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| 07:48.41 | mirela666 | HI, |
| 07:49.05 | mirela666 | Has anyone experienced high CPU usage when having long calls on 1.8.13.0 |
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| 08:10.39 | slima | Why when I have same => n,GotoIf($[${REGEX("2433712[0-4][0-9]" ${CALLERID(num)})}]?ok:notok) 0243371200 is ok? |
| 08:13.18 | slima | oh, missing ^ |
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| 08:28.48 | Get_The_Fish | hello all, I'm looking to add a parameter to the SIP to field, much like the transport is appended to it, like this <sip:example@localhost:5060;transport=tls;messagetype=SMS>. |
| 08:29.03 | Get_The_Fish | so far I haven't been able to figure it out. |
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| 10:43.30 | davlefou | Hi, is it good : GotoIf($[${sercurite} = Sorti]... |
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| 11:15.03 | rcp | hi guys, can anyone tell me how I block a caller by number. |
| 11:17.24 | Rasputin3711 | http://stackoverflow.com/questions/33598924/asterisk-blacklist-a-number-by-country-code-or-area-code |
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| 11:31.18 | rcp | I did this 186698*CLI> database put blockcaller 001616418550 1 |
| 11:31.26 | rcp | hopefully the asshat won't keep calling. |
| 11:31.30 | rcp | thc |
| 11:31.31 | rcp | thx |
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| 13:42.11 | ntz | hello |
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| 13:43.01 | ntz | I have question about did in asterisk .... I've fax solution based on the t38modem and hylafax (on solaris) but it seems to me, that asterisk is devouring the DID (NDID) .... I have this Dial() cmd: |
| 13:43.25 | ntz | ``Dial(SIP/${EXTEN}@T38modem_mgec212,15,g);'' and reading this http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 13:43.45 | ntz | seems that only "g" flag is not enough - anyone has a clue what I am missing ? |
| 13:44.47 | Samot | Ntz: show a call and how * os devouring it. |
| 13:45.12 | ntz | Samot: minute, creating for you a paste |
| 13:48.45 | ntz | Samot: http://susepaste.org/view/raw/97196809 |
| 13:49.05 | ntz | as you can see, a 7:NDID is empty and I am not sure why |
| 13:49.34 | Samot | Show a real debug |
| 13:49.50 | Samot | Asterisk -rvvvvvvvvvv |
| 13:50.11 | Samot | Sip set debug on |
| 13:51.12 | ntz | ok |
| 13:53.53 | ntz | Samot: killed and started asterisk again - this is whole log from routing (successfully) fax via asterisk: http://susepaste.org/view/raw/3651951 |
| 13:55.43 | ntz | Samot: on the fax machine side it's still valid http://susepaste.org/view/raw/97196809 |
| 13:55.50 | ntz | si NDID is empty |
| 13:55.54 | ntz | s/si/so/ |
| 14:00.18 | Samot | ntz; I don't want a log. |
| 14:00.25 | Samot | I want a live debug of the call. |
| 14:00.29 | Samot | With the commands I gave. |
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| 14:03.05 | ntz | Samot: so just to run asterisk, connect to console (rasterisk), do `sip debug' and paste here what it will print when fax goes through ? |
| 14:03.24 | Samot | Asterisk should already be running. |
| 14:03.29 | ntz | sure |
| 14:03.39 | Samot | You just need to connect to the Asterisk CLI and get the verbosity up. |
| 14:03.54 | Samot | That's what asterisk -rvvvvvvvvvvvv does |
| 14:03.54 | ntz | ok |
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| 14:04.43 | ntz | minute ..... btw Samot ,, shouldn't do G(context^exten^pri) that according to this http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial ?? eg reminding, my dial line looks like this: Dial(SIP/${EXTEN}@T38modem_mgec212,15,g); |
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| 14:11.15 | ntz | Samot: http://susepaste.org/view/raw/22081906 |
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| 14:17.36 | ntz | Samot: I'm thinking of I have not properly a ``Dial(SIP/${EXTEN}@T38modem_mgec212,15,g);'' command ..... |
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| 14:20.52 | [TK]D-Fender | ? |
| 14:21.04 | [TK]D-Fender | Why are you even using "g"? |
| 14:21.17 | [TK]D-Fender | You care calling from one IAXmodem right to another one |
| 14:21.42 | ntz | nope, between two t38modems via asterisk |
| 14:22.07 | [TK]D-Fender | Sorry, meant to say T38 |
| 14:22.20 | [TK]D-Fender | but yes, that is what you are calling there. |
| 14:22.29 | [TK]D-Fender | So why would you even be thinking about using "g"? |
| 14:22.44 | ntz | no prob .... please note - I am not asterisk expert, I was using it cuz it was in hylafax docs |
| 14:23.02 | ntz | [TK]D-Fender: so shall I kick out "g" from Dial() ? |
| 14:23.25 | [TK]D-Fender | You can't tell us WHY you are using it. Do you not understand what it does? Do you see a reason to do this? |
| 14:24.39 | ntz | I've read it here also but I considered it harmless :: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 14:25.18 | [TK]D-Fender | Should it continue to do other things? |
| 14:25.29 | *** join/#asterisk KaliLinuxGR (~alexandro@188.117.250.184) |
| 14:25.42 | ntz | probably not, but this has nothing to do with my original question - missing DID (or NDID) |
| 14:25.44 | [TK]D-Fender | ntz> minute ..... btw Samot ,, shouldn't do G(context^exten^pri) that according to this http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial ?? eg reminding, my dial line looks like this: Dial(SIP/${EXTEN}@T38modem_mgec212,15,g); <- and what about this? |
| 14:26.05 | Samot | Sorry, I'm on a call now. |
| 14:26.25 | ntz | ^^ yep, that was legitimate question [TK]D-Fender .... I clearly don't understand what G option for Dial() does |
| 14:26.43 | [TK]D-Fender | "G(context^exten^pri): If the call is answered, transfer both parties to the specified context and extension. The calling party is transferred to priority x, and the called party to priority x+1. This allows the dialplan to distinguish between the calling and called legs of the call (new in v1.2). You cannot use any options that would affect the post-answer state if this option is used." |
| 14:26.45 | ntz | [TK]D-Fender: this is my PoC dialplan (extensions.ael): http://susepaste.org/view/raw/6185473 |
| 14:26.48 | [TK]D-Fender | What part of that is not clear? |
| 14:27.16 | [TK]D-Fender | When the other side answers, instead of them TALKING they BOTH get thrown somewhere else in the dialplan. |
| 14:27.21 | [TK]D-Fender | Do you have a REASON to do this? |
| 14:27.27 | ntz | okay, no |
| 14:27.36 | ntz | so let's please get to my original issue |
| 14:28.02 | [TK]D-Fender | So what's the actual issue? You are just asking if command XYZ is right ... and not saying what you're actually trying to do |
| 14:28.48 | [TK]D-Fender | ntz> as you can see, a 7:NDID is empty and I am not sure why |
| 14:28.50 | ntz | i'm asking, howto make DID on receiving site working: http://susepaste.org/view/raw/97196809 |
| 14:28.59 | [TK]D-Fender | Your callerID is empty... if your callerID is EMPTY |
| 14:29.14 | [TK]D-Fender | Your call was from one t38modem to another. |
| 14:29.18 | [TK]D-Fender | set the callerID in your PEER |
| 14:29.33 | [TK]D-Fender | ^ |
| 14:30.11 | ntz | o.O |
| 14:31.11 | ntz | [TK]D-Fender: I have it exactly like this chapter 4.2.1 in here https://github.com/hehol/t38modem or this http://www.hylafax.org/archive/2009-03/msg00030.php |
| 14:31.32 | ntz | my faxes are working, the only problem is that I'm unable to know DID on the receiving side |
| 14:31.42 | [TK]D-Fender | <[TK]D-Fender> set the callerID in your PEER <---- |
| 14:31.57 | ntz | I have multiple numbers per side |
| 14:32.01 | ntz | that's my problem |
| 14:32.02 | [TK]D-Fender | <[TK]D-Fender> set the callerID in your PEER <---- |
| 14:32.39 | ntz | [TK]D-Fender: please slow down .... where do I have to set it - in hylafax or asterisk ? |
| 14:32.46 | [TK]D-Fender | ASTERISK SIP PEER |
| 14:32.47 | [TK]D-Fender | From: "root" <sip:T38modem_mgec212@10.0.255.129>;tag=as00007a8d |
| 14:32.56 | [TK]D-Fender | THIS is what your call you your other one looks like |
| 14:32.59 | [TK]D-Fender | that is not a NUMBER |
| 14:33.09 | ntz | hmmm |
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| 14:33.19 | ntz | I'm slowly starting catching it |
| 14:33.19 | [TK]D-Fender | Sec a CALLERID for your peer if you want the CALLING T38modem to show as being FROM a specific number |
| 14:33.42 | ntz | I don't want FROM but TO :D |
| 14:33.59 | ntz | I want DID and not CID |
| 14:34.09 | [TK]D-Fender | eliably Transmitting (no NAT) to 10.0.255.211:5060: |
| 14:34.09 | [TK]D-Fender | INVITE sip:111211@10.0.255.211 SIP/2.0 |
| 14:34.13 | [TK]D-Fender | To: <sip:111211@10.0.255.211> |
| 14:34.26 | [TK]D-Fender | thatt call sent TO that 2nd t3m does have it in the To: |
| 14:34.38 | [TK]D-Fender | so any problem you have with that is t3m-based |
| 14:34.49 | ntz | [TK]D-Fender: thanks, I'm going to reread what you've said and my logs |
| 14:40.50 | ntz | [TK]D-Fender: last question - is my problem in asterisk or not ? |
| 14:40.58 | ntz | i'm totally confused now |
| 14:42.40 | ntz | [TK]D-Fender: http://susepaste.org/view/raw/16975677 << this is *entirely* the configuration on asterisk side |
| 14:43.23 | ntz | ^^ _entire_ cfg on asterisk side - so please, is THAT wrong or the problem is on fax machine (hylafax + t38modem) side ? |
| 14:43.38 | ntz | I have to mention, that faxes are working fine |
| 14:48.12 | [TK]D-Fender | You can see the from & to on the INVITE being sent to the other t38modem |
| 14:48.24 | [TK]D-Fender | the From looks bad, but the "to" has the # |
| 14:48.50 | [TK]D-Fender | So if HF is supposed to do something with 111211 then that's it's problem |
| 14:48.54 | [TK]D-Fender | its* |
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| 14:54.23 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-akdlkelbcjzfyxag) |
| 14:59.23 | ntz | [TK]D-Fender: so in the other words, there's nothing I can do with my Dial() command to explicitely tell it propagate-DID-better-than-you-do-now ? |
| 14:59.43 | [TK]D-Fender | To: = target #, From = Callerid. |
| 14:59.47 | [TK]D-Fender | You have both |
| 15:00.09 | ntz | weird is, that I went to full debugging on HF side and I don't see any mark of DID (To:) completely from HF |
| 15:00.10 | [TK]D-Fender | So if HF is supposed to be doing something with that # in terms of choosing where it goes, then that is HF's problem |
| 15:00.28 | [TK]D-Fender | (also T.38modem's job) |
| 15:00.37 | [TK]D-Fender | basically "not asterisk" |
| 15:00.42 | ntz | ok |
| 15:01.09 | ntz | [TK]D-Fender: what exactly do you mean by mentioning `#' o.O please ? |
| 15:01.21 | [TK]D-Fender | # = number |
| 15:01.24 | ntz | oh |
| 15:01.26 | ntz | ok |
| 15:02.46 | ntz | but according to this: http://susepaste.org/view/raw/22081906, there's `To: <sip:10.0.255.211>' |
| 15:06.27 | ntz | [TK]D-Fender: okay, just answer yes/no (o.O - and thanks again for your valuable input): shall I expect an answer on my Q: in this http://susepaste.org/view/raw/54397922 ?? |
| 15:06.34 | ntz | ^^ from hylafax-config manpage |
| 15:06.55 | [TK]D-Fender | Reliably Transmitting (no NAT) to 10.0.255.211:5060: |
| 15:06.55 | [TK]D-Fender | INVITE sip:111211@10.0.255.211 SIP/2.0 |
| 15:07.00 | [TK]D-Fender | From: "root" <sip:T38modem_mgec212@10.0.255.129>;tag=as00007a8d |
| 15:07.00 | [TK]D-Fender | To: <sip:111211@10.0.255.211> |
| 15:07.03 | [TK]D-Fender | There you have it |
| 15:07.24 | [TK]D-Fender | that's what you're sending to the other t3m |
| 15:07.58 | [TK]D-Fender | So if it's supposed to do something because of 111211 then you did it wrong there |
| 15:08.14 | ntz | hmm |
| 15:09.32 | ModFather | [TK]D-Fender is just the best on asterisk world ;) |
| 15:10.03 | [TK]D-Fender | I just have pretty decent eyes for the obvious :) |
| 15:10.18 | ModFather | :P |
| 15:10.45 | ntz | and my brain doesn't catch it (yet) :D |
| 15:10.48 | [TK]D-Fender | There are users here more experienced than me in any given aspect. |
| 15:10.55 | ntz | but surely, [TK]D-Fender is voip pope |
| 15:11.16 | [TK]D-Fender | I'm just a reflection of the sanity check users should be doing themselves. |
| 15:11.52 | ModFather | and believe me.. you dont want to make [TK]D-Fender get angry :) |
| 15:12.12 | ModFather | :) |
| 15:17.16 | ntz | heck, on the receiving side I see: Jul 28 17:16:28 srv0 FaxGetty[2371]: RECV FAX (000000011): from 111212, page 1 in 0:00:13, A4 x INF, 3.85 line/mm, 2-D MMR, 14400 bit/s |
| 15:17.36 | ntz | but this `from' ain't accessible from inside !!!! uaaaaaaaa :D |
| 15:19.01 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 15:19.01 | ntz | wait - I'm overworked - I need still To: |
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| 16:41.01 | ntz | [TK]D-Fender: JFYI, I was able to resolve that .... partially :D |
| 16:41.36 | ntz | I am able to get correctly the subaddress if the number is specified via $location_prefix#$subaddr |
| 16:43.00 | davlefou | S.O.S, some one have an solution to convert mp3 to asterisk sound file? |
| 16:44.04 | [TK]D-Fender | "man sox" <- |
| 16:44.11 | [TK]D-Fender | tons of WIKI docs for this |
| 16:44.16 | [TK]D-Fender | Also, * can PLAY mp3 files |
| 16:44.21 | [TK]D-Fender | and convert them from CLI |
| 16:44.26 | [TK]D-Fender | Read you rCLI command set |
| 16:44.35 | [TK]D-Fender | file <tab> |
| 16:45.04 | file | tabs |
| 16:46.38 | davlefou | i try sox with no result: http://pastebin.com/K3ndMQ2K |
| 16:46.47 | davlefou | i become crazy! |
| 16:47.15 | [TK]D-Fender | just so you know it started LONG before this incident |
| 16:47.20 | [TK]D-Fender | Your name says it all |
| 16:47.38 | davlefou | more crazy... |
| 16:47.40 | [TK]D-Fender | le fou en fait est trop honnete. |
| 16:48.02 | [TK]D-Fender | file convert <tab> |
| 16:48.11 | file | converts oxygen to code |
| 16:48.16 | davlefou | j'assume ma folie sans complexe. C'est mieux! |
| 16:54.06 | davlefou | Et pour mon probléme, pas de solution? |
| 16:55.33 | [TK]D-Fender | Je t'ai donne DEUX |
| 16:56.01 | [TK]D-Fender | [TK]D-Fender> "man sox" <- |
| 16:56.09 | [TK]D-Fender | <[TK]D-Fender> and convert them from CLI |
| 16:56.15 | [TK]D-Fender | <[TK]D-Fender> file convert <tab> |
| 16:56.22 | davlefou | Avec Sox, je galére totalement! |
| 16:56.28 | [TK]D-Fender | OUVRE T'ES YEUX |
| 16:57.20 | [TK]D-Fender | <davlefou> Avec Sox, je galére totalement! <- il y a des pages PRECISEMENT pour ca aussi.... |
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| 17:21.09 | davlefou | Ai je raté quelque chose http://pastebin.com/NBm5w3FF? |
| 17:22.23 | [TK]D-Fender | Erreur completement NUL |
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| 17:23.03 | [TK]D-Fender | t'ajout JAMAIS l'extension du fichier avec Playback, Background, etc |
| 17:29.55 | davlefou | Le truc con que j'avais oublié... Merci!!!! Je vais m'acheter une corde et me pendre! |
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| 17:31.28 | davlefou | [TK]D-Fender, merci pour ton aide. |
| 17:55.03 | Get_The_Fish | Hi all, I'm working with SIP messaging in Asterisk, and I was wondering if there was any way to catch events where the sip packet failed to send |
| 17:56.03 | Get_The_Fish | In the dialplan, that is |
| 17:58.02 | rmudgett | No. There are too many layers between the dialplan and actually sending for that to bubble back up. |
| 17:59.12 | Get_The_Fish | any other way to do it off the top of your head? AMI, ARI? |
| 18:01.38 | rmudgett | no |
| 18:01.46 | rmudgett | It is best effort |
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| 18:36.15 | Get_The_Fish | rmudgett: so actually, Asterisk does set a variable on the failure of a SIP message, which is MESSAGE_SEND_STATUS |
| 18:36.51 | Get_The_Fish | found that by setting the debug up to 3 and sending a message to a SIP address was defined but not registered. |
| 18:37.00 | Get_The_Fish | just FYI |
| 18:37.15 | rmudgett | That status won't always tell you if the message actually got sent. |
| 18:38.00 | rmudgett | just FYI |
| 18:38.13 | Get_The_Fish | right, sure, but I will tell me basic failure conditions, such as device isnt registered. |
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| 18:46.12 | jr220 | Hello. AMI question. From the documentation it seems that âNewstateâ event should tell me when a channel goes âDownâ, but Iâm not catching any events with the ChannelStateDesc of âDownâ⦠The only Down I ever get is on Newchannel when the channel is first created.. Any ideas why the Down event wouldnât be there or how I can capture it? |
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| 18:47.00 | file | define "when a channel goes down" and what technology? |
| 18:49.11 | jr220 | Itâs all SIP technology.. The back story is I am trying to write an integration with some software and asterisk and I need to capture incoming and outgoing calls. So, I am capturing when the channels come up to ring the extensions, but I need to know when that channel is no longer active so I can remove it from my list of channels Iâm dealing with.. |
| 18:49.48 | file | you'd get a different event to indicate that the channel has been hung up |
| 18:49.57 | jr220 | I have been just using the Newstate to capture the events, and then Extensionstatus looking for idle to determine when the phone has hung up, but the issue with all this assume one line per phone which isnât the case.. |
| 18:50.08 | jr220 | ok.. Hangup then? |
| 18:50.15 | jr220 | I think I remember a hangup event? |
| 18:50.24 | file | yes |
| 18:51.06 | jr220 | ok, thanks. let me look at that.. I guess perhaps Down states on channels might not apply to SIP technologies? |
| 18:51.22 | file | they don't go to a down state when hung up |
| 18:51.36 | file | they just hang up and disappear |
| 18:52.16 | jr220 | gotcha.. This was so simple in my mind in the start.. :-/ |
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