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| 03:54.26 | bjhaid | hi |
| 03:56.10 | *** join/#asterisk cihhan (~cihan@ip72-222-162-91.ph.ph.cox.net) |
| 03:57.18 | cihhan | hi all, im trying to set an asterisk based pbx in our small office -- basic installation is done (i can call outside and can receive calls). but I want to add one more feature: When there is an incoming call, I want all the phones to ring. Is that possible? |
| 03:57.53 | drmessano | ~book |
| 03:57.54 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 03:58.16 | drmessano | Yes, entirely possible.. and several different ways to do it, depending on need |
| 03:58.21 | drmessano | Along with MANY other things |
| 04:04.46 | cihhan | drmessano: ooh thanks :) can you suggest some methods or references? |
| 04:04.56 | bjhaid | so I am running into some problems with outbound calls |
| 04:04.58 | cihhan | As you can guess, Im learning Asterisk now |
| 04:05.03 | bjhaid | via a sip provider |
| 04:05.04 | drmessano | I just gave you one |
| 04:05.10 | drmessano | The book |
| 04:05.15 | bjhaid | if I connect zoiper to the provider it works fine |
| 04:05.43 | cihhan | oooh ok, thanks a lot drmessano |
| 04:05.44 | bjhaid | but if I connect zoiper -> asterisk -> sip provider I don't get audio on the receivers end |
| 04:06.07 | bjhaid | that's the phone reached via my network |
| 04:06.19 | bjhaid | I have put asterisk in my DMZ still doesn't work |
| 04:06.29 | drmessano | ~sipnar |
| 04:06.31 | drmessano | ~sipnat |
| 04:06.31 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 04:06.36 | drmessano | ^^^^ bjhaid |
| 04:07.10 | drmessano | DMZ is not the answer.. you need ports forwarded (SIP and RTP) as well as the above params |
| 04:10.25 | bjhaid | drmeassano: I have port forwarding configured |
| 04:10.34 | bjhaid | no juice, so I decided to try DMZ |
| 04:12.47 | bjhaid | I have not configured directmedia and externhost |
| 04:12.54 | bjhaid | would do that and come back if I have problems |
| 04:13.37 | drmessano | and localnet |
| 04:13.40 | drmessano | All those work together |
| 04:14.06 | drmessano | nat, directmedia, externhost/externip, and localnet |
| 04:14.56 | bjhaid | I have localnet configured |
| 04:15.02 | bjhaid | I am confused about directmedia |
| 04:15.07 | bjhaid | should it be yes or nonat |
| 04:15.19 | bjhaid | the documentation in the sip.conf isn't clear |
| 04:16.34 | drmessano | nonat |
| 04:16.39 | bjhaid | thanks |
| 04:17.33 | drmessano | What do you have for localnet? |
| 04:20.00 | bjhaid | localnet=192.168.29.0/255.255.0.0 |
| 04:20.16 | bjhaid | I just reloaded asterisk, and still can't get it to work |
| 04:20.24 | drmessano | You have a /23 ? |
| 04:20.35 | drmessano | Interesting |
| 04:20.37 | drmessano | Ok |
| 04:21.01 | drmessano | Did you set nat=yes on the sip.conf config for the zoiper endpoint? |
| 04:21.44 | bjhaid | yes |
| 04:21.48 | drmessano | Your firewall matches your RTP ports? |
| 04:21.57 | bjhaid | I want to post my sip.conf |
| 04:22.04 | drmessano | pastebin |
| 04:24.37 | bjhaid | you mind gist.github ? |
| 04:25.02 | bjhaid | https://gist.github.com/bjhaid/c5bf37fdc24de8c28cd4 |
| 04:25.52 | drmessano | you dont specify a port for extenip |
| 04:25.55 | drmessano | you dont specify a port for externip |
| 04:26.06 | drmessano | and that allow=gsm; ? Not sure about the ; |
| 04:26.15 | bjhaid | okay |
| 04:26.27 | drmessano | Fix that, if that doesnt work.. sip debug |
| 04:26.55 | bjhaid | is it externip or externaddr |
| 04:26.55 | bjhaid | ? |
| 04:26.56 | drmessano | Hang on |
| 04:27.09 | drmessano | Those need to be in the [general] section of sip.conf |
| 04:27.20 | bjhaid | okay |
| 04:27.39 | drmessano | externip = address ... externhost = hostname (asterisk looks up the address in DNS.. Like for dyndns) |
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| 04:33.43 | bjhaid | drmeassano: no juice still |
| 04:40.02 | bjhaid | with debug enabled in the console I see: |
| 04:40.04 | bjhaid | <--- Reliably Transmitting (NAT) to 192.168.29.213:50232 ---> SIP/2.0 401 Unauthorized |
| 04:41.51 | ChannelZ | that's not necessarily wrong |
| 04:42.02 | ChannelZ | it's what happens after |
| 04:52.59 | bjhaid | so from tcpdump I can see that during the period of the call no packets come back from zoiper to asterisk |
| 04:55.08 | bjhaid | which might be the cause of the problem |
| 04:55.19 | bjhaid | I am not sure if ^ is true though |
| 04:55.36 | bjhaid | neither do I know how to fix it if it is the cause |
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| 05:10.35 | [TK]D-Fender | that config is broken |
| 05:10.40 | [TK]D-Fender | localnet=192.168.29.0/255.255.0.0 ; RFC 1918 addresses |
| 05:10.44 | [TK]D-Fender | has no place in a peer entry |
| 05:10.49 | [TK]D-Fender | externaddr=x.x.x.x:5060 |
| 05:10.51 | [TK]D-Fender | Nor this |
| 05:11.45 | bjhaid | I put that in the [general] |
| 05:11.47 | bjhaid | Fender |
| 05:13.09 | [TK]D-Fender | https://gist.github.com/bjhaid/c5bf37fdc24de8c28cd4 |
| 05:13.13 | [TK]D-Fender | not from wha I see in here |
| 05:13.33 | bjhaid | I have changed that config |
| 05:13.35 | [TK]D-Fender | Show new configs and the actual debug |
| 05:13.43 | bjhaid | from drmeassano's advice |
| 05:13.53 | [TK]D-Fender | I shouldn't take it on faith if we're to be properly debuggin.... |
| 05:14.15 | bjhaid | okay |
| 05:14.17 | [TK]D-Fender | Show the new configs and new debug |
| 05:14.21 | bjhaid | I would show you my modified config |
| 05:17.15 | bjhaid | Fender: see update |
| 05:17.15 | bjhaid | https://gist.github.com/bjhaid/c5bf37fdc24de8c28cd4 |
| 05:18.08 | [TK]D-Fender | Still broken |
| 05:18.26 | [TK]D-Fender | register has to come AFTER everything else under [general] and BEFORE the first other entry |
| 05:18.49 | [TK]D-Fender | And you didn't specify "nat=yes" for [general] |
| 05:19.35 | [TK]D-Fender | Also why did you jsut change "externaddr" to "externip"? |
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| 05:24.00 | bjhaid | updated config |
| 05:24.16 | bjhaid | Fender, still not getting audio on the outside phone |
| 05:24.36 | [TK]D-Fender | And I don't see an updated link |
| 05:24.56 | [TK]D-Fender | Nor do I see th CALL. I ahve no idea if what is actually coming in does not actually look like garbage. |
| 05:25.22 | [TK]D-Fender | Show what you now consider as "fixed"... and show the actual call. |
| 05:31.59 | bjhaid | Fender: https://gist.github.com/bjhaid/c5bf37fdc24de8c28cd4 |
| 05:32.16 | bjhaid | should I post a full dump of the debug of the call |
| 05:32.17 | bjhaid | ? |
| 05:33.02 | [TK]D-Fender | You STILL don't have "nat=yes" under [general] |
| 05:33.42 | [TK]D-Fender | <bjhaid> should I post a full dump of the debug of the call <- I asked you for this THREE times already now. |
| 05:36.38 | [TK]D-Fender | You should also be PREVENTING reinvites. All of your peers should simply have "directmedia=no" |
| 05:37.42 | [TK]D-Fender | And you if this is remote device then you shouldn't trust that is honest with the IP's it is offering and therefor assume they are behind NAT and cannot be trusted for media IP's |
| 05:37.51 | [TK]D-Fender | Those should ALSO be "nat=yes" |
| 05:41.23 | bjhaid | thanks it works now |
| 05:41.41 | bjhaid | setting directmedia=no seems to do the trick |
| 05:41.49 | bjhaid | I had it has nonat previously |
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| 05:43.03 | bjhaid | thanks Fender |
| 05:43.08 | [TK]D-Fender | And meanwhile you weren't defining their as being NAT'd and thus untrustworthy. |
| 05:43.17 | [TK]D-Fender | And on that note... bed time.... |
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| 06:16.19 | MertsA | Does anybody know how to debug issues with T.38 and UDPTL? |
| 06:18.08 | MertsA | I'm getting some weird issues with Asterisk sending an INVITE with image 4303 udptl t38 but when UDPTL traffic is sent to 4303 Asterisk doesn't send it back out to the ATA that started the INVITE |
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| 07:27.48 | tiuman | how disable dial and continioe dialplan, from php |
| 07:28.34 | tiuman | how disable exten dial, and continioe dialplan from php |
| 07:32.23 | tiuman | WIMPy |
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| 08:34.04 | phix | WIMPy |
| 08:34.26 | phix | tiuman: You mean from python right? |
| 08:37.10 | phix | Why you would want to use PHP for? It is slow (unless you use HHVM), buggy (including HHVM), 99% of third party apps are written by kiddies or people with no programming fundamentals and the syntax / core library is poorly written and contains mismatched idioms. |
| 08:37.54 | phix | If you want a a decent scripting / psudeo-programming language a recommend python or perl |
| 08:38.11 | phix | (but mostly python :)) |
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| 08:53.51 | nafg_ | Hello. I'm doing some asterisk stuff after a while of not, plus I lost my old configuration files, and I'm getting stuck |
| 08:54.21 | nafg_ | Right now I'm just trying to originate a call from the CLI via callwithus |
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| 08:54.51 | nafg_ | (FTR I'm doing this inside docker but I don't think that's the issue here) |
| 08:56.29 | nafg_ | https://gist.github.com/nafg/56342a25a9c24559b665 |
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| 08:58.06 | nafg_ | Sholom aleichem ;) |
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| 09:05.27 | tiuman | i do it from macros. |
| 09:05.54 | tiuman | but it work bad method |
| 09:06.22 | tiuman | disconnect him doin only in moment answer |
| 09:06.38 | tiuman | disconnect him doing only in moment answer |
| 09:06.39 | nafg_ | What? |
| 09:06.50 | tiuman | how disable exten dial, and continioe dialplan from php |
| 09:06.52 | nafg_ | Ok I realized need to write SIP/callwithus |
| 09:07.03 | tiuman | i realized with macros |
| 09:07.22 | tiuman | Set(MACRO_RESULT=CONTINUE) |
| 09:07.45 | tiuman | but it disconnect only moment answer |
| 09:07.57 | tiuman | i need any moment! |
| 09:09.21 | tiuman | WIMPy!!!!!! |
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| 09:10.00 | tiuman | http://pastebin.com/UBU4iACB |
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| 09:23.26 | babak | Hi, do you know who in confirming joining Asterisk users mailing list ? I am 3 days trying not succeeful to join |
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| 09:28.37 | ipalmer | hi all, hopefully a simple question. When using originate to make an outbound call using the ami, the source endpoint is rung first, then when answered it calls the destination. Is there a way to get the source to auto answer so an outbound call is a one step process? |
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| 09:41.06 | tiuman | Action: Hangup \n Channel: /^SIP/101-.*$/ \n Message: No such channel |
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| 09:41.31 | tiuman | asterisk not uderstart regex? |
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| 12:17.20 | [TK]D-Fender | ipal |
| 12:17.35 | [TK]D-Fender | ipalmer: depends what you are calling. |
| 12:18.00 | nunne | Having enormous problems getting asterisk 11 working with ice (for webrtc) in ubuntu 14.04. I have the libuuid +dev-packades. But it just will never send ice-ufrag and ice-pwd. Anyone have this setup working? tried bother 11.18 and 11.10. |
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| 12:25.37 | file | [TK]D-Fender, what's the weather like up your way? |
| 12:26.47 | [TK]D-Fender | pretty hot and clear skies since mid Saturday. 1 day of tiny rain predicted then clear again for a week & half |
| 12:27.03 | [TK]D-Fender | #accuweatherproxy |
| 12:37.01 | ipalmer | D-fender: Cheers, I'm calling a Zoiper soft phone, using pjsip |
| 12:38.36 | [TK]D-Fender | Go check it's manual to see if it supports it. I'm doubtful... |
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| 12:39.09 | ipalmer | will do was just looking at it anyway, using the sip add header |
| 12:40.34 | [TK]D-Fender | That's the normal means for average "hard" SIP phones |
| 12:40.55 | ipalmer | ah ok, I'll have a look in to it, thanks |
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| 13:06.32 | nunne | ipalmer: If i remember correctly Zoiper supports "SIPAddHeader(Call-Info: answer-after=0)" |
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| 14:14.28 | BuddyButterfly | hi |
| 14:14.46 | BuddyButterfly | is it possible to record conferences is confbridge in gsm format? |
| 14:15.10 | BuddyButterfly | wav is too big. |
| 14:15.37 | phix | It is possible as asterisk is opensource |
| 14:15.58 | phix | The feature may not exist yet but you can code it in if you like |
| 14:15.58 | BuddyButterfly | I mean through customizing. |
| 14:16.17 | BuddyButterfly | or spezifying a file name with .gsm extension? |
| 14:16.28 | phix | *shrugs* |
| 14:16.38 | [TK]D-Fender | You always specify the format for recording. |
| 14:16.41 | BuddyButterfly | phix: so your answer is: no? |
| 14:18.01 | BuddyButterfly | I just wanted to ask for the currently available functionality. |
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| 14:18.44 | BuddyButterfly | if it is not available, no problem. just wanted to know. |
| 14:19.03 | [TK]D-Fender | it is |
| 14:19.09 | [TK]D-Fender | You always choose the format |
| 14:19.15 | BuddyButterfly | oh |
| 14:19.18 | BuddyButterfly | where? |
| 14:19.58 | phix | BuddyButterfly: the answer is I do not know |
| 14:20.00 | BuddyButterfly | documentation just says wav |
| 14:20.14 | phix | BuddyButterfly: but it is possible |
| 14:20.24 | phix | BuddyButterfly: so I was answering your questions correctlky |
| 14:20.42 | phix | [TK]D-Fender: <3 |
| 14:20.59 | [TK]D-Fender | Should be able to do this by specifying the name |
| 14:21.37 | BuddyButterfly | [TK]D-Fender: you mean giving an explicit file name with .gsm extension. OK, will try that. |
| 14:21.47 | phix | [TK]D-Fender: What's the latest? I didn't get my TDM working so I used a linksys 3200 |
| 14:21.52 | phix | or what ever the model number is |
| 14:22.01 | [TK]D-Fender | phix: Latest what? |
| 14:22.13 | phix | [TK]D-Fender: The latest news |
| 14:22.40 | BuddyButterfly | [TK]D-Fender: tnx. |
| 14:22.50 | [TK]D-Fender | phix: SSDD |
| 14:23.17 | phix | [TK]D-Fender: Sounds impressive |
| 14:23.42 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-eatxjdsmyanjojwr) |
| 14:23.49 | [TK]D-Fender | phix: I'm suspecting you don't recognize the term... |
| 14:24.43 | nunne | I actually think confbridge recording only is wav due to the internal sampling of confbridge. so i think you'd how to find other means of converting the files to another format. |
| 14:25.05 | phix | [TK]D-Fender: What gave that away ;) |
| 14:25.19 | [TK]D-Fender | phix: considering it "impressive" |
| 14:25.41 | phix | :) |
| 14:25.41 | nunne | is it that the file gets to big or that you just want to make them smaller for the person thats going to download it or whatever? otherwise you could just run a script that converts the file to mp3 or gsm or whatever. |
| 14:25.50 | phix | ~SSDD |
| 14:25.52 | infobot | somebody said ssdd was Same Shit Different Day |
| 14:26.04 | phix | thnx infobot |
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| 15:10.07 | Inf0r | Asterisk1.8/voicemail.conf -- Can you configure the pager to ALSO attach the voicemail as a wav,etc? |
| 15:11.09 | mjordan | what would a pager do with an attachment? |
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| 15:19.14 | Inf0r | pager is the "secondary email address" it works, just doesn't attach a file. didn't know there is a way to do it or i just have to use mail alias |
| 15:20.28 | [TK]D-Fender | Inf0r: No |
| 15:20.37 | [TK]D-Fender | Inf0r: You'll need an alias |
| 15:20.44 | Inf0r | k thanks |
| 15:20.53 | [TK]D-Fender | Pager is explicitly without attachement |
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| 15:54.24 | Plixic | For phantom calls, is it better to just change the port or try and mess with iptables? Using a VPS |
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| 16:28.11 | catphish | in older versions of asterisk, i have an extension *8 in my dialplan, but in asterisk 13, *8 seems to be intercepted and tries to do a pickup instead of executing what's in my diaplan |
| 16:28.13 | catphish | is this configurable somewhere so that I can make the *8 in my dialplan work? |
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| 16:43.52 | rmudgett | catphish: See features.conf.sample for ;pickupexten = *8 ; Configure the pickup extension. (default is *8) |
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| 16:44.06 | catphish | rmudgett: thank you! |
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| 16:51.01 | yun1989 | Hello all |
| 16:51.30 | yun1989 | I have one doubt it's possible when exist one user in conference show one music ? |
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| 20:11.21 | spicyramen_ | hi people, anybody have use sipp using a sip uri in this format: sip:22@spicyrame.sip.com, the only format I can see is sip:22@1.1.1.1 |
| 20:11.39 | spicyramen_ | I emailed the forum, but mabye in this forum can get some info |
| 20:15.13 | Synthase_ | If the hostname resolves correctly, that can work. |
| 20:33.05 | spicyramen_ | @Synthase, Im trying this ./sipp -s username -ap password -r 1 x.sip.twilio.com -sf scenarios/uac_sip_authentication.xml -m 1 -d 10000, and in the SIP INVITE, looks like sipp is doing the nslookup and replacing x.sip.twilio.com to the ip address |
| 20:38.30 | spicyramen_ | is using INVITE sip:username@107.21.222.153:5060 SIP/2.0 |
| 20:38.55 | spicyramen_ | where 107.xxxx is the A host for my spicyramen.sip.twilio.com |
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| 21:10.45 | *** join/#asterisk Apocryphal (~js@0x3ec66261.inet.dsl.telianet.dk) |
| 21:14.07 | Apocryphal | I'm trying to diagnose an audio issue I'm experiencing. When I try to use the DIAL app to reach another peer, audio travels fine between caller and asterisk. But callee (the one dial to via the DIAL app). doesn't receive any audio. Though from wireshark it looks like RTP packages are running fine in both directions (to and from callee <-> asterisk). However, if I dial callee from a local channel, and do a call announcement, that audio goes |
| 21:14.27 | Apocryphal | Really weird. I'm hoping someone else has experienced something like this before because I don't even know where to look anymore |
| 21:14.40 | Apocryphal | SIP packages flows just fine as well. |
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| 21:49.14 | newtonr | Apocryphal, there was a bug recently with goofy audio through local channels. What version are you using? |
| 21:49.35 | newtonr | Apocryphal, try the git head of whatever branch you are using to see if the issue persists |
| 21:49.47 | newtonr | or the very latest release version as it should be fixed in that |
| 21:50.05 | newtonr | assuming it is the same issue |
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| 22:10.08 | PHunter | Anyone know a analog gateway or card that works with Japanese Telco? (NTT CID I think..) |
| 22:29.22 | Apocryphal | newtonr: I'm on 13.2 - it works on local channels though. IT's when just doing a straight dial. Answer() Dial(SIP/someone); Hangup(); that the audio doesn't work on SIP/someone. Really strange |
| 22:29.35 | Apocryphal | I'm upgrading an instsance to 13.5 now to see if that helps with anything |
| 22:30.17 | Apocryphal | What really bugs me is that as far as I can see in wireshark, SIP/someone gets all the RTP packages just fine, and is also sending them just fine (as evident by the caller getting all audio |
| 22:30.29 | Apocryphal | If I record stuff in Asterisk, asterisk also gets audio from both phones. |
| 22:32.00 | newtonr | Oh.. I thought you were saying the opposite |
| 22:32.05 | newtonr | I'm tired! |
| 22:32.33 | Apocryphal | No problem ;) |
| 22:33.24 | Apocryphal | I've got a Kamailio that sits as the registrar - but I doubt that has any impact. IT doesn't meddle with the SDP packages. And the fact thatit works if the call is done through a local channel |
| 22:35.53 | Apocryphal | Ah, let me correct. It works if I dial through a local channel, that local channel plays some audio to the callee, before returning and bridging to the caller |
| 22:36.45 | Apocryphal | F.. my english is defective tonight. ** dial through a local channel, and that local channel plays some audio....... |
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