| 00:02.21 | tompaw | Where are channel's queued frames processed? |
| 00:08.35 | *** join/#asterisk KaliLinuxGR (~KaliLinux@unaffiliated/kalilinuxgr) |
| 00:08.51 | scriptinghelp | http://pastebin.com/9jJDxmh6 |
| 00:09.03 | scriptinghelp | thatâs how the new output / modified config look |
| 00:09.05 | scriptinghelp | looks |
| 00:10.02 | [TK]D-Fender | Certainly looks a lot better. |
| 00:11.25 | [TK]D-Fender | You can also use the "QueuePauseMember" and "QueueUnPauseMember" dialplan apps to let members directly pause their member state so they can opt-out from getting calls if they are there but can't be disturbed, etc |
| 00:13.18 | *** join/#asterisk monsterco (~monsterco@toroon474aw-lp130-01-70-49-228-179.dsl.bell.ca) |
| 00:14.04 | monsterco | on a Sipura or SPA unit where can I set the proxy port? My provider uses port 5160 instead of 5060 and I don't see a place where to set it |
| 00:17.33 | scriptinghelp | thanks alot [TK]D-Fender |
| 00:17.43 | [TK]D-Fender | You're welcome |
| 00:17.50 | scriptinghelp | sorry to keep on with the questions - but any idea how i can see what music on hold file is currently playing? |
| 00:18.33 | [TK]D-Fender | "Not AFAIK |
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| 00:19.23 | tompaw | Whoever called the local channel module "unreal" knew what he was doing. |
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| 00:32.42 | Samot | monsterco: The same way you do it normally -> HOSTIP:5160 |
| 00:33.02 | monsterco | yep thats what i tried and works; tnx |
| 00:33.11 | Samot | monsterco: If the :PORT isn't specified in an URI, it defaults to 5060 |
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| 01:17.59 | scriptinghelp | thanks everyone |
| 01:18.05 | scriptinghelp | especially [TK]D-Fender i am grateful for your help |
| 01:18.15 | scriptinghelp | have a lovely day gents |
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| 02:34.52 | scriptinghelp | hi again everyone |
| 02:35.15 | scriptinghelp | how can i show the music on hold filename.wav playing currently in the asterisk cli? |
| 02:35.18 | scriptinghelp | if its sort=random |
| 02:39.52 | scriptinghelp | are there any paid consultants online? i really could do with one in my contact list :P |
| 02:41.29 | [TK]D-Fender | * doesn't show what it's playing |
| 02:41.41 | [TK]D-Fender | there simply isn't a status dump I'm aware of to do so |
| 02:42.03 | [TK]D-Fender | Beyond showing it what practical action or information is that supposed to lead to? |
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| 09:58.36 | samwierema | Where can I find what type of audio files Asterisk can play (using the Playback application)? |
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| 11:15.14 | Samot | http://wiki.asterisk.org |
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| 13:30.44 | tsearle | hello, all I'm looking into chan_sip to chan_pjsip migration and there are some config flags I can't seem to align, may I ask ref them here? |
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| 13:37.47 | tsearle | if a chan_pjsip endpoint is only used for outbound, can you have several endpoints use the same AOR? |
| 13:40.19 | file | yes. |
| 13:41.38 | tsearle | cool |
| 13:42.37 | tsearle | also, trying to find an equivalent of insecure=port, I assume it should be some flag on the aor lvl |
| 13:43.57 | file | it's a completely separate config section |
| 13:44.00 | file | it's the type=identify section |
| 13:44.15 | file | it matches based on an IP address, multiple IP addresses, or subnet ranges and associates it to an endpoint |
| 13:44.48 | tsearle | ok, so I used a type=identify instead of type=aor ? |
| 13:45.33 | file | a type=aor specifices how to contact something |
| 13:45.45 | file | a type=identify specifies how to match an incoming request to an endpoint |
| 13:45.49 | file | if you want to do both, you need both |
| 13:45.55 | tsearle | ha ok |
| 13:46.18 | tsearle | so use identify for the ingress endpoint and aor for the egress |
| 13:46.30 | file | yes |
| 13:48.22 | tsearle | ok, and one last one for now, is there a way to set the useragent header? |
| 13:49.32 | file | yes, there is a user_agent option in the global section |
| 13:56.09 | ppd1990 | Hello, I have a slight problem with my Asterisk 14 setup. I have two VLANs (wifi and phones) and the server listens on both. The clients in the wifi network are Android phones with the CsipSimple client. If I call from the mobile to the "hardphones", everything works perfectly. However, the other way round results in no Audio in both directions. No direct media is set. I have pjsip logs for both directions: yealink hardphone to mobile: |
| 13:56.09 | ppd1990 | http://pastebin.com/A9pmytEL does NOT work and csipsimple to yealink: http://pastebin.com/m7qCjEnq) which does work. Is there anything glaringly obvious I'm doing wrong? |
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| 14:08.50 | Samot | ppd1990: What's the IP of the PBX? |
| 14:09.50 | ppd1990 | Samot: 10.0.3.26 and 10.0.4.26 |
| 14:11.36 | Samot | Which one is the primary VLAN? |
| 14:12.31 | ppd1990 | the primary is actually 10.0.1.26. 10.0.3.26 is the phone VLAN and 10.0.4.26 is the trusted wifi VLAN |
| 14:12.49 | Samot | Using Chan_SIP? |
| 14:13.10 | ppd1990 | Chan_PJSIP |
| 14:13.32 | Samot | Listening on 5061? |
| 14:14.15 | ppd1990 | bind=0.0.0.0:5061 :) |
| 14:15.10 | Samot | Then why does the one that works not have :5061 in it's R-URIs in the INVITEs, etc..? |
| 14:15.20 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 14:17.54 | Samot | Why is one device sending calls to 5060 and the other sending them to 5061? |
| 14:18.51 | ppd1990 | Samot: I see. I didn't have the outbound proxy set on the Yealink phone. I just set the the SIP port to 5061 and assumed that'd be it |
| 14:19.33 | Samot | Does setting the OB proxy fix the isse? |
| 14:19.36 | Samot | Does setting the OB proxy fix the issue? |
| 14:23.18 | ppd1990 | it is the same. THe mobile rings, the call gets established and then utter silence. Calling from the mobile works |
| 14:25.19 | Samot | Well this is 100% networking? |
| 14:25.22 | Samot | Well this is 100% networking. |
| 14:29.24 | ppd1990 | I suppose so. Thanks for your help! At least I know it's nothing Asterisk related |
| 14:29.25 | monsterco | what is the command to load a new codec into asterisk without restart? |
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| 14:36.00 | Samot | After you've recompiled Asterisk with it? |
| 14:37.39 | Samot | If you've recompiled Asterisk for any reason, you need to restart it. |
| 14:37.52 | Samot | It won't know of the changes until it is. |
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| 16:17.40 | tompaw | Morning |
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| 17:02.19 | hexanol | small question about mp3 support in res_musiconhold: https://wiki.asterisk.org/wiki/display/AST/MP3+Support says it's deprecated -- I'm guessing the information is exact ? I did not found anywhere else the mention that it was deprecated. And if it's deprecated, does it mean there is some plan to eventually remove it from res_musiconhold ? |
| 17:03.32 | [TK]D-Fender | probably no plan, and that is mpg123, not general file-based MP3 support |
| 17:04.32 | hexanol | right, I forgot for a moment that the "files" mode can play mp3 if you have the format_mp3 |
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| 17:11.45 | igcewieling1 | why not convert it to ulaw and save all the transcoding? |
| 17:44.32 | hexanol | igcewieling1: I totally agree on a technical standpoint, and that's what I personally do, but this is for some PBX application that has MOH "mp3" support (for historical reason) and was wondering about removing it |
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| 17:50.17 | igcewieling1 | hexanol: even if it was removed, you could still use the custom MoH class and something like mpg123 |
| 18:03.53 | mub | robodialer.net allows you to make a free test account that gets unlimited channels and CPS |
| 18:04.02 | mub | I'm surprised they can do that |
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| 18:58.36 | rolledgold | Hello all. New 11.25.1 install, IP address is bound, SIP.conf filled, SIP show peers has them listed. Reg a softphone and TCP dump confirms ivoted gets to asterisk. Asterisk does nothing, no 404, no 401, dead. No responce. Any dieas? |
| 18:59.42 | eric_hill | core set verbose 5 and sip set debug on would be a good start. |
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| 19:02.15 | rolledgold | sip debug is on and verbose was 3 |
| 19:03.11 | eric_hill | ~pb |
| 19:03.11 | infobot | i heard 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:03.23 | eric_hill | Now can you pb the sip trace to see the inbound. |
| 19:03.43 | eric_hill | Also show the output of netstat -anp | grep 5060 |
| 19:03.48 | rolledgold | console => notice,warning,error,debug |
| 19:06.54 | rolledgold | http://pastebin.ca/3757743 |
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| 19:11.43 | rolledgold | <PROTECTED> |
| 19:11.50 | rolledgold | it's up and listening |
| 19:12.37 | eric_hill | So asterisk is listening on 5060, but shows nothing in the console when sip debugging is on? |
| 19:13.21 | rolledgold | yes, very strange. |
| 19:13.25 | *** part/#asterisk hexanol (~hexanol@45.72.150.78) |
| 19:13.57 | eric_hill | Your pb shows a raw socket dump, but doesn't have headers decoded. Is there a firewall redirecting port 5060 traffic? |
| 19:15.04 | rolledgold | it'svery a VPN |
| 19:15.08 | rolledgold | via a |
| 19:15.52 | eric_hill | Are you sure your VPN is dropping the traffic into the local LAN and not localhost? |
| 19:16.21 | rolledgold | yes, this is just a new server in the farm. |
| 19:16.42 | rolledgold | the tcpdump was on the same server |
| 19:17.32 | eric_hill | You have something eating packets before they hit asterisk. |
| 19:17.47 | eric_hill | They're being sent elsewhere or blocked. |
| 19:18.54 | *** join/#asterisk MrMojito (~MrMojito@unaffiliated/mathisen/bot/mrmojito2) |
| 19:19.17 | rolledgold | how could that be? the dump is from the asterisk server. |
| 19:23.30 | MrMojito | Is there anyone who could assist me to get an SNMP deamon installed on an Asterisk? I want to monitor our Asterisk for CPU/Memory load but we have no one in house who can set this up. If there is any documentation I´m willing to read that as well. |
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| 19:30.27 | eric_hill | Without seeing any other details about your installation, it could be any number of things. I'd start with the VPN configuration. Routing specifically. |
| 19:32.22 | rolledgold | but I know the packets get to the server, I see them arive. |
| 19:32.31 | rolledgold | but asterisk takes no actions |
| 19:34.27 | eric_hill | rolledgold: Start eliminating variables. Try it without the VPN and see what happens. |
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| 21:14.04 | OnTheLake | Anyone every come hear of Asterisk crashing on a regular schedule? Every 8 hours exactly. 11.3 |
| 21:16.15 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 21:20.17 | igcewieling1 | OnTheLake: anyone who did would have updated to the latest Asterisk 11.x (as long as you stay int he Asterisk 11 branch updates are fairly easy, |
| 21:21.47 | OnTheLake | We've been running it for 4 years without issues, the odd crash after a year or OS patches. We don't have the problem on other boxes running 11.25 compiled. The 11.3 is binary |
| 21:22.14 | OnTheLake | approx 270 extensions |
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| 22:02.36 | drmessano | OnTheLake: Is the PBX on the public internet? |
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| 23:30.42 | tompaw | Is there a way to disable CONNECTEDLINE propagation system-wide? ,,I doesn't work for stasis bridges |