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| 01:24.16 | Michael_Sia | Hi, I am newbie here. Have some questions to ask regarding my one of my servers, SG PBX trunk to HK server, it suddenly becomes unreachable. Anyone here can help me out with it? Thank you |
| 01:25.45 | TazzNZ | Michael_Sia, did you confirm both servers are up ? |
| 01:26.00 | TazzNZ | what has changed since it was working ? Firewall/IP's/etc ? |
| 01:26.40 | Michael_Sia | yes both server are up. no changes has been done. it just suddenly stop working. We have few servers in different country, so only one server, SG server not able to connect to it. |
| 01:27.51 | Michael_Sia | usually i restart the server, it will work again, but now I have tried many times restarting both servers and it wont help. |
| 01:28.07 | TazzNZ | are you sure asterisk is running on the server ? |
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| 01:28.35 | Michael_Sia | yes, because my server is connecting to 4 trunks, all other trunks working except this one |
| 01:29.10 | Michael_Sia | to be specific, this trunk is working one way, which means SG server not able to connect to HK server, but HK server can connect to SG server with this trunk |
| 01:29.23 | Michael_Sia | have check through the peers setup and nothing has been changed. |
| 01:29.27 | TazzNZ | so the only one that is broke, is SG to HK. the other servers can connect to HK ? |
| 01:29.35 | Michael_Sia | yes correct |
| 01:29.36 | TazzNZ | s/broke/broken |
| 01:29.50 | TazzNZ | and the SG server can ping the HK server ? |
| 01:30.26 | TazzNZ | yeah - the problem is, you need to check *everything* for changes, from the firewall to routing to networks |
| 01:30.29 | TazzNZ | even disks |
| 01:30.40 | Michael_Sia | not able to ping the HK server |
| 01:30.49 | Michael_Sia | i am not sure which is blocking the ping |
| 01:31.03 | Michael_Sia | but i am not able to ping it, even from the other servers that is working |
| 01:31.17 | TazzNZ | ok - so that rules out ping as a test |
| 01:31.27 | TazzNZ | can you SSH to the HK server from the SG server ? |
| 01:31.27 | Michael_Sia | in fact i have a few servers not able to ping, but they all in working status |
| 01:31.35 | Michael_Sia | yes i can ssh |
| 01:31.42 | Michael_Sia | i can web remote |
| 01:31.46 | TazzNZ | I would check firewalls |
| 01:31.57 | TazzNZ | I suspect something is blocking your SIP/IAX traffic |
| 01:33.10 | Michael_Sia | i see. any suggestion to what to check? I have check through my firewall, my firewall only block incoming, but allowed all for outgoing |
| 01:33.38 | TazzNZ | and the HG firewall ? |
| 01:33.43 | TazzNZ | HK* |
| 01:34.26 | Michael_Sia | HK only have port forwarding enabled, no firewall there. so it have the port 4096 enabled for forwarding |
| 01:36.00 | TazzNZ | do you run fail2ban on the HK box ? |
| 01:36.18 | TazzNZ | do you see attempted registrations on the asterisk console ? |
| 01:36.32 | Michael_Sia | sorry is port 4569 forwarded |
| 01:36.48 | Michael_Sia | let me check again on fail2ban for HK |
| 01:40.17 | Michael_Sia | fail2ban is not enabled for HK |
| 01:40.47 | Michael_Sia | sorry I am really new in asterisk, for the asterisk console, are you referring to the log? |
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| 02:23.34 | Michael_Sia | what i can see here it shows the peers is unreachable for the trunk |
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| 02:58.00 | tengulre | hi,all |
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| 03:15.32 | phix | oh hai their tengulre |
| 03:15.34 | phix | sup buddy? |
| 03:18.23 | Michael_Sia | Hi I am a new to asterisk server. Need help with my IP Phone server trunk issues. I have HK, SG, MY, TH server. Currently all can connect to the trunk of HK server except SG in only one way. which means HK -> SG ok, SG -> HK, peer unreachable. Have been working for the pass one year without issue, no settings or firewall has changed or setup, the problem comes suddenly, have tried restarted |
| 03:18.23 | Michael_Sia | the SG and HK server but no help. |
| 03:18.23 | Michael_Sia | Anyone can help advise with this issue? Let me know what information i need to provide. Thank you very much |
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| 04:42.00 | jmordica | I use adaptive jitter buffer on 13.4 and I'm wondering why it adapts quickly on the way up (during network congestion) but it is very slow to come back down to normal. It's not such a big deal but it causes the audio to be garbled while it adjust to different thresholds. Is there another param I can set to speed it up on the way back down (after network |
| 04:42.00 | jmordica | congestion is resolved)? |
| 04:44.11 | jmordica | The scenario is, I am uploading a large file, the large file takes most of the upload bandwidth, therefore the packets don't arrive at the same time and the timestamps could be out of order so the jitter buffer begins to work it's magic and slow down the stream to wait for the lagging packets to arrive. After the file is finished, it takes another minute or |
| 04:44.11 | jmordica | so to compensate for the bandwidth restoration and during this time the audio sounds slightly garbled until it has fully adjusted. |
| 04:44.50 | jmordica | I guess I'm looking for something similar to jbshrinkrate= which is a param for IAX |
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| 06:15.04 | babak | Hi, musiconhold wav files will load in RAM ? |
| 06:15.24 | babak | cahce |
| 06:16.08 | babak | I want to have 10,000 music file to users choose from |
| 06:18.17 | ChannelZ | The OS will cache a certain amount of everything on its own |
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| 08:48.41 | pazifi | hi all. in asterisk 11, how can i play a join-sound to all user, when a user joines to conferes? with confibridge "sound_join" thw sound will played only to the joined user, not to all. |
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| 10:00.37 | pazifi | hm |
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| 12:08.32 | Rewt34 | hello |
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| 14:00.16 | Rico | are the ps_* realtime tables useless if not using sorcery ? |
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| 15:19.26 | Rico | I'm getting mad with asterisk 13 |
| 15:21.17 | WIMPy | thinks that worked decently with other versions as well. |
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| 15:23.17 | mjordan | Rico: PJSIP always uses sorcery. Period. It's the only way it interacts with Realtime. |
| 15:24.09 | file | Rico, and why are you getting mad? |
| 15:24.11 | mjordan | Rico: Sorcery acts as an abstraction on top of any number of data layer backends, realtime being one of them. It provides a lot of nice things - thread safety, object lifetime consistency, as well as the ability to specify via sorcery.conf which object should be stored where |
| 15:24.41 | file | Sorcery is good(tm) |
| 15:28.28 | Rico | file: because I come from asterisk 1.8 and I used to play with realtime DB a lot |
| 15:28.34 | Rico | and chan_sip |
| 15:28.47 | Rico | many things seems new there ... |
| 15:29.26 | mjordan | there are new things. Have you looked at the guides for configuring PJSIP with Realtime on the wiki? |
| 15:29.57 | Rico | this one ? https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Sections+and+Relationships |
| 15:30.26 | mjordan | I was thinking this one: https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime |
| 15:36.38 | Rico | mjordan: mmh, ok |
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| 15:37.05 | Rico | I've seen strange things using alembic for building DB (voicemail one) |
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| 15:37.10 | Rico | mjordan: still here ? |
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| 15:57.12 | hexanol | hello |
| 15:57.21 | hexanol | I'm doing some tests with asterisk 13, was previously using asterisk 11 |
| 15:57.39 | hexanol | I have some dialplan which has something like "Dial(${INTERFACE},,t)" |
| 15:57.49 | hexanol | i.e. so the callee can transfer via the feature codes |
| 15:58.09 | hexanol | if ${INTERFACE} is SIP/foo, then it works fine, the callee can transfer via the feature code |
| 15:58.45 | hexanol | but if ${INTERFACE} is Local/123@foobar, and 123@foobar does a Dial(SIP/foo), then the callee can NOT transfer via the feature code |
| 15:58.55 | hexanol | it used to work in asterisk 11 |
| 15:59.17 | hexanol | so wondering, is this a known limitation ? is there a new way to for this kind of scenario to work ? |
| 16:00.35 | [TK]D-Fender | Show us |
| 16:00.59 | hexanol | ? |
| 16:01.01 | hexanol | I don't know |
| 16:01.52 | [TK]D-Fender | What do you mean "I don't know"? |
| 16:01.59 | [TK]D-Fender | Show us your configs and the attempt of each |
| 16:04.05 | hexanol | ok, I'll get a clean config and show you |
| 16:04.58 | [TK]D-Fender | VS? |
| 16:07.20 | file | Rico, what do you mean by "strange things"? |
| 16:07.42 | Rico | file: alembic is unable to install voicemail things |
| 16:07.44 | Rico | let me pastebin it |
| 16:08.04 | *** part/#asterisk aandrew (foobar@gromit.mixdown.ca) |
| 16:09.16 | Rico | file: http://pastebin.com/ENHb8Ajd |
| 16:10.12 | hexanol | here's the two attempts (one not-working, one working) and a part of extensions.conf: http://pastebin.com/raw.php?i=tg3ezGxh |
| 16:10.13 | file | I'd suggest filing an issue then, it should all work - if it's not then that's a problem needing to be solved |
| 16:10.44 | Rico | las time I talked about that, somebody told me that "this chan isn't alembic support one" |
| 16:10.45 | Rico | ... |
| 16:11.16 | file | what? |
| 16:11.51 | Rico | nevermind |
| 16:11.58 | Rico | I'll fill an issue tomorrow |
| 16:13.29 | [TK]D-Fender | hexanol: Don't forget that loca channels normally get optimized away <- |
| 16:13.43 | [TK]D-Fender | hexanol: You should have tested with /n at the end to prevent this |
| 16:14.32 | hexanol | I understand that they do get optimized in most case, this one included |
| 16:14.45 | hexanol | in asterisk 11 they were "optimized" too |
| 16:15.26 | hexanol | indeed, with the /n in asterisk 13 it works |
| 16:17.06 | hexanol | but the message "pbx-transfer" message is played in english (the non-local "callee" channel's language is set to fr_FR) |
| 16:17.58 | [TK]D-Fender | because the channel is the local one. Set the language in there. |
| 16:19.38 | Rico | another question about fax : which module should I use with asterisk 13 to have t38 support ? |
| 16:20.03 | Rico | spandsp ? |
| 16:20.05 | hexanol | yeah but this is a bit cumbersome / not practical in the more "real life" scenario |
| 16:20.10 | hexanol | I'll take a look at it tough |
| 16:20.17 | [TK]D-Fender | Rico: Yes. As I've heard FFA = DOA |
| 16:20.24 | Rico | DOA ? |
| 16:20.39 | Rico | don't know that |
| 16:22.21 | adeeln | i have in my sip.conf progressinband=never & prematuremedia=yes, so if a device is sending me a 183 w/SDP and i want * to send that upstream, do i just need to call the Progress() application before the dial? |
| 16:22.52 | hexanol | that said I still feel like this is a regression from asterisk 11, I'll take another look at the upgrade / changes notes |
| 16:23.44 | hexanol | (I have to go for now, I'll be back a later) |
| 16:27.08 | Rico | in menuselect I have for pjsip resource module : Depends on: pjproject(E), res_sorcery_config(M), res_sorcery_me |
| 16:27.34 | Rico | pjproject is installed |
| 16:27.45 | Rico | res_sorcery* are available and selected |
| 16:27.52 | Rico | and I can't enable pjsip |
| 16:27.54 | Rico | any tip ? |
| 16:30.07 | Rico | devel was missing |
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| 19:46.43 | zpertee | Hi all. I have hopefully a quick/simple/easy question for someone. I admittedly don't have a ton of experience with Asterisk, but am confident I can work my way through this if somebody can give me a shove in the right direction... Here's the scenario we have two external/hosted conferencing system. We broadcast church services via these. My goal is to be able to setup Asterisk, install... |
| 19:46.44 | zpertee | ...a softphone on another computer, and connect it to the Asterisk box. I then want to be able to dial extension 100, have it place two independent outbound calls, wait 3 seconds, enter the DTMF, and then bridge the two calls together. I have an audio cable running from our sound equipment to the computer where the softphone is. It uses this line as the microphone. This allows us to do a... |
| 19:46.46 | zpertee | ...broadcast. How do I get one extension to create two call paths, and how do I get them merged? |
| 19:48.52 | WIMPy | By throwing them in to the 3rd conference. |
| 19:50.44 | zpertee | WIMPy: that's the idea. I get confused when I see multiple apps. i.e. MeetMe vs Bridge... |
| 19:51.35 | WIMPy | Unless you need low latencies, forget about MeetMe. |
| 19:51.54 | zpertee | Ok. Great. Also, can I place two simultaneous calls or is the dial plan more like call first number >> dump in bridge >> call second number >> dump in bridge. |
| 19:52.19 | zpertee | I've only used dial app for basic calling... |
| 19:52.57 | WIMPy | You need to originate the other two calls. |
| 19:53.14 | zpertee | ok. great. thanks. |
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| 20:45.51 | mattsl | Can anyone help me figure out why I have a SIP phone that is not even on and doesn't show up in sip show peers but then it shows Not in Use in the queue show instead of Unavailable? |
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| 22:36.51 | Micc_ | Is there a version of asterisk where AMI can be trusted to not crash? Or what version does an alternative become available? |
| 22:37.49 | [TK]D-Fender | What are you running now? |
| 22:37.54 | WIMPy | Any that is less than 10 years old; and none so far. |
| 22:38.18 | Micc_ | I've had issues with AMI hanging my machines running 1.8.28-cert3 when I'm just using it to gather call info/device state. |
| 22:38.45 | [TK]D-Fender | <PROTECTED> |
| 22:38.55 | [TK]D-Fender | You are several releases out of date |
| 22:39.10 | [TK]D-Fender | And that brach is a few weeks away from DOA |
| 22:39.20 | [TK]D-Fender | And there have been no BUG FIXES for it for many months now |
| 22:39.24 | [TK]D-Fender | You should NOT be using it at all |
| 22:40.25 | Micc_ | What do you recommend? I've been testing 13.1-cert2 on a couple servers and having some issues with runaway cpu. |
| 22:40.48 | [TK]D-Fender | ) -=- LTS: 13.4.0 (2015/06/04), |
| 22:41.05 | [TK]D-Fender | I recommend you rad the channel topic and realize that ever version you have been putting your hands on is not up to date |
| 22:41.09 | Micc_ | Isn't certified asterisk generally more stable? |
| 22:41.24 | [TK]D-Fender | That's a THEORY |
| 22:42.12 | Micc_ | I used to always run the latest versions, then I realized my stuff was always crashing, better to find the best one and stick with it. Have been good until recently wanting to use AMI. |
| 22:43.28 | WIMPy | What exactely happens? |
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| 22:44.56 | Micc_ | WIMPy, I've not been able to get a back trace on it yet. But it looks like its working but sip set debug on shows no activity and you can't get a response to anything on 5060. |
| 22:45.31 | Micc_ | I should enable debug locks on that server. |
| 22:45.50 | WIMPy | And how is that related to AMI? |
| 22:46.04 | Micc_ | WIMPy, well, that is just a theory. |
| 22:46.20 | Micc_ | Since we've been running this on 20+ servers for years without any issues until now. |
| 22:48.42 | Micc_ | I guess I'll try 13.4.0 |
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| 22:51.23 | Micc_ | But even without AMI I do see that same server lag out from time to time. Current calls are not affected, but it stops accepting new calls for a few seconds, but then comes back. Could be adding the AMI adds more load or something that pushes it into a bigger lock state. |
| 22:51.37 | Micc_ | So might not be related to AMI at all. |
| 22:52.22 | [TK]D-Fender | The answer is : Yes, you do have to actually do a proper trace to identify this. |
| 22:55.30 | Micc_ | Well, of course. But it may not be worth it at this point, we know 1.8 is out of date. There are other good reasons to move ot 13.4, if it doesn't have the runaway cpu problem I'm seeing with 13.1cert2. |
| 22:59.03 | [TK]D-Fender | You the put out versions after that... because there were bugs to be fixed.... |
| 22:59.06 | [TK]D-Fender | #science |
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| 23:03.40 | sweettea | http://packages.asterisk.org/centos/6/asterisk-11-certified/ |
| 23:03.44 | sweettea | anyone know why this is empty? |
| 23:04.36 | WIMPy | Micc_: You should try to find out where the performance issue comes from. |
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| 23:46.48 | no2pencil | I am trying to parse the value of : ${VM_CALLERID} for email with ${VM_CALLERID:3:-7} - ${VM_CALLERID:1:-4} but it's giving the the full string up to the point that I want to cut |
| 23:47.20 | no2pencil | so say for example 8001234567 I get (8) 8001 - 8001234 |
| 23:47.25 | no2pencil | What am I doing wrong? |
| 23:49.43 | WIMPy | We can't tell you unless you tell us what you want. |
| 23:51.50 | no2pencil | I'm trying to parse ${VM_CALLERID} to be displayed (800) 123 -4567 |
| 23:52.19 | WIMPy | Well, first of all, that ar three parts, not two. |
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| 23:53.01 | WIMPy | And why do you try to use negative lengths? |
| 23:53.16 | no2pencil | right... I was expcted one would say how the digits worked, & why they go in the wrong direction |
| 23:53.39 | no2pencil | The string shows :incoming 8001234567 <8001234567> |
| 23:53.55 | WIMPy | It's offset and length. |
| 23:54.03 | no2pencil | so my thought process was that I had to back 1 off > & go 4 to the left to capture 4567 |
| 23:54.45 | WIMPy | It is from the left. |
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