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| 00:33.50 | ntz | hello |
| 00:34.28 | ntz | stupid Q:, how do I explicitely print out, how the config files are assembled (I mean order) when I use #include from several of them ? |
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| 00:41.04 | ntz | okay, this looks good: http://susepaste.org/view/raw/74907093 (from `config list') |
| 00:43.39 | ntz | but how do I explicitely print out this for chan_ael ?? I don't see in `config list' ael files which are: http://susepaste.org/view/raw/77545652 |
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| 01:06.54 | ntz | anyhow, okay, let's discard my previous question .... why am I asking: according to this http://susepaste.org/view/raw/67313703 I know, that wildcard is not allowed from .ael but from other files it is |
| 01:07.04 | ntz | can I do something with it ? |
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| 11:12.47 | Haris | I have agents on (wired) LAN, at some distance, physically, yet connected to my asterisk server. This is a freepbx distro install. We are showing agent connectivity latency to be 105 ms. where-as icmp echo (ping) latency between the agent and freepbx/asterisk server is 1 ms. Its LAN!. How can one improve this latency ? |
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| 11:34.20 | sriharsha | Hi! When is the subroutine set to REDIRECTING_SEND_SUB called? |
| 11:34.53 | sriharsha | I seem to find no references to it in the documentation.. :-( |
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| 14:37.44 | Prelude2004c | good morning everyone.. has anyone tried codec2 with asterisk compared to say opus ? |
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| 14:38.03 | Prelude2004c | and also, has anyone been able to get transcoding from opus > ulaw working? mine doesn't work |
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| 15:50.17 | phrearch | hello |
| 15:51.01 | phrearch | whats the best way to debug dtls? i'm trying to setup a call, but all i see in the console is DTLS ECDH initialized (automatic), faster PFS enabled |
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| 15:56.25 | Inf0r | is pbxware asterisk platform any good? |
| 15:58.48 | [TK]D-Fender | Noboddy talks about that GUi anywhere really |
| 15:59.21 | Samot | It still exists? |
| 15:59.30 | Samot | Like, updated and maintained? |
| 16:01.29 | [TK]D-Fender | Their site seems to say so. |
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| 16:11.43 | cresl1n | phrearch: usually wireshark is your friend |
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| 17:24.36 | igcewieling | It would be awesome if this message tells the IP address. [Mar 6 12:23:14] WARNING[2677] chan_sip.c: Retransmission timeout reached on transmission b11223f8585eadc713c4303d254c6ed98b0c7b50c8f134b06b0-0117-4635 for seqno 1 (Critical Response) |
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| 19:04.50 | jeffspeff | using PJSIP and app queue. when the queue sends a call to a queue member with multiple pjsip contacts does the call go to only the first contact returned or to all contacts registered? |
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| 19:21.59 | Samot | I'm going to guess it's going to call all the AORs. |
| 19:22.06 | Samot | Like normal. |
| 19:23.59 | jeffspeff | Samot, so you're saying it will behave like Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})}) as described in https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels ? |
| 19:26.27 | Samot | Aah PJSIP handles dialing AORs differently. |
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| 19:26.47 | rmudgett | app_queue knows nothing about PJSIP_DIAL_CONTACTS, You'll have to use a local channel member to do that kind of dialing. |
| 19:26.51 | Samot | Dial(PJSIP/100) will look for the AOR that matches that contact. |
| 19:27.31 | jeffspeff | thanks rmudgett |
| 19:28.22 | Samot | rmudgett: So you have to use a local channel to dial each contact, correct? |
| 19:29.23 | rmudgett | You would need a local channel to do Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})}} |
| 19:30.33 | Samot | OK, that's what I thought.. |
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| 19:31.09 | Bhakimi | what is a good timing source to get the highest CPS on asterisk 11? |
| 19:31.24 | Samot | How many CPS? |
| 19:31.25 | Bhakimi | currently use DAHDI and its not performing so well |
| 19:31.40 | Bhakimi | i mean as high as possible without causing deadlocks |
| 19:32.13 | igcewieling | Bhakimi: there is no one thing which make "max CPS". |
| 19:32.17 | Samot | Asterisk 11 shouldn't be using DAHDI for very little. |
| 19:32.23 | Samot | In regards to timing. |
| 19:32.29 | Bhakimi | at some point i get these messages "Exceptionally long voice queue length queuing to" and all sip peers become unavaliable |
| 19:32.47 | Bhakimi | im just wondering which time source would perform best when it comes to CPS |
| 19:33.13 | Bhakimi | i specifcily exluded all timing sources and only use dahdi |
| 19:33.16 | Samot | res_timing_pthread.so <-- This is used first |
| 19:33.20 | Bhakimi | i assume it was the most stable source for timing |
| 19:38.33 | igcewieling | have you looked at the page for the timers? |
| 19:38.46 | rmudgett | The timing module priorities are timerfd, dahdi, pthread. From most preferred to least. |
| 19:39.47 | igcewieling | https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces |
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| 19:52.22 | Bhakimi | looks like asterisk preffers dahdi over pthread for iax trunks |
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| 20:02.09 | [TK]D-Fender | <Bhakimi> what is a good timing source to get the highest CPS on asterisk 11? <- you shouldn't bee using 11 |
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| 20:03.41 | phrearch | hi |
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| 20:05.22 | Bhakimi | our application was build around 11, if i had the resources to rewrite the whole thing to be compatible with 13 or 14 i would |
| 20:05.44 | Bhakimi | im stuck with 11 for at least the next few years |
| 20:08.10 | igcewieling | Bhakimi: you will be told many times to use Asterisk 13 and there are many good reasons to use recent Asterisk versions. However, I also have no plans to upgrade past Asterisk 11, mainly because of a large installed base of customer owned Asterisk servers which I maintain. |
| 20:10.03 | Bhakimi | understood =) |
| 20:10.35 | Bhakimi | it looks like restarting asterisk i can push the cps high for a while until it craps out |
| 20:10.53 | Samot | How many CPS are you pushing? |
| 20:11.00 | Bhakimi | then after restarting it again it can handle it until it throws those error messages again |
| 20:11.07 | Bhakimi | like 8 |
| 20:11.16 | Samot | Every second? |
| 20:11.32 | Bhakimi | yea |
| 20:11.57 | Samot | That's around 1200-1300+ sim calls. |
| 20:12.06 | Samot | If they are short. |
| 20:12.30 | Bhakimi | they aqre all short |
| 20:12.36 | Bhakimi | avr call is 20 seconds |
| 20:12.39 | Samot | What type of server? |
| 20:12.48 | Bhakimi | dell r610 |
| 20:12.57 | [TK]D-Fender | Old box too... |
| 20:13.03 | Bhakimi | dual quad core, with hyp thread its at 16 cores |
| 20:13.11 | Bhakimi | the load avg is at 1 |
| 20:13.22 | Bhakimi | its not even touching the cpu |
| 20:13.59 | Bhakimi | sorry r620 |
| 20:14.41 | zaf | anyone having issues compiling with bundled pjsip today? I'm getting errors running './configure --with-pjproject-bundled' where the wget dns times out connecting to pjsip.org, but when i wget manually, it downloads fine |
| 20:20.10 | dan_j | zaf: nope |
| 20:20.22 | dan_j | what asterisk version are you building? |
| 20:20.27 | zaf | latest 13.14 |
| 20:21.30 | dan_j | 'm running trunk. One moment. I'll give 13.14 a go. |
| 20:24.30 | zaf | sigh, it's dns/routing |
| 20:24.32 | dan_j | zaf: no problem with bundled here |
| 20:24.45 | zaf | thanks for checking |
| 20:27.03 | zaf | strangely, wget manually was failing over to the secondary server in resolv.conf, but running through ./configure was not... |
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| 21:02.32 | Kobaz | soooooo, any idea why a polycom would connect to the dhcp-given option 66 provisioning server just once |
| 21:02.35 | Kobaz | and then never again? |
| 21:03.01 | Kobaz | i'm dumping traffic at the local switch and all it does it get dhcp and then just sit there... it doesn't even *try* and connect to anything |
| 21:03.25 | zerohalo | did the prov server send a config that turned off option 66? |
| 21:03.32 | Kobaz | nope |
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| 21:08.52 | Kobaz | i wish the web interface would work on this |
| 21:11.45 | zaf | Kobaz, do you have the phone jumping to an invalid/unreachable vlan? |
| 21:11.48 | igcewieling | Kobaz: idea: phone downloaded the config files, which have some sort of setting which prevents them from connecting again. |
| 21:12.17 | igcewieling | happened to me once or twice. 8-| |
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| 21:26.31 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 21:44.47 | Kobaz | zerohalo: i figured it out |
| 21:44.52 | elguero | Kobaz: What version are you working with? Is Vlan involved? |
| 21:45.00 | Kobaz | zerohalo: same old, same old, lldp on polycom never works right |
| 21:45.09 | Kobaz | it's talking on both vlan 1 and vlan 50 |
| 21:45.16 | zerohalo | ahhh |
| 21:45.20 | Kobaz | even though the vlan is hard coded on the phone to vlan 50 |
| 21:45.26 | Kobaz | and lldp is sending out vlan 50 |
| 21:45.35 | Kobaz | you need vlan filtering enabled on the polycom to completely isolate it |
| 21:45.37 | Kobaz | so annoying |
| 21:45.41 | Kobaz | almost 4 hours on this |
| 21:45.57 | Kobaz | i forget about this, because some firmwares it works on, and some it doesn't |
| 21:48.37 | Kobaz | it's provisioning now |
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