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| 00:47.33 | ars | hi all. I have a setup similar to https://jtanx.github.io/2016/02/24/using-asterisk-to-route-calls-through-mobile/. I have both the asterisk system and my softphone on the same lan, no issues of NAT. I'm only seeing audio going from the softphone to the asterisk system, nothing going the other way. PCAP shows SDPs are showing the correct IP addresses. I'm not certain why asterisk isn't sending audio, even comfort noise/blank audio. Any suggestions w |
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| 01:05.06 | Samot | ars: You're saying that there's no audio between the mobile phone and Asterisk? |
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| 01:29.05 | ars | Samot: hard to say, as it's a bluetooth connection. On the pcap, I see audio from the softphone (on a pc) going to the asterisk system. No audio coming back. Doing a pcap from the asterisk system itself, not the pc. |
| 01:29.36 | Samot | Show an Asterisk SIP debug of the call. |
| 01:29.40 | Samot | asterisk -rvvvvvvv |
| 01:29.43 | Samot | sip set debug on |
| 01:29.46 | Samot | ~pb |
| 01:29.46 | infobot | rumour has it, 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. |
| 01:36.34 | zaf | having problems trying to get DPMA working on asterisk 13.14. i've followed the wiki as closely as i can, but getting 'No matching endpoint found' when the phone tries to send the SIP message. config file and error paste at pastebin.ca/3775012 |
| 01:40.06 | ars | Samot: http://pastebin.com/Vw7CW8pc |
| 01:40.53 | zaf | oops, dpma config got munged in my paste, new one at pastebin.ca/3775014 |
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| 01:44.08 | ars | Samot: here's a second one that doesn't have as much register stuff to it: http://pastebin.com/9hKyJ5iP |
| 01:44.37 | Samot | ars: Your softphone isn't offering any codecs |
| 01:45.24 | Samot | ars: Tell X-Lite to use at least gsm or ulaw |
| 01:46.14 | Samot | But that is the issue. X-Lite isn't offering codecs. |
| 01:46.33 | ars | It's set to both. I had it set only to gsm, becuase ulaw was giving the same problem. Second pb you can see it in line 104 |
| 01:47.06 | ars | In the first one you can see line 608 it shows the peer (softphone) allows gsm |
| 01:49.04 | lorsungcu | ars: rtp set debug on |
| 01:49.17 | ars | lorsungcu: you want it concurrently with SIP? |
| 01:49.37 | lorsungcu | just confirm it's sending sometihng to 192.168.1.204 |
| 01:49.46 | lorsungcu | should see it both directions |
| 01:50.16 | lorsungcu | then go confirm that the softphone has a valid way to present audio to you (proper output device selected, etc) |
| 01:50.32 | ars | sure, it's not, one moment |
| 01:52.02 | ars | lorsungcu: http://pastebin.com/hkwJQrxi |
| 01:53.04 | ars | the really screwy thing is that you'll see in line 484, I hit a few numbers. number tones via rfc2833 come through. No voice. I can hook up two softphones on the local network and have two-way audio. |
| 01:53.13 | ars | without reinvites (they're disabled) |
| 01:54.14 | lorsungcu | establish the call and pastebin 'sip show channels' |
| 01:54.30 | ars | ok, killing debug then. |
| 01:55.36 | ars | http://pastebin.com/1GzcAwLC |
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| 01:56.07 | lorsungcu | no idea how to troubleshoot bluetooth stuff :) |
| 01:56.19 | ars | yeah, that's where I'm hamstrung |
| 01:56.39 | ars | But the oddball thing is that there's zero RTP from asterisk to the softphone |
| 01:56.50 | ars | I mean, it should even be sending idle noise |
| 02:00.30 | lorsungcu | ars: mobile search shows your phone? |
| 02:01.20 | ars | can't do a search because it's already bound. mobile show devices shows it connected |
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| 04:55.27 | drmessano | ars: What Bluetooth dongle are you using? |
| 05:18.49 | Samot | Dongle. |
| 05:19.23 | Samot | Dangerously Dangling Dongle. |
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| 10:21.26 | MacroMan | What is `tag` in relation to callerid? I see it as an available datatype in the wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_CALLERID |
| 10:22.17 | MacroMan | And can I use it for my own purpose to pass along a flag with a call? |
| 10:35.11 | MacroMan | Think I've answered my own question. Doing `Set(CALLERID(tag)=myTag)` doesn't actually change the tag parameter in an invite header. |
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| 11:28.01 | dan_j | Still having issues with pjsip and v13. :( |
| 11:28.16 | dan_j | Please can someone explain this issue that causes all traffic to stop working. |
| 11:28.19 | dan_j | https://www.irccloud.com/pastebin/XWJM3pz5/ |
| 11:28.53 | dan_j | I had this issue in 13.11 and was hoping it was fixed in 13.14 but it doesn't seem to be. |
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| 12:52.11 | Samot | Show the peer that is having this issue |
| 12:52.24 | Samot | Because there's no transport called Asterisk |
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| 13:03.15 | dan_j | It's all the peers that are having the issue when it occurs. |
| 13:03.36 | dan_j | Samot: It's fine for days, and then goes bananas. |
| 13:06.12 | dan_j | Samot: Just PMd you a fuller log |
| 13:07.07 | Samot | Grep for this 24834 <-- |
| 13:07.15 | Samot | Show the logs releated to that log id. |
| 13:07.20 | Samot | Show the logs related to that log id. |
| 13:07.23 | WIMPy | Is that a chan_sip compatibility setting? *scnr* |
| 13:09.27 | dan_j | Samot: Sent via PM |
| 13:09.57 | dan_j | WIMPy: Hi. Is that a question to me? If it is, I didnt understand it. |
| 13:10.18 | WIMPy | was just being sarcastic. |
| 13:12.30 | Samot | dan_j: Is your customer in the UK? |
| 13:12.41 | dan_j | Most of them are in the UK |
| 13:12.45 | Samot | OK |
| 13:14.34 | dan_j | Created an issue, hopefully the powers that be have seen this before. https://issues.asterisk.org/jira/browse/ASTERISK-26836 |
| 13:18.13 | Samot | 27.109.9.91 <-- That IP was in the same debug you just gave me. |
| 13:18.35 | Samot | Does that customer have agents in India? |
| 13:20.13 | dan_j | Yes. That sounds right. That client is has offices in a few countries. |
| 13:22.01 | *** join/#asterisk grodriguezdata4 (be9e7d93@gateway/web/cgi-irc/kiwiirc.com/ip.190.158.125.147) |
| 13:22.17 | Samot | Are they the only ones having the issue? |
| 13:24.44 | dan_j | The issue affects every endpoint |
| 13:25.07 | dan_j | Not just specific clients |
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| 13:29.12 | Samot | dan_j: Show all your transport conext's. |
| 13:31.34 | grodriguezdata4 | there, I keep with the same trouble, i can´t keep my asterisk server working using my ISP´s gateway http://www.arris.com/products/touchstone-telephony-gateway-tg862/, I found this: http://support.esi-estech.com/index.php?/Knowledgebase/Article/View/261/80/blacklist-routers-with-known-sip-issues, does anyone have some idea... thanks in advance ;-) |
| 13:34.37 | Samot | Did your ISP say that VoIP should work on the connection? |
| 13:35.17 | dan_j | Samot: ive just had to step out. Will be back in 20 minutes. |
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| 13:44.41 | grodriguezdata4 | Samot: They said that the service is capable to handle VOIP calls, based on their softphone service or the virtual telephone, i can use my analogical number on a softphone provided by them an make and receive calls |
| 13:46.19 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:46.42 | Samot | You can't have an analog line on a softphone. |
| 13:46.57 | Samot | So this sounds like their voice service is interfering with yours. |
| 13:51.26 | grodriguezdata4 | They provided this service as an special feature, they offer you the possibility to use the same hired number on a softphone and make or receive calls if you have an incoming call it first ring on you hardphone then after like 9 rings the sofphone rings |
| 13:53.28 | Samot | This is an issue with their service/equipment. |
| 13:53.35 | Samot | You're going to have to work with them. |
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| 13:56.21 | dan_j | https://www.irccloud.com/pastebin/Dl8ANssT/ |
| 13:56.26 | dan_j | Samot: ^^ |
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| 14:00.19 | Samot | All all the peers in the database have the proper transport? |
| 14:01.21 | dan_j | I've not defined the transport for the peers as they could connect from external (transport-public-udp) or they could connect via a VPN connection and end up using (transport-private-udp) |
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| 14:03.12 | dan_j | They are all defined like this. |
| 14:03.19 | dan_j | https://www.irccloud.com/pastebin/xjOlNtZM/ |
| 14:03.42 | [TK]D-Fender | bind = 127.0.0.1 |
| 14:03.59 | [TK]D-Fender | funny... |
| 14:04.27 | dan_j | [transport-local] nothing wrong with that. it's used by pacemaker to confirm asterisk is running. |
| 14:05.49 | [TK]D-Fender | earlier PB looks like some sort of memory leak |
| 14:06.42 | dan_j | yep. i've reported it but apart from that output, i've got no other diagnostic info to share. hopefully, it's not just me! if it is just me, i assume it's going to take a long time to fix! |
| 14:07.35 | Samot | You only have the one server? |
| 14:07.54 | dan_j | Got a few. This issue has happened on both over the last 30 days |
| 14:08.11 | [TK]D-Fender | dan_j, Have you tried this on the last 2-3 releases? |
| 14:09.07 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 14:09.20 | dan_j | this is 13.14 |
| 14:09.25 | [TK]D-Fender | nvm..your bug post covered ... ALL of this.. |
| 14:09.43 | [TK]D-Fender | Seems thorough |
| 14:09.50 | [TK]D-Fender | Bump it |
| 14:10.13 | dan_j | However, I've got installed the latest commit from source on one of my other servers with pjsip 2.6. Waiting to see if it still dies. |
| 14:10.40 | dan_j | s/got/just |
| 14:11.34 | dan_j | Bump it? |
| 14:15.17 | [TK]D-Fender | the thread |
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| 14:28.24 | dan_j | [TK]D-Fender: you mean just add another comment to it? |
| 14:32.35 | [TK]D-Fender | How long since last activity? |
| 14:32.48 | dan_j | 2 hours |
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| 14:37.29 | dan_j | Submitted it, shortly after my initial post here as it was clear to me to be some sort of leak |
| 14:41.30 | [TK]D-Fender | ok, too soon for a bump then... |
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| 15:39.03 | zaf | people may find thise useful - uses 'jq' and 'crudini' on bash to create an includable file for DPMA firmware config - http://pastebin.ca/3775233 |
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| 16:27.07 | MacroMan | I've been noticing a few of these in my log: WARNING[1468]: db.c:288 db_execute_sql: Error executing SQL (COMMIT): database is locked |
| 16:27.30 | MacroMan | I don't make any db queries as far as I know and I've only been seeings these recently. |
| 16:28.35 | MacroMan | Anything fishy going on? |
| 16:30.59 | igcewieling | Asterisk uses SQLite to store things like registrations. |
| 16:31.55 | MacroMan | Ah OK. Should I be concerned with these warnings? |
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| 16:34.49 | igcewieling | I would be. the file is usually /var/lib/asterisk/astdb.sqlite3 maybe there is a .lock file laying around |
| 16:35.40 | MacroMan | Can't see a .lock file in that dir |
| 16:35.47 | *** part/#asterisk EnrgySmth (d8eba101@gateway/web/freenode/ip.216.235.161.1) |
| 16:35.59 | MacroMan | I have a sqlite.db file too |
| 16:36.18 | MacroMan | Which I assume is the actual data |
| 16:41.49 | igcewieling | is the directory owned by the user Asterisk runs as? |
| 16:41.53 | igcewieling | (and the file too) |
| 16:42.02 | MacroMan | yes |
| 16:42.30 | MacroMan | I see, I'm getting whenever a sip client registers |
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| 16:42.51 | MacroMan | But they do seem to register OK |
| 16:43.57 | MacroMan | I restarted Asterisk and they're not showing for registrations now |
| 16:44.10 | MacroMan | Thanks. I'll keep it monitored |
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| 17:47.05 | ars | drmessano: Broadcom Corp. BCM20702A0 Bluetooth 4.0 |
| 17:48.25 | ars | it's made by IOGear |
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| 20:22.33 | sakifahmed | hello i am trying to install asterisk on ubuntu server. Everything is running good but i can not browse site on browser. Note: on this server i run two more server librenms & observium with apache2. is there anybody can tell me what can be the reason ? |
| 20:24.24 | igcewieling | Asterisk does not have a web based GUI |
| 20:25.42 | sakifahmed | i installed asterisk-gui |
| 20:26.21 | *** join/#asterisk Chotaire (chotaire@oahu.chotaire.net) |
| 20:26.57 | igcewieling | you might want to ask on #asterisk-gui. So few people use it you might have some trouble finding help. |
| 20:27.27 | sakifahmed | thnx |
| 20:27.51 | *** part/#asterisk sakifahmed (673e9052@gateway/web/freenode/ip.103.62.144.82) |
| 20:30.18 | [TK]D-Fender | Darn... |
| 20:30.24 | [TK]D-Fender | wish I was there for that... |
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| 20:43.40 | newtonr | [TK]D-Fender, lol |
| 20:44.31 | [TK]D-Fender | Because you know ... the beta release now supports * 1.8! |
| 20:58.00 | *** join/#asterisk lankanmon (~LKNnet@CPEa84e3fc94903-CMa84e3fc94900.cpe.net.cable.rogers.com) |
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| 21:11.22 | fridgefire | Hi, I build and configured Asterisk 13.11 on my machine and it works great. I also build Asterisk 13.14 on an external server with the same config file but with that one, video doesn't work. I do not get errors (core set verbose 3). Is it possible that this is a firewall problem? Audio works. |
| 21:11.40 | *** join/#asterisk newtonr (~newtonr@173-17-133-211.client.mchsi.com) |
| 21:11.40 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 21:15.47 | fridgefire | Or is it possible to get more logs? |
| 21:17.04 | *** join/#asterisk sysadmin999 (~josh@95.211.168.148) |
| 21:17.32 | sysadmin999 | hi everyone, i am wondering if asterisk has a capability to detect SDP / ringing when it happens? |
| 21:19.22 | [TK]D-Fender | Depends when you mean |
| 21:19.30 | [TK]D-Fender | You shouldn |
| 21:19.38 | [TK]D-Fender | 't get that in the middle of a bridged call for instance... |
| 21:20.00 | sysadmin999 | on a sip channel, how can i detect that sdp is starting/started? |
| 21:20.04 | sysadmin999 | before connection is made? |
| 21:21.54 | *** join/#asterisk friedrich (~friedrich@ubermail.me) |
| 21:25.44 | [TK]D-Fender | It doesn't just "happen" |
| 21:26.12 | [TK]D-Fender | You should normally only get ringing (SIP 180 usually)) when you are doing an INVITE out. |
| 21:26.29 | sysadmin999 | correct, how can i detect that 180?? |
| 21:26.50 | [TK]D-Fender | And there is nothing for "your" to detect. * gets itt. * wil also either pass it on OOB, or inband as audio if the channel bettwen the other leg and * has been established. |
| 21:27.15 | [TK]D-Fender | The entire idea of "detecting" something random at some random time in the middle of all other processing is not an * thing |
| 21:27.37 | [TK]D-Fender | Describe a very specific scenario you are trying to accomodate |
| 21:27.47 | [TK]D-Fender | Because "just detect" is SUPER VAGUE |
| 21:28.38 | tompaw | Guys, does anyone know Switchvox® forVMware® pricing? |
| 21:29.04 | [TK]D-Fender | www.digium.com or their resellers probably do |
| 21:29.11 | [TK]D-Fender | IRC doesn't offer pricing |
| 21:29.14 | [TK]D-Fender | We don't sell it |
| 21:30.23 | tompaw | found it, thx. |
| 21:31.38 | [TK]D-Fender | ...and he's gone. |
| 21:32.02 | [TK]D-Fender | packs up to head home |
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| 23:33.19 | Samot | Jesus, what's with n00bs PMing me about simple things?! |
| 23:33.38 | Samot | You know what I mean, drmessano?! |
| 23:34.00 | drmessano | STFU |
| 23:35.23 | Samot | ð ð ð |
| 23:35.26 | drmessano | HAH |
| 23:35.29 | drmessano | Got it |
| 23:36.19 | Samot | What was it? |
| 23:36.34 | drmessano | Not mono |
| 23:36.40 | drmessano | Logic was fine |
| 23:41.18 | Samot | Ahhh |
| 23:41.47 | drmessano | Having one of those days where nothing is going right |
| 23:42.02 | drmessano | and im thinking "I've done this before, but not this way.. does this even work?" |
| 23:43.19 | *** join/#asterisk Bhakimi (~textual@208.78.137.2) |
| 23:44.11 | drmessano | Actually I dont think i've ever tried that |
| 23:44.30 | freebs | relax |
| 23:44.53 | drmessano | Because using the VM destinations to go to the announcement to then go back to the VM back seemed like too much inception to work |
| 23:44.59 | Bhakimi | hi guys, i got a very odd issue, i have 10 asterisk boxes that are trunked to eachother, 9 of them place outbound calls and send those to a single server using a iax trunk. randomly all of the 9 servers plcing calls all at the same time print this error: chan_sip.c: Hanging up call 50b63cf119fbf7c9732ff9504bad069f@xxx.xxx.xxx.xxx:5060 - no reply to our critical packet, once this happens no calls placed work and all sip channels become unresponsive |
| 23:47.47 | *** part/#asterisk kharwell (kharwell@nat/digium/x-zomyhivhoheckkxs) |
| 23:55.43 | Samot | It's all Twins Peaksy |
| 23:56.22 | Samot | How do IAX trunks trigger a chan_sip error? |
| 23:58.41 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:59.12 | [TK]D-Fender | Samot, MAGIC |