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| 00:20.36 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:20.36 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); 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 |
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| 00:48.46 | [TK]D-Fender | yaymuffins, you don't need a proxy or multiple IP's |
| 00:49.16 | [TK]D-Fender | * works just fine from behind NAT. There are settings to configure like anything else |
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| 02:01.04 | yaymuffins | [TK]D-Fender, Well, there some sort of a conflict on my network, if I'm correct |
| 02:01.43 | yaymuffins | When I look at https://www.outsideopen.com/pfsense-asterisk/ , it tell me to open 10000 to 50000, it's a lot of port, but beside that, there some in the range that's already in use |
| 02:04.06 | yaymuffins | And when I tried to follow that guide, minus the couple I used already |
| 02:04.24 | yaymuffins | I had registration, but no voices coming through once in a call. |
| 02:06.51 | zigggggy | yaymuffins awesome nick |
| 02:39.51 | Samot | yaymuffins: pfsense is awful. Especially with SIP |
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| 03:26.44 | drmessano | 10000 to 50000 is WRONG too |
| 03:26.55 | drmessano | Not even the goddamn guides for PFSense get anything correct |
| 03:27.01 | drmessano | Nevermind PFsense itself |
| 03:27.12 | drmessano | Avoid |
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| 07:42.02 | pawiecki | Hi. I have incoming calls from SIP Trunk that come via incoming context. Then are sent to another context, where Asterisk dials external cellphone number. This cellphone number have the ability to transfer the call, via atxfer or blindxfer (option t in Dial). Now when cellphone user answers, and tries to transfer the call, Asterisk searches for given number in the incoming context, not the second context where the outgoing Dial was made. Can I change i |
| 07:42.04 | pawiecki | t? |
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| 08:44.51 | maurice2k | anyone in here that could help me with TALK_DETECTION and corresponding ARI event ChannelTalkingStarted? |
| 08:51.15 | maurice2k | i enabled TALK_DETECT in my dialplan and then calling statis application |
| 08:51.51 | maurice2k | but i the ChannelTalkingStarted event is never triggered |
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| 08:52.27 | maurice2k | however it is triggered as soon as i'm starting to play a sound file |
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| 11:29.58 | Jabi | hi |
| 11:30.49 | Jabi | Hi how do uou call the GSM number from SIP protocol |
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| 11:31.23 | Jabi | How to call Indian GSM number using SIP Protocol |
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| 11:33.48 | file | SIP doesn't care. You contact an upstream device or carrier and it completes the call. |
| 11:36.45 | Jabi | I have Voice blue next 2n device how to contact that device from SIP |
| 11:37.06 | Jabi | I have registered that device with my asterisk with a user account |
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| 11:59.37 | maurice2k | anyone here that has successfully used "ChannelTalkingStarted"? |
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| 12:24.48 | [TK]D-Fender | Jabi, How to call Indian GSM number using SIP Protocol <- this isn't a "thing". |
| 12:25.07 | [TK]D-Fender | You call a "GSM number" using a service provider taht has access to the PSTN |
| 12:25.17 | [TK]D-Fender | SIP doesn't magically get there by itself |
| 12:26.29 | [TK]D-Fender | If you're referring to calling OUT using tha GSM gateway then call it like you would any other provider using peer you define with the auth, codecs, etc all set |
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| 14:02.57 | imcdona | There's a linked internal Jira ticket on ASTERISK-27799 that I don't have access to. Anyone know if a fix for this is in the works? |
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| 17:31.38 | a|3x | hi |
| 17:32.05 | a|3x | im trying to upgrade my setup from asterisk 13 to 15 and im having issues |
| 17:32.57 | a|3x | im using tls and srtp |
| 17:34.29 | a|3x | i copied my previous config files but call initiator is only getting audio (one way), any idea what i am doing wrong? |
| 17:36.20 | igcewieling | a|3x: if you've not already done so read UPGRADE-14.txt and UPGRADE-15.txt, they tell you things. |
| 17:36.47 | a|3x | i skimmed that stuff on the web docs |
| 17:37.02 | a|3x | but didn't see much that i have to do |
| 17:37.08 | a|3x | my setup is rather simple |
| 17:37.39 | a|3x | im using twilio with tls and srtp, i did have to apply this patch to make it work in 13: https://gist.github.com/madsen/51d6160a906187b026d0 |
| 17:37.54 | a|3x | i ported the patch to 15 |
| 17:38.11 | a|3x | when i call the test number, i can hear the other end, but they can't hear me |
| 17:38.37 | a|3x | when i initiate a test call to my number, i can't hear them |
| 17:39.44 | a|3x | btw, things were working ok with 13 (except some issues with jitter and some other bugs) |
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| 17:43.08 | a|3x | btw i don't see UPGRADE-15.txt in the repo |
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| 18:02.01 | a|3x | so one of the bugs i have with asterisk 13 is during long calls i suddenly get very loud static and have to hang up and call back |
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| 18:56.40 | Worldexe | does loud static happen only with srtp? my users have reported similar things, and im also using asterisk 13; but im not using encryption |
| 19:08.32 | a|3x | Worldexe, i didn't try it without |
| 19:09.45 | a|3x | so i am also hearing loud static with version 15, but it starts right away and is only in one direction |
| 19:10.04 | a|3x | man.. sip is such utter shit |
| 19:10.17 | a|3x | nothing ever works right |
| 19:12.36 | a|3x | also, it depends on the client, some clients it is just silence |
| 19:17.42 | igcewieling | try not using srtp and tls, it makes things much more simple. |
| 19:19.50 | a|3x | and have the call unencrypted going through the pipes? also leak your passwords all over the place? no thanks |
| 19:20.29 | igcewieling | 1) passwords are not leaked, they are not even sent. |
| 19:20.51 | igcewieling | 2) it would tell you if the issue is with srtp/tls or with Asterisk. |
| 19:22.06 | a|3x | how does it authenticate? |
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| 19:23.25 | igcewieling | It use the nonce, pretty much the same as http digest authentication. the password is sent, but not in plaintext |
| 19:23.43 | a|3x | then it is vulnerable to replay attacks |
| 19:23.53 | a|3x | man in the middle attacks as well |
| 19:34.25 | igcewieling | It doesn't help much if you can't make it work. |
| 19:35.19 | a|3x | tbh i am having difficulties with making the non tls/srtp version work too |
| 19:35.24 | igcewieling | I'm not saying "stop using encryption" I'm saying "when something isn't working, try disabling encryption when debugging it." |
| 19:37.35 | a|3x | sure, thats a sound strategy |
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| 19:58.46 | jkroon | i'm having some trouble load res_pjsip - it complains about "load_dlopen: Error loading module 'res_pjsip': /usr/lib64/asterisk/modules/res_pjsip.so: undefined symbol: pj_ssl_cipher_name" - as I understand this should be part of pjproject but if I grep for that name in all of the pjsip libraries I come up empty handed. Ideas? |
| 20:07.43 | jkroon | ok, so I found it in ./pjlib/src/pj/ssl_sock_symbian.cpp ... now just to figure out why it doesn't actually go into lib. |
| 20:08.39 | file | because that's the platform specific code for Symbian |
| 20:08.50 | file | in the current bundled version of PJSIP in 13 I can see it |
| 20:08.58 | file | https://www.irccloud.com/pastebin/FspLuwe0/ |
| 20:09.24 | jkroon | ah ... ./pjlib/src/pj/ssl_sock_ossl.c contains an alternate. |
| 20:09.53 | file | generally stuff like that can happen if you have two installs of PJSIP on the system |
| 20:10.03 | file | headers were used for build, runtime uses other library |
| 20:10.22 | jkroon | the only ref to that function I can find at all is in the pjproject installed headers. |
| 20:10.40 | jkroon | well, that and objectdump -T res_pjsip.so indicates that it's required linking. |
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| 20:24.26 | jkroon | file, stupid bug in pjproject, if you explicitly spec --enable-ssl it actually disables it. |
| 20:27.39 | a|3x | can packet loss cause the decryption of srtp stream start producing white noise? |
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| 20:36.04 | jkroon | a|3x, that sounds sensible, but to be honest, that would be a design flaw. |
| 20:36.08 | jkroon | or a bug. |
| 20:37.40 | jkroon | rtp frames does have sequence numbers so assuming that we're using a rolling IV vector (ie, the actual key for each packet varies, and given the key, hmac and other factors, you can calculate the next key, but not previous ones, if you get our of sync and start applying the wrong per-packet keys ... white noise would make sense. |
| 20:39.39 | a|3x | what chan_sip configuration variable should i adjust to attempt to solve this issue? |
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| 21:31.29 | a|3x | so this may be related to the issue i am having, |
| 21:33.28 | a|3x | sip_xmit of 0x7f04a403d7a0 (len 977) to xxx.xxx.xxx.xxx:5060 returned -2: Broken pipe |
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| 22:17.47 | a|3x | looks like i see this sip_xmit error every time the white noise / silence happens, also, it appears to happen roughly 15 minutes into the call |
| 22:21.18 | a|3x | also there are some messages like this: 'Retransmission timeout reached on transmission' |
| 22:27.02 | [TK]D-Fender | probably either firewalled out our you have failed to bind the SIP prot due to conflist like with pjsip |
| 22:29.28 | a|3x | im using chan_sip |
| 22:29.39 | [TK]D-Fender | go verify they aren't fighting over the port |
| 22:29.49 | [TK]D-Fender | and check your firewalls on your server |
| 22:29.59 | a|3x | there is no firewall |
| 22:30.17 | [TK]D-Fender | time to prove it all |
| 22:30.56 | a|3x | can it be related to mtu? |
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| 23:02.46 | blu_ | hi guys. I'm looking to build a SIP to GSM gateway. I have a landline number in germany with sipgate. I can access it via SIP / VOIP. When I travel outside the EU, sipgate wants 1 Euro per Minute for incoming calls. I only care about incoming calls. My idea is to have a raspberry PI in the country outside the EU I travel in, which connects to sipgate via SIP / internet. I also want to attach a GSM modem to the raspi. When a call comes in via SIP |
| 23:02.47 | blu_ | on the raspi, I want it to "redirect" that call to the GSM modem attached to it. This GSM modem has a sim of the country I"m in, so has my mobile phone. The call is then to be "proxied" to my mobile phone (so in the country I'm in outside the EU, I buy two SIMs, one for the raspi and one for my phone). Is there a project for this already? If so do you have a link? |
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| 23:06.10 | [TK]D-Fender | No, there is no "project" for this. |
| 23:06.34 | [TK]D-Fender | This is a simple * config between your Pi and whatever GSM interface you're having it talk to |
| 23:10.15 | blu_ | [TK]D-Fender, the GSM interface I thought of would be a common HUAWEI UMTS Modem (E193 or similar) |
| 23:10.48 | blu_ | well project as in software ready to install and go I mean. Maybe sometihng with a nice webui that shows what calls came in and what not |
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