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| 12:34.37 | DannyA | hey all. how do you set a custom variable on an endpoint and then use that variable in the dialplan? |
| 12:34.45 | DannyA | i want to define a custom caller ID for a few endpoints |
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| 12:44.40 | Samot | DannyA: Setting a custom variable in an endpoint is an endpoint feature. So does your endpoint allow it? |
| 12:44.52 | Rico | hello. facing a fu**ing issue for hiurs now. asterisk is listening on 0 |
| 12:44.53 | DannyA | i'm using PJSIP with wizard |
| 12:44.57 | Rico | 0.0.0.0:5060 |
| 12:45.15 | Rico | tcpdump show SIP requests on NICs, but nothing in asterisk debug |
| 12:45.22 | Samot | DannyA: So you mean a Chan_PJSIP endpoint. Not like a phone/device endpoint. |
| 12:45.27 | sibiria | Rico: firewall? |
| 12:45.39 | sibiria | tcpdump sniffs on the interface, *before* firewall |
| 12:46.29 | DannyA | @Samot correct |
| 12:46.55 | Samot | set_var=NAME=VALUE |
| 12:47.02 | DannyA | wait, no. it's an IP phone. it's configured as an endpoint in pjsipwizard.conf |
| 12:47.08 | Rico | sibiria: no firewall, no selinux |
| 12:48.12 | sibiria | pjsip logging is enabled i presume? |
| 12:48.52 | sibiria | "pjsip set logger on" |
| 12:49.20 | sibiria | and your "debug=>" entry in logger.conf is correctly set to include debug messages? |
| 12:49.48 | sibiria | (or whatever you call your log file entry) |
| 12:50.02 | Rico | sibiria: I'm using chan_sip |
| 12:50.15 | sibiria | please don't |
| 12:50.23 | Rico | yes I know |
| 12:50.31 | Rico | but that is an old (big) install |
| 12:50.41 | Rico | can't update right now |
| 12:50.45 | Rico | but it's planned |
| 12:50.51 | sibiria | "sip set debug on" for chan_sip |
| 12:51.46 | Rico | sibiria: as I said, asterisk does not receive any SIP message |
| 12:51.51 | Rico | even if debug is enabled |
| 12:52.30 | Rico | strace -f -e trace=network -p $(pidof asterisk) does not show packets neither |
| 12:52.42 | sibiria | yeah it really does sound as if you're firewalling |
| 12:52.56 | sibiria | but i believe you when you say you verified that none is running |
| 12:53.42 | Rico | yes |
| 12:53.43 | Rico | I checked |
| 12:53.52 | Rico | and nc show packets on host |
| 12:53.57 | Rico | un UDP:5060 |
| 12:54.36 | Rico | s/un/on/ |
| 12:56.14 | sibiria | so if you shut asterisk down and run netcat as listener |
| 12:56.23 | sibiria | something like nc -ul 5060 |
| 12:56.27 | sibiria | do you see stuff coming in then? |
| 12:56.38 | Rico | sibiria: just to resume : tcpdump show SIP incoming packets. nc -luv -p 5060 show SIP incoming packets. strace -f -e trace=network -p $(pidof asterisk) does not show SIP incoming packets |
| 12:56.55 | sibiria | that is odd |
| 12:57.10 | Rico | yes, that's why I'm asking here |
| 12:57.11 | sibiria | i bet it's something very simple and stupid, like it usually is, but i'm at a loss right now |
| 12:57.12 | Rico | :] |
| 12:57.36 | Rico | working on that for hours now, and I'm not a beginner in networking / asterisk |
| 12:57.58 | Rico | I'm just.... don't know where to look for |
| 12:57.58 | Samot | DannyA: If you want to set the custom setting/value on the phone then the phone must support it. |
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| 12:58.31 | Samot | Rico: Is this Chan_PJSIP or Chan_SIP? |
| 12:58.34 | DannyA | right, but if i just want the variable to be set on whatever device is registering with that endpoint, then i just need to define it the way you mentioned, right? |
| 12:58.46 | Samot | DannyA: Yes. |
| 12:58.49 | DannyA | ok, thanks! |
| 12:58.54 | Rico | Samot: chan_sip |
| 12:59.09 | Samot | Rico: What version of Asterisk again? |
| 12:59.37 | Rico | 16.12 |
| 12:59.41 | Rico | Samot: ^ |
| 12:59.50 | Samot | OK and you've noloaded Chan_PJSIP |
| 13:00.01 | Samot | Or told it to NOT listen on 5060? |
| 13:00.01 | sibiria | you mentioned that it's a very old setup... are you perhaps by mistake loading pjsip as well? |
| 13:00.08 | sibiria | or using some pre-16 config thing that may be interfering? |
| 13:00.19 | Samot | sibiria: Using anything pre-12 |
| 13:00.27 | Rico | Samot: pjsip is not loaded |
| 13:00.27 | Samot | Chan_PJSIP has been around since 12. |
| 13:00.32 | Rico | only chan_sip is |
| 13:00.35 | Samot | You're sure? |
| 13:00.39 | Samot | OK. |
| 13:00.44 | Rico | yes I'm sure |
| 13:00.51 | sibiria | so you're not autoloading it by mistake, but explicitly noloading it |
| 13:00.56 | Rico | yes |
| 13:01.38 | Samot | Why? |
| 13:01.52 | Rico | I've got warnings like these ones too : |
| 13:01.54 | Rico | WARNING[7682] chan_sip.c: sip_xmit of 0x5603ebf3b790 (len 586) to 192.168.43.179:5060 returned -1: Interrupted system call |
| 13:02.13 | sibiria | if it were me i would at this point revisit my sip.conf thoroughly, and following that just replace the current config with a simple default and see if that gets any traffic, just to get an indication on whether it's a host network issue or an asterisk config issue |
| 13:02.20 | Rico | Samot: because it's a freepbx install which has been made before freepbx was compatible with pjsip |
| 13:02.39 | Samot | Rico: None of those versions support Asterisk 16 |
| 13:02.55 | Samot | FreePBX has been using Chan_PJSIP since FreePBX 12 |
| 13:03.09 | Samot | Chan_PJSIP became the primary driver for FreePBX in v13 |
| 13:03.14 | Samot | It's been like 6 years. |
| 13:03.15 | Rico | yeah, as I said it is an old install |
| 13:03.25 | Samot | Then it's an install that won't support Asterisk 16 |
| 13:03.41 | Samot | You're talking about FreePBX that is 11 or older. |
| 13:03.50 | Rico | fwconsole ma list | grep core |
| 13:03.50 | Rico | | core | 14.0.28.71 | Activé | GPLv3+ | |
| 13:03.54 | Samot | WTF |
| 13:04.00 | Rico | we upgraded. We just did not moved to pjsip |
| 13:04.03 | Samot | 14 has Chan_PJSIP as the primary |
| 13:04.13 | Samot | OK. |
| 13:04.59 | Rico | I'm just trying to understand why asterisk (chan_sip module) does not see incoming SIP requests |
| 13:05.04 | Rico | just this |
| 13:05.14 | Samot | Uhm. |
| 13:05.21 | Samot | Rico: How are you on Asterisk 16.12 |
| 13:05.26 | Rico | asterisk registers to SIP gateways, so outgoing trafic work |
| 13:05.50 | Rico | home made packages (which work correctly) on dozens of install |
| 13:06.07 | Samot | OK so you aren't using the repo |
| 13:06.56 | Rico | nope |
| 13:07.29 | Samot | So you installed Asterisk outside the supproted way in the FreePBX distro. |
| 13:07.36 | Samot | supported* |
| 13:08.16 | Rico | yes ! installed it years ago |
| 13:08.19 | Rico | worked for years |
| 13:08.27 | Samot | What is years ago? |
| 13:08.27 | Rico | just stopped working this morning |
| 13:08.39 | Rico | don't know, I was not in this company |
| 13:08.54 | Samot | So when did this get updated to 14? |
| 13:09.12 | Rico | don't know, not me who made it |
| 13:09.28 | Samot | So it was already on 14 when you got there? |
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| 13:10.54 | Rico | yes |
| 13:11.08 | Samot | Well here's the deal. |
| 13:11.18 | Samot | You're having issues with Chan_SIP which is pretty much unsupported. |
| 13:11.42 | Samot | You're having issues with the FreePBX distro in which you or someone installed Asterisk from somehting home rolled. |
| 13:12.00 | Samot | So outside the normal supported method for the distro which uses an RPM release. |
| 13:12.28 | Samot | What has changed recently? |
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| 13:13.48 | Rico | Samot: I would say "nothing" but ... something may has changed |
| 13:14.01 | Rico | have |
| 13:14.03 | Samot | So no one did updates, added or removed things? |
| 13:14.16 | Samot | Is this only on of your installs having a problem? |
| 13:14.25 | Rico | we dist-upgrade and upgrade to 16.12, expecting this would fix our pbm |
| 13:14.36 | Rico | Samot: yes, first problem like this |
| 13:14.49 | Samot | OK but 16.12 isn't in the repos. |
| 13:14.50 | Rico | we have about 200-250 installs like this one |
| 13:14.56 | Samot | So you installed Asterisk how? |
| 13:15.05 | Samot | How did you update to 16.12? |
| 13:15.06 | Rico | grom our repositories |
| 13:15.09 | Rico | from |
| 13:15.18 | Samot | Which is an RPM you built? |
| 13:15.21 | Samot | ??? |
| 13:15.24 | Rico | a .deb, but yes |
| 13:15.29 | Samot | Wait |
| 13:15.35 | Samot | The disto is RHEL |
| 13:15.41 | Rico | who sai that ? |
| 13:15.42 | Rico | said |
| 13:15.46 | Samot | SAngoma. |
| 13:15.50 | Rico | ?? |
| 13:15.59 | Rico | cat /etc/debian_version |
| 13:15.59 | Rico | 10.5 |
| 13:16.01 | Samot | Is this a distro install of FreePBX? |
| 13:16.03 | Samot | JFC. |
| 13:16.27 | Samot | So this is a MANUAL install of FreePBX. |
| 13:16.30 | Rico | yep |
| 13:16.35 | Rico | :D |
| 13:16.45 | Samot | Explain yourself better going forward. |
| 13:16.49 | Rico | forget about freepbx |
| 13:17.01 | Rico | let's just talk about asterisk |
| 13:17.16 | Samot | Well what version was it on when the problem started? |
| 13:17.23 | Rico | 16.2 |
| 13:17.34 | Samot | And what did you do then to troubleshoot? |
| 13:17.55 | Rico | the exact same thing : nc, strace, sip debug... |
| 13:18.47 | Samot | OK show the result of "sip show settings" |
| 13:19.41 | Rico | Samot: https://paste.swordarmor.fr/Pbzj |
| 13:20.13 | Samot | UDP Bindaddress: 192.168.32.1:5060 |
| 13:20.19 | Samot | That's not 0.0.0.:5060 |
| 13:20.23 | Samot | And you're behind NAT. |
| 13:21.04 | Rico | we change for the tests. phones are in 192.168.32.0/20 |
| 13:21.09 | Samot | 8:44:57 AM <Rico> 0.0.0.0:5060 |
| 13:21.16 | Samot | GD. |
| 13:21.21 | Rico | yes, we changed this a few mins ago |
| 13:21.24 | Rico | sorry |
| 13:21.31 | Rico | but same problem with 0.0.0.0 |
| 13:21.50 | Rico | but it may be a problem on the OS side |
| 13:21.52 | Rico | maybe |
| 13:22.02 | Samot | OK so what is in front of this? |
| 13:22.08 | Samot | Since it's behind NAT. |
| 13:22.16 | Rico | strace does not see the packets relayed to pid of asterisk |
| 13:22.33 | Samot | Show a packet. |
| 13:22.37 | Rico | no. SIP phones are not behind NAT. SIP phones are in 192.168.32.0/24, IPBX is 192.168.32.1 |
| 13:22.47 | Rico | let me pastebin |
| 13:23.27 | Rico | Samot: https://paste.swordarmor.fr/mhUt |
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| 13:25.55 | Rico | what would cause the OS or kernel to not pass ip packets to a process ? |
| 13:26.02 | bbhoss | iptables |
| 13:26.10 | bbhoss | bad routing table |
| 13:26.30 | bbhoss | not listening on the right bind address |
| 13:27.31 | Rico | checked all of them... :'( |
| 13:30.30 | Samot | Does this happen to devices on the /24? |
| 13:30.43 | Samot | The 192.168.74.0/24? |
| 13:31.28 | Rico | Samot: I have no SIP devices directly connected on this subnet. |
| 13:31.42 | Rico | I used to, but not anymore |
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| 13:48.10 | Rico | Samot: chan_sip.c:4321 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data |
| 13:48.15 | Rico | nice |
| 13:49.45 | Rico | mmh, forger |
| 13:49.47 | Rico | forget |
| 13:50.09 | igcewieling | Why are you using chan_sip? |
| 13:52.36 | Rico | igcewieling: backlog :) |
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| 18:40.23 | the_1 | Hi |
| 18:41.24 | the_1 | is there a way to change the RTP Packetization time for a codec when using pjsip driver in asterisk 16? |
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| 19:22.51 | kfife | Is it possible to ask Asterisk from the CLI what my SIP password is (i.e. for provisioning scripting) or must I parse it from users.conf / sip.conf etc? |
| 19:23.26 | kfife | not 'mine' per say, but user x |
| 19:25.01 | robmal | grep -A 10 $username /etc/asterisk/sip.conf | grep secret | sed 's/secret=//' |
| 19:25.24 | robmal | This is the "parsing" you're asking about? |
| 19:27.07 | kfife | I'm asking if Asterisk can be asked for a SIP secret as an alternative to [duh]. |
| 19:28.00 | robmal | AFAIK not. |
| 19:29.23 | kfife | kinda what one might expect, but years ago I kinda remember it being displayed for the asking. |
| 19:32.24 | robmal | Oh, ok. Lets wait, maybe someone else knows the trick, i'll report it as a security issue ASAP :-) |
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| 22:27.51 | igcewieling | You could use the CONFIG() function. |
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| 23:40.01 | Samot | Ast_config but that would do the trick |
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| 23:53.39 | kfife | igcewieling: example? |