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| 01:11.25 | gusto | ah |
| 01:11.29 | gusto | here someone uses IAX2? |
| 01:33.18 | wyoung | I do |
| 01:33.34 | wyoung | but I am nearly off my lunch break |
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| 05:27.34 | misc-- | hi I have a strange problem. We have two internet connections connected to our router. Asterisk has externip to public IP 1. All traffic goes out fine with correct source address. When I change externip=<public IP2> and then run a traffic capture on the router, I can see asterisk udp packets sending out from public IP1 where it should be in fact public IP2 |
| 05:28.07 | misc-- | the only way asterisk will send correct source IP is if I reboot the asterisk server. Restarting asterisk service, restarting networking has no affect - packets still have a source IP of public IP1 |
| 05:28.46 | misc-- | I'm at a complete loss as to what's going on and how to fix it. Any ideas? Every now and again we need to change asterisk to go to the second internet connection and it always requires a reboot (which takes 5 mins and is way too much down time) |
| 05:34.38 | MrTAP | misc--: Look at your router. When you change the externip in Asterisk you'll need to flush the Network Address Table in the router so the connections can all reestablish through the correct connection. |
| 05:35.56 | misc-- | MrTAP: hmmm yeah that's what I am doing. I already flush the cache. If I ping from the asterisk server, everything is fine - it goes out the correct interface with correct source IP. It's just that asterisk udp packets are incorrect. And to fix, it's nothing to do with the router but just reboot the asterisk box |
| 05:37.07 | MrTAP | are you doing a packet capture from the WAN side of the router or from the Asterisk box itself? |
| 05:38.21 | misc-- | from the wan side of the router |
| 05:39.02 | misc-- | so, on the router, I set traffic from asterisk to go over public IP 2 (interface 2). I then capture traffic from interface 2 but it shows UDP packets from public IP 1 |
| 05:39.16 | MrTAP | rebooting your asterisk server will kill all of the connections, ie. they'd timeout while it reboots. I'd still suggest looking at the router because that sounds exactly like it's still mapping the existing connections to the old IP... new connections would correctly be mapped to the new IP. |
| 05:39.46 | MrTAP | If you can view a status page on the router to show all of the addres mappings you might be able to validate that's what's happening htere. |
| 05:40.17 | MrTAP | I suspect stopping the asterisk service for a few minutes would have the same result as rebooting that machine altogether |
| 05:40.27 | misc-- | that's what I was thinking, that it's existing connections. Didn't think that after reboot that because it takes so long, that it times out and fixes itself. Possibility I guess |
| 05:41.08 | misc-- | yeah, interesting, never though of that. It's a linux box so all I really do is just an ip route flush cache which fixes all new connections. I'll see if I can figure out how to flush existing connections |
| 05:42.26 | MrTAP | this might point you in the right direction http://conntrack-tools.netfilter.org/conntrack.html |
| 05:43.39 | misc-- | ok great, thanks =) |
| 05:44.02 | MrTAP | No prob. Dual WAN always makes things a little more interesting :) |
| 05:44.50 | misc-- | oh for sure... I thought I had it 100% under control, except for this issue. Can be confusing how it all works |
| 05:46.36 | MrTAP | Another thing to familiarize yourself is ip_conntrack_udp_timeout which is pretty self explanatory |
| 05:46.44 | *** join/#asterisk gerhard7_ (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 05:47.11 | MrTAP | But simply flushing all existing connections when you switch WANs would likely be best, especially if you're switching because one failed. |
| 05:50.26 | misc-- | ok I will check that out |
| 05:52.55 | misc-- | yeah I just assumed that doing a route flush would fix it, which it does for everything except for udp. I'll flush all connections and see. Just don't want to lock myself out. Will set up an auto reboot in case something goes wrong |
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| 05:56.49 | misc-- | ahhhhh........ |
| 05:57.10 | misc-- | conntrack -D may have fixed it |
| 05:58.14 | MrTAP | :) |
| 05:58.52 | misc-- | yeah.. do you know how long I've battled with that issue? I just gave up and did a reboot every time, whch is only a handful of times a year but having a busy phone system down for 5 mins is never good |
| 05:59.55 | MrTAP | Haha yeah, I've experienced that and spent way too many hours learning how the conntrack and nat modules work. I recognized your issue from first hand experience. |
| 06:01.24 | misc-- | yeah.. really appreciate that! I'll have to read up a bit more on all of that |
| 06:02.07 | MrTAP | also: the sip modules work well, you don't need to tell Asterisk the externip. the linux box will inspect and rewrite the SIP and SDP addresses quite accurately in y experience, as long as you're sticking with UDP and TCP. No good with TLS of course since the linux box can't see into the packets. |
| 06:02.19 | misc-- | I remember when the official iptables tutorial site was so small back in the day. It's grown so big, I haven't kept up with that in ages |
| 06:02.55 | MrTAP | Yeah, iptables is really it's own ecosystem now |
| 06:03.09 | misc-- | oh right I see. Ok well I'll take a look at that as well |
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| 08:38.49 | bazou2 | Hello, I have an asterisk 13 and a kamailio 4.4, Asterisk delays in responding to INVITE messages |
| 08:39.12 | bazou2 | kamailio keeps sending invite messages without any response |
| 08:39.51 | bazou2 | <PROTECTED> |
| 08:40.03 | bazou2 | to note that the 2 severs are behind a NAT |
| 08:40.52 | *** join/#asterisk bipolar (~bipolar@offsite.guru) |
| 08:41.14 | bazou2 | I thought that the issue is related to DNS but am using chan_sip in * |
| 08:42.30 | pawiecki | bazou2: so it' the same subnet, and they are on the same switch and ping each other ok? |
| 08:42.38 | bazou2 | @help is the issue related to network /NAT ? or what it' weird because sometimes the asterisk responds immediatly |
| 08:43.38 | pawiecki | bazou2: can you check if asterisk or server is busy while responds are delayed? Maybe there's high traffic when this happens? |
| 08:44.26 | bazou2 | yes same subnet and they can ping each other |
| 08:44.34 | pawiecki | are pings or other packets being sent and received at the same time? Are you sure it's not a switch/cable issue? |
| 08:44.53 | bazou2 | the 2 servers are built on aws |
| 08:45.34 | bazou2 | and the asterisk is connected to a database outside the network |
| 08:47.30 | pawiecki | bazou2: can you show us sip logs / debug with this issue? |
| 08:49.04 | *** join/#asterisk SoBlindWolf (~SoBlindWo@go.pcshost.co) |
| 08:49.12 | bazou2 | yes a moment |
| 08:59.14 | bazou2 | 2017-11-28 08:14:40.206404 IP 192.168.0.7.5060 > 192.168.0.5.5060: SIP: INVITE sip:callingnumber@192.168.0.5 SIP/2.0 |
| 08:59.14 | bazou2 | 2017-11-28 08:14:40.256608 IP 192.168.0.5.5060 > 192.168.0.7.5060: SIP: SIP/2.0 401 Unauthorized |
| 08:59.14 | bazou2 | 2017-11-28 08:14:40.257537 IP 192.168.0.7.5060 > 192.168.0.5.5060: SIP: ACK sip:callingnumber@192.168.0.5 SIP/2.0 |
| 08:59.14 | bazou2 | 2017-11-28 08:14:40.331995 IP 192.168.0.7.5060 > 192.168.0.5.5060: SIP: INVITE sip:callingnumber@192.168.0.5 SIP/2.0 |
| 08:59.16 | bazou2 | 2017-11-28 08:14:40.776503 IP 192.168.0.7.5060 > 192.168.0.5.5060: SIP: INVITE sip:callingnumber@192.168.0.5 SIP/2.0 |
| 08:59.19 | bazou2 | 2017-11-28 08:14:41.776422 IP 192.168.0.7.5060 > 192.168.0.5.5060: SIP: INVITE sip:callingnumber@192.168.0.5 SIP/2.0 |
| 08:59.24 | bazou2 | 2017-11-28 08:14:43.776454 IP 192.168.0.7.5060 > 192.168.0.5.5060: SIP: INVITE sip:callingnumber@192.168.0.5 SIP/2.0 |
| 08:59.27 | bazou2 | 2017-11-28 08:14:47.776508 IP 192.168.0.7.5060 > 192.168.0.5.5060: SIP: INVITE sip:callingnumber@192.168.0.5 SIP/2.0 |
| 08:59.30 | bazou2 | 2017-11-28 08:14:51.777484 IP 192.168.0.7.5060 > 192.168.0.5.5060: SIP: INVITE sip:callingnumber@192.168.0.5 SIP/2.0 |
| 08:59.33 | bazou2 | 2017-11-28 08:14:55.776432 IP 192.168.0.7.5060 > 192.168.0.5.5060: SIP: INVITE sip:callingnumber@192.168.0.5 SIP/2.0 |
| 08:59.36 | bazou2 | 2017-11-28 08:14:59.776470 IP 192.168.0.7.5060 > 192.168.0.5.5060: SIP: INVITE sip:callingnumber@192.168.0.5 SIP/2.0 |
| 08:59.39 | bazou2 | 2017-11-28 08:15:03.776449 IP 192.168.0.7.5060 > 192.168.0.5.5060: SIP: INVITE sip:callingnumber@192.168.0.5 SIP/2.0 |
| 08:59.42 | bazou2 | 2017-11-28 08:15:07.776465 IP 192.168.0.7.5060 > 192.168.0.5.5060: SIP: INVITE sip:callingnumber@192.168.0.5 SIP/2.0 |
| 08:59.45 | bazou2 | 2017-11-28 08:15:18.545222 IP 192.168.0.5.5060 > 192.168.0.7.5060: SIP: SIP/2.0 100 Trying |
| 08:59.47 | bazou2 | 2017-11-28 08:15:18.783913 IP 192.168.0.5.5060 > 192.168.0.7.5060: SIP: SIP/2.0 100 Trying |
| 08:59.49 | bazou2 | 2017-11-28 08:15:18.786667 IP 192.168.0.5.5060 > 192.168.0.7.5060: SIP: SIP/2.0 100 Trying |
| 08:59.53 | bazou2 | 2017-11-28 08:15:18.786834 IP 192.168.0.5.5060 > 192.168.0.7.5060: SIP: SIP/2.0 100 Trying |
| 08:59.55 | bazou2 | 2017-11-28 08:15:18.786910 IP 192.168.0.5.5060 > 192.168.0.7.5060: SIP: SIP/2.0 100 Trying |
| 08:59.57 | bazou2 | 2017-11-28 08:15:18.786987 IP 192.168.0.5.5060 > 192.168.0.7.5060: SIP: SIP/2.0 100 Trying |
| 08:59.59 | bazou2 | 2017-11-28 08:15:18.787064 IP 192.168.0.5.5060 > 192.168.0.7.5060: SIP: SIP/2.0 100 Trying |
| 09:00.01 | bazou2 | 2017-11-28 08:15:18.787149 IP 192.168.0.5.5060 > 192.168.0.7.5060: SIP: SIP/2.0 100 Trying |
| 09:00.03 | bazou2 | 2017-11-28 08:15:18.787258 IP 192.168.0.5.5060 > 192.168.0.7.5060: SIP: SIP/2.0 100 Trying |
| 09:00.05 | bazou2 | 2017-11-28 08:15:18.787357 IP 192.168.0.5.5060 > 192.168.0.7.5060: SIP: SIP/2.0 100 Trying |
| 09:00.07 | bazou2 | 2017-11-28 08:15:19.236873 IP 192.168.0.5.5060 > 192.168.24.190.5060: SIP: INVITE sip:callingnumber%7C60%7CHRgrL(1980000%3A61000%3A30000)@192.168.24.190 SIP/2.0 |
| 09:00.10 | bazou2 | 2017-11-28 08:15:19.239321 IP 192.168.24.190.5060 > 192.168.0.5.5060: SIP: SIP/2.0 100 Trying |
| 09:00.12 | bazou2 | 2017-11-28 08:15:21.405940 IP 192.168.24.190.5060 > 192.168.0.5.5060: SIP: SIP/2.0 183 Session Progress |
| 09:00.14 | bazou2 | 2017-11-28 08:15:21.406724 IP 192.168.0.5.5060 > 192.168.0.7.5060: SIP: SIP/2.0 183 Session Progress |
| 09:00.16 | bazou2 | where 192.168.0.7 kamailio 192.168.0.5 is the asterisk |
| 09:00.27 | bazou2 | these are the logs on the asterisk side |
| 09:07.16 | pawiecki | bazou2: You didn't... |
| 09:07.34 | bazou2 | sorry? |
| 09:07.36 | pawiecki | Why would you paste logs inside chat? Please use paste |
| 09:07.39 | pawiecki | ~paste |
| 09:07.39 | infobot | rumour has it, paste is http://pastebin.org/ or http://bin.cakephp.org/ or http://pastebin.ca/ |
| 09:17.32 | bazou2 | https://pastebin.com/CRRgUgf6 |
| 09:17.40 | bazou2 | sorry i didn't notice |
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| 09:21.20 | pawiecki | bazou2: those logs are from the asterisk? |
| 09:21.41 | bazou2 | yes |
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| 09:24.01 | pawiecki | technically it sends response to every INVITE, but with massive delay. Can you verify, that asterisk was not abnormally busy (or the whole server) at that time? |
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| 09:34.45 | bazou2 | asterisk is connected to a2billing database outside the network through a VPN, can this be related to the issue? |
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| 09:39.28 | pawiecki | bazou2: in my opinion no, because it should respond to INVITE before it can process a call, if I'm not mistaken. That is if you don't have some exotic custom setup, that operates on SIP packets |
| 09:46.57 | bazou2 | could it be a misconfiguration in sip.conf file? |
| 09:52.17 | pawiecki | bazou2: Can you show the call log, from * CLI, with SIP debug active?? |
| 09:52.33 | pawiecki | sorry about that second question mark :) |
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| 12:01.11 | [sID] | Trying to install wanpipe I have inserted a sangoma card, and gets the information: |
| 12:01.25 | [sID] | "FATAL: Error inserting wanpipe (/lib/modules/2.6.32-696.el6.x86_64/kernel/drivers/net/wan/wanpipe.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
| 12:01.28 | [sID] | Warning: Dahdi modules not found: Wanpipe not build for DAHDI |
| 12:01.30 | [sID] | Error: Dahdi and/or Zaptel modules not found! Please install DAHDI and compile Wanpipe for DAHDI! |
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| 12:56.41 | mmlj4 | you have a kernel/module mismatch, or need to rebuild wanpipe with daddy for your current kernel |
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| 20:56.23 | rrittgarn | in voicemail.conf for the emailbody contents, we previously had a variable ${UNIQUEID} in the body of the email to track issues with messages. I'm not seeing this variable in any documentation and am wondering if it never should have worked or if it's been deprecated. |
| 20:57.22 | rrittgarn | right now it produces an almost accurate uniqueid that i would expect to see in the CDR, but i don't see any log entries matching it, nor are there any CDR entries matching it |
| 20:57.55 | rrittgarn | so i guess the question is: Is it too far gone and i need to get creative to solve the issue (as in it should have never worked), or are there other variables that are less commonly used, but are still there? |
| 20:58.17 | rrittgarn | trying to associate a specific voicemail to a specific call record is the goal |
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| 20:59.02 | rrittgarn | (this is all to associate stuff outside asterisk - specifically when * sends emails) |
| 21:10.05 | MrTAP | I'm not sure you can really access anything like that from the voicemail app without changing code |
| 21:10.10 | MrTAP | When did it stop working for you? |
| 21:13.11 | rrittgarn | I'm not sure. I just noticed today that the ID isn't accurate |
| 21:13.17 | rrittgarn | so i don't know where the heck its coming from |
| 21:13.49 | MrTAP | From my understanding a channel is spawned by the voicemail app so it can process the message, so the uniqueid you get from there is for it's own channel, not the one the call was associated with |
| 21:14.30 | MrTAP | I could be mistaken on that but going from memory with Ast 11 I seem to recall making that discovery when working on something similar |
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| 21:22.39 | rrittgarn | that makes sense because I can't find any record of that unique id in my logs |
| 21:22.48 | rrittgarn | so if it was just an internal channel it would not show up |
| 21:23.03 | rrittgarn | i saw a hacky way of doing it by manipulating the caller id variables |
| 21:23.10 | rrittgarn | but was hoping for something a little more eloquent |
| 21:23.41 | MrTAP | Yeah I think I ultimately just created my own patch to make a few more channel variables visible to the voicemail system to solve my needs |
| 21:24.18 | MrTAP | If you compile you're own it's not too tricky to follow what's going on inside app_voicemail.c |
| 21:24.30 | MrTAP | search for X-Asterisk- |
| 21:34.07 | rrittgarn | yeah i saw that for imap stuff |
| 21:35.08 | MrTAP | So there are many ways to do it but to give you just one idea, here's what I did for one of my applications: |
| 21:35.22 | MrTAP | Copy one of those linse to the non-imap section, so it inserts that header into the regular notification mails |
| 21:35.46 | MrTAP | in voicemail.conf set sendmail to your own script |
| 21:36.08 | MrTAP | your own script can then access those headers and adjust the message body as necessary, for example by doing it's own substitution or otherwise processing it before sending it to sendmail for real |
| 21:40.34 | rrittgarn | yeah i'm already using a custom one and that's where i found the uniqueid 'problem' |
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