| 00:00.54 | TheGallopingFox | and the base |
| 00:10.42 | twitchnln | any cisco guys in here ever setup rotary phone on fxs port, anyone know if its possible? |
| 00:11.41 | [TK]D-Fender | depends on the interface |
| 00:11.46 | [TK]D-Fender | obviously |
| 00:12.01 | [TK]D-Fender | it's a phone |
| 00:12.07 | [TK]D-Fender | the only question is signalling |
| 00:12.28 | reveal | Does this mean firewall or permissions(for sockets) issues? |
| 00:12.30 | reveal | chan_sip.c:4263 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data |
| 00:14.21 | [TK]D-Fender | could be firewall blocking you from sending out |
| 00:14.38 | [TK]D-Fender | it could be a global failure of the SIP stack due to a screwup in port assignment |
| 00:14.50 | [TK]D-Fender | it could be because it's asked to send to an impossible destination |
| 00:14.54 | [TK]D-Fender | I'm betting on #2 |
| 00:25.25 | *** join/#asterisk j-fish (uid178161@gateway/web/irccloud.com/x-tkdeoeutlsptfvdn) |
| 00:50.03 | *** join/#asterisk ganbold (~ganbold@173.244.215.173) |
| 00:51.40 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |
| 00:53.11 | *** join/#asterisk woddy (~woddy@unaffiliated/woddy) |
| 01:18.41 | *** join/#asterisk TheGallopingFox (~TheGallop@gateway/vpn/privateinternetaccess/thegallopingfox) |
| 01:21.52 | *** part/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 02:19.56 | twitchnln | D-Fender-- 4fxs-did can and will :-) |
| 02:40.33 | *** join/#asterisk shanth_ (~shanth@wsip-98-182-126-226.ph.ph.cox.net) |
| 02:54.45 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 02:55.15 | *** join/#asterisk lankanmon (~LKNnet@CPE64777dd7e053-CM64777dd7e050.cpe.net.cable.rogers.com) |
| 02:59.59 | KNERD | I have compiled , installed, and entered ldconfig for Iksemel, then ran the configure for Asterisk 14, but chan_motif still has XXX in the selction in menuselect. Suggestions? |
| 03:16.37 | KNERD | though, I did hear pjsip jas a similar module which can do the same as iksemel |
| 03:24.46 | Samot | This isn't for Google Voice is it? |
| 03:26.51 | Samot | Because Google shut down third party XMPP connections in June. |
| 03:26.59 | Samot | Some may lag out there, but it's done. |
| 03:27.05 | Samot | GV is google only now. |
| 03:27.15 | Samot | Hangouts, etc. |
| 03:34.25 | *** join/#asterisk lankanmon (~LKNnet@CPE64777dd7e053-CM64777dd7e050.cpe.net.cable.rogers.com) |
| 03:49.50 | KNERD | Samot: Well they said that, but it still functions |
| 03:53.03 | *** join/#asterisk lankanmon (~LKNnet@CPE64777dd7e053-CM64777dd7e050.cpe.net.cable.rogers.com) |
| 03:57.07 | Samot | Like I said, some may lag. |
| 03:57.19 | Samot | But it's not something I would put trust in. |
| 03:59.10 | KNERD | it stopped working, then * foks pathced it |
| 03:59.21 | KNERD | so it's been workign for me for couple of years now |
| 03:59.44 | Samot | Back in Feb. Google announced EOL stuff |
| 03:59.54 | Samot | Google Voice was on the list for June 26th |
| 04:00.09 | Samot | When it would start being shut down and stopping third party XMPP connections |
| 04:00.16 | *** join/#asterisk mmlj4 (~mmlj4@47-44-49-2.static.unas.mo.charter.com) |
| 04:00.30 | Samot | Working previous years has no bearing on future. |
| 04:00.42 | KNERD | I think they announced something similar back in 2014 |
| 04:01.02 | Samot | I'm just letting you know. |
| 04:01.25 | KNERD | yeah, but till then just wanted to get chan_motif functioning |
| 04:01.39 | *** join/#asterisk MajesticFudgie (~MajesticF@178-32-65-132.eur1.stack.ilkotech.co.uk) |
| 04:01.42 | Samot | Well that's something I can't help with. I've never used it. |
| 04:01.50 | KNERD | okay |
| 04:01.57 | MajesticFudgie | Is there any reason a phone would ring when it's not connected o-o |
| 04:01.59 | Samot | I was just give information so you can base a choice on it |
| 04:02.12 | Samot | What do you mean "not connected"? |
| 04:02.32 | MajesticFudgie | It's not connected to the SIP server |
| 04:02.33 | Samot | You mean not registered to Asterisk or not on a network? |
| 04:02.40 | Samot | But it's on a network? |
| 04:02.52 | MajesticFudgie | Its on the network but not connected to asterisk. |
| 04:02.53 | Samot | And that network is connected to the Internet? |
| 04:03.00 | MajesticFudgie | Yet I got a call from 1004@myhomeip |
| 04:03.03 | MajesticFudgie | yeah |
| 04:03.12 | Samot | Then your network is being scanned |
| 04:03.16 | KNERD | very common |
| 04:03.18 | Samot | Phones listen on ports |
| 04:03.23 | MajesticFudgie | Ah |
| 04:03.25 | Samot | Fix your firewall |
| 04:03.26 | MajesticFudgie | What ports? |
| 04:03.26 | KNERD | you can disable that |
| 04:03.30 | Samot | Fix your firewall |
| 04:03.38 | Samot | Someone is probing your network |
| 04:03.44 | KNERD | in some phones there are setting for acctping calls only from trusted sourcs |
| 04:03.50 | Samot | OK |
| 04:03.57 | Samot | That doesn't fix the bigger issue |
| 04:04.16 | Samot | The network is being scanned |
| 04:04.19 | Samot | So someone is in it |
| 04:04.45 | MajesticFudgie | how they're in I dont know as the firewalls pretty tight except for certain common ports |
| 04:04.59 | Samot | Are those certain common ports open to everyone? |
| 04:05.09 | Samot | And do they need to be open to everyone? |
| 04:05.29 | MajesticFudgie | Open to everyone, but closed now |
| 04:05.41 | Samot | Is the PBX local? |
| 04:05.43 | MajesticFudgie | Were only Minecraft iirc and nother |
| 04:05.46 | Samot | Or is it hosted somewhere? |
| 04:05.49 | MajesticFudgie | yeah all my phone stuff is internal |
| 04:05.59 | MajesticFudgie | Asterisk on my home server and phone on the same network |
| 04:06.05 | Samot | So there is no need to have it open to the world for SIP |
| 04:06.14 | MajesticFudgie | It's not |
| 04:06.22 | MajesticFudgie | The firewall doesnt allow SIP in |
| 04:06.30 | Samot | They are ringing your phones |
| 04:06.33 | MajesticFudgie | It'll ofc let outgoing via asterisk and such |
| 04:06.33 | Samot | You know how they do that |
| 04:06.37 | Samot | they send an INVITE |
| 04:06.48 | MajesticFudgie | How the invite go in I dont know |
| 04:06.53 | MajesticFudgie | got* |
| 04:07.07 | MajesticFudgie | The firewall is tight on all ports except a few which I've now removed |
| 04:08.26 | Samot | This is common place. |
| 04:08.53 | Samot | They will trying to scan the network for SIP devices by sending OPTIONS and/or INVITEs to see if the INVITE will be passed |
| 04:09.11 | MajesticFudgie | But how would they scan the network if the firewall is tight? |
| 04:09.13 | Samot | They also see if they can then get to the phones GUI or other management |
| 04:09.18 | Samot | I don't know |
| 04:09.22 | Samot | What ports did you have open? |
| 04:09.31 | KNERD | Usually they use a program called "SIP Vicous" . On some phones there is a " Allow IP Call" you can turn off |
| 04:09.32 | MajesticFudgie | Right now none, but I did have 25565 open |
| 04:09.51 | Samot | And it's deny all otherwise? |
| 04:09.55 | MajesticFudgie | yes |
| 04:10.00 | MajesticFudgie | It's using my ISP's firewall |
| 04:10.08 | Samot | Perhaps they got in that way, who knows. |
| 04:10.11 | Samot | But they were in. |
| 04:10.13 | MajesticFudgie | that comes with the router, configured via gui |
| 04:10.33 | MajesticFudgie | Would be nice if I had logs of anything but I don't sadly |
| 04:14.45 | MajesticFudgie | tried sipvicious on my network now and it found nothing |
| 04:19.16 | KNERD | i guess you closed it off then |
| 04:20.22 | drmessano | This phone isnât registered to some provider ? |
| 04:20.58 | MajesticFudgie | No its a polycom pointed to my asterisk server on a home server |
| 04:21.13 | MajesticFudgie | and turns out my phone has a web gui, changed the password from default but I doubt anyone got into it |
| 04:21.26 | drmessano | Odd. Youâve never configured it to work with some provider? |
| 04:21.51 | KNERD | if your PBX availabel through the network? |
| 04:22.03 | drmessano | I ask because of itâs regged to some outside host, ISP routers sometimes donât randomize ports |
| 04:22.15 | drmessano | So they use 5060 for the NAT |
| 04:22.18 | MajesticFudgie | Phone -> Asterisk -> SipGate |
| 04:22.30 | drmessano | Uh |
| 04:22.35 | drmessano | That explains it |
| 04:22.37 | MajesticFudgie | Asterisk is connecting to sipgate through the firewall and sipgate would report the number |
| 04:22.49 | MajesticFudgie | The phone number as 1004@my.ip.here |
| 04:22.52 | drmessano | Your router isnât using random ports |
| 04:22.55 | MajesticFudgie | where my.ip.here is my external ip |
| 04:23.27 | drmessano | You need to disallow anonymous sip |
| 04:23.54 | drmessano | And or replace the router |
| 04:24.07 | MajesticFudgie | Latter I can't do |
| 04:24.13 | MajesticFudgie | and disable anonymous sip on what? |
| 04:24.25 | KNERD | i bet just turning off Anonymous SIP calls will do it |
| 04:24.25 | drmessano | The PBX |
| 04:24.27 | KNERD | yoru PBX |
| 04:24.34 | KNERD | in SIP settings |
| 04:24.51 | drmessano | Is this a FreePBX box? |
| 04:25.19 | MajesticFudgie | No |
| 04:25.23 | drmessano | Didnât think so |
| 04:25.28 | MajesticFudgie | I'm running Asterisk on a home server |
| 04:25.31 | drmessano | So no âsip settingsâ |
| 04:25.32 | MajesticFudgie | which connects to SipGate |
| 04:25.38 | drmessano | Yes we know |
| 04:25.44 | MajesticFudgie | any configuration I can do is directly in the asterisk configs |
| 04:25.51 | drmessano | Right |
| 04:25.52 | MajesticFudgie | Not bothered with any gui crap |
| 04:26.00 | drmessano | So itâs allowguest I believe |
| 04:26.34 | KNERD | then why asking about all that in FreePBX? |
| 04:26.39 | MajesticFudgie | What cfg would that be in? |
| 04:26.51 | drmessano | sip.conf |
| 04:27.09 | drmessano | KNERD |
| 04:27.20 | drmessano | FFS |
| 04:27.24 | drmessano | Wrong channel |
| 04:27.33 | KNERD | allowguest=no |
| 04:28.09 | KNERD | in the [general] settings |
| 04:28.09 | drmessano | He's not in the wrong channel, YOU are |
| 04:28.19 | KNERD | oh...heh... |
| 04:28.26 | KNERD | I am in righ tchannel. |
| 04:28.35 | drmessano | This is #asterisk |
| 04:28.41 | KNERD | I was thinking this was FreepBX |
| 04:28.47 | drmessano | We know |
| 04:28.53 | MajesticFudgie | http://paste.ubuntu.com/25442948/ |
| 04:29.08 | MajesticFudgie | Thats my sip.conf pretty much, removed any passwords and other stuff thats irrelevant |
| 04:29.58 | MajesticFudgie | eh I left a password in but it's useless without my other details |
| 04:30.02 | drmessano | Are you saying allowguest was set? |
| 04:30.07 | drmessano | or you just added it? |
| 04:30.18 | MajesticFudgie | thats how it was all set |
| 04:30.24 | MajesticFudgie | I've not touched the config in months |
| 04:31.19 | Samot | The INVITE never touched the PBX |
| 04:31.26 | Samot | It went straight to the phone. |
| 04:31.40 | MajesticFudgie | How it got to the phone I really dont know |
| 04:31.46 | Samot | Nor do I |
| 04:31.51 | Samot | I am telling what happened |
| 04:31.56 | MajesticFudgie | I know |
| 04:32.24 | drmessano | Something opened that port |
| 04:32.28 | KNERD | try sipvious from the outside |
| 04:32.33 | drmessano | This sounds like a shitty NAT issue |
| 04:32.38 | MajesticFudgie | I have KNERD, it turned up nothing |
| 04:32.42 | KNERD | you can try alwaysauthreject=yes |
| 04:32.45 | drmessano | Like the phone poked a hole |
| 04:32.47 | Samot | You were probed like a redneck during an abduction. |
| 04:32.53 | KNERD | in that same GENERAL section |
| 04:32.55 | MajesticFudgie | Samot lul |
| 04:33.09 | Samot | KNERD |
| 04:33.13 | MajesticFudgie | KNERD I doubt it came through asterisk I checked logs and found nothing |
| 04:33.21 | Samot | Again, that is a band aid |
| 04:33.32 | Samot | The call never touched the PBX to begin with |
| 04:33.34 | MajesticFudgie | Last thing in the logs was asterisk regaining connection as I've been working on sorting my home network |
| 04:33.46 | Samot | This isn't something an Asterisk setting is going to fix |
| 04:33.54 | Samot | This is something a proper firewall is going to fix |
| 04:34.00 | drmessano | I missed the part about it being the one phone only |
| 04:34.02 | drmessano | So year |
| 04:34.04 | drmessano | So yeah |
| 04:34.07 | Samot | Yeah, I figured. |
| 04:34.08 | drmessano | the phone |
| 04:34.13 | MajesticFudgie | Well I have a softphone on my pc |
| 04:34.15 | drmessano | Well |
| 04:34.19 | MajesticFudgie | Which never got affected |
| 04:34.22 | Samot | FFS. |
| 04:34.23 | Samot | Right |
| 04:34.36 | Samot | Your PC probably has a firewall |
| 04:34.38 | KNERD | usualy you can see anonymous calls hitting a CDR if you have it set up |
| 04:34.38 | Samot | It's Windows |
| 04:34.42 | drmessano | 99.999% of the time this happens, it's because a phone is connected to an outside peer through a shitty nat |
| 04:34.44 | Samot | ???? |
| 04:34.50 | Samot | How does that help this? |
| 04:34.51 | drmessano | and it uses port 5060 and not some random port |
| 04:34.58 | Samot | The INVITE went to the PHONE |
| 04:34.59 | MajesticFudgie | The networks as secure as I can get it |
| 04:35.01 | Samot | Not the PBX |
| 04:35.18 | drmessano | The phone doesnt have a second account on it? |
| 04:35.23 | drmessano | Like a backup to sipgate? |
| 04:35.24 | MajesticFudgie | What I'd like to know is why the call showed my outgoing ip |
| 04:35.35 | MajesticFudgie | Rather than any other IP |
| 04:35.40 | drmessano | The phone doesnt have a second account on it? |
| 04:35.44 | drmessano | Like a backup to sipgate? |
| 04:35.46 | MajesticFudgie | No drmessano |
| 04:35.55 | MajesticFudgie | Only one line is active |
| 04:36.03 | MajesticFudgie | the other is named but isnt connected to anything |
| 04:44.01 | MajesticFudgie | I ran nmap on the sip phone |
| 04:44.12 | MajesticFudgie | It has 80, 443 and 5060 open |
| 04:44.21 | KNERD | well yes, |
| 04:44.38 | MajesticFudgie | 80 and 443 are the gui |
| 04:44.50 | MajesticFudgie | Is 5060 _required_ or may there be an option to close it? |
| 04:44.56 | drmessano | It is |
| 04:45.03 | KNERD | yeah, if you dont want to receive calls on it |
| 04:45.04 | drmessano | It's SIP |
| 04:45.08 | drmessano | You kinda need it |
| 04:45.28 | MajesticFudgie | Well the Phone connects to SIP, wouldn't that be how it should recieve and place calls? |
| 04:45.43 | drmessano | The phone is SIP |
| 04:45.52 | MajesticFudgie | Or does it sit idle and Asterisk makes a connection to 5060 and tells it there's something going on |
| 04:46.58 | drmessano | This is basic IP networking 101 |
| 04:47.18 | MajesticFudgie | So 5060 is required for SIP to work and can be ignored? |
| 04:47.29 | drmessano | Correct |
| 04:48.24 | MajesticFudgie | Okie |
| 04:48.51 | MajesticFudgie | Well I don't think I can do much more, Firewall is locked down. Secured the web gui on the phone. |
| 04:48.57 | MajesticFudgie | So it should be fine now |
| 04:49.11 | drmessano | Maybe |
| 04:49.15 | drmessano | But I have my doubts |
| 04:49.24 | drmessano | I mean, this can only occur certain ways |
| 04:49.43 | MajesticFudgie | Well what else can I do? |
| 04:49.56 | MajesticFudgie | The firewall is set to deny all incoming on all ports and only allow outgoing |
| 04:49.58 | drmessano | And everything has been an exercise in confirming you had things set up in a certain way, which you do.. |
| 04:50.06 | drmessano | and already did |
| 04:50.14 | drmessano | So you still have no idea what it was |
| 04:50.17 | drmessano | Just what it wasnt |
| 04:50.17 | MajesticFudgie | mhm |
| 04:50.43 | MajesticFudgie | The phone doesnt even show a log of the phone number since I restarted the phone. Nor has it rang since restart |
| 04:51.07 | drmessano | Something on your firewall was open to 5060 on that phone |
| 04:52.32 | drmessano | Thats the troubling part |
| 04:52.44 | MajesticFudgie | 5060 was open but pointed to a dead ip |
| 04:52.50 | drmessano | Thats why I asked about having a connection to another peer somewhere |
| 04:52.57 | MajesticFudgie | Which shouldn't allow it anywhere but that dead ip |
| 04:53.19 | drmessano | Ok that takes care of 5060 |
| 04:53.24 | drmessano | On the firewall |
| 04:53.34 | drmessano | But again |
| 04:53.37 | drmessano | Something on your firewall was open to 5060 on that phone |
| 04:54.04 | MajesticFudgie | Well, no idea what was as only one IP had 5060 open and it /was/ asterisk but the ip changed |
| 04:54.58 | drmessano | On the firewall |
| 04:55.05 | MajesticFudgie | Yes |
| 04:55.20 | drmessano | Thats not entirely important |
| 04:55.48 | MajesticFudgie | Well other than that its closed down. But I've even removed that so nothings open at all |
| 04:56.02 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-138.ph.ph.cox.net) |
| 04:56.02 | drmessano | Well |
| 04:56.08 | drmessano | Nothing is "Forwarded" |
| 04:56.15 | drmessano | There are ports OPEN all day long |
| 04:57.01 | drmessano | I just dont see how a random port was open to 5060 on a phone |
| 04:57.21 | MajesticFudgie | Same |
| 04:58.09 | drmessano | The phone registers to your private IP address? |
| 04:58.17 | drmessano | It wasnt hairpin'ing to the external IP? |
| 04:58.35 | MajesticFudgie | It should't be able to |
| 04:58.44 | MajesticFudgie | its config has it set to just connect to asterisk |
| 04:59.20 | drmessano | ..... |
| 04:59.29 | drmessano | I asked a very specific question |
| 04:59.43 | drmessano | By "ASTERISK" you mean the "Private IP" |
| 04:59.47 | drmessano | Right? |
| 04:59.50 | MajesticFudgie | yes |
| 04:59.55 | drmessano | I asked for steak or fish and you told me potato |
| 04:59.56 | drmessano | Ok |
| 05:00.05 | MajesticFudgie | Well you asked if its "hairpin'ing" |
| 05:00.06 | drmessano | I dunno then |
| 05:00.08 | MajesticFudgie | never heard the word |
| 05:00.34 | drmessano | Phone connecting your Asterisk box using the external IP which routes back in through the NAT |
| 05:00.52 | MajesticFudgie | Shouldn't be possible |
| 05:00.53 | drmessano | Which would hit all the firewall rules |
| 05:01.12 | drmessano | Its not about being possible.. it about what you have configure |
| 05:01.13 | MajesticFudgie | The phone was pointed to 192.168.2.204 which is currently a dead ip |
| 05:01.14 | drmessano | Its not about being possible.. it about what you have configured |
| 05:01.17 | drmessano | Ok |
| 05:01.45 | MajesticFudgie | But even then the call skipped Asterisk, when I answered I got silence |
| 05:01.50 | MajesticFudgie | As creepy as that is |
| 05:02.03 | drmessano | No, ive seen this a lot |
| 05:03.48 | MajesticFudgie | Oh well, guess I'll just have to wait for it to ever happen again and hopefully get a log from the phone |
| 05:04.10 | drmessano | There wont be really anything to see |
| 05:04.22 | drmessano | But |
| 05:04.26 | drmessano | If it does happen again |
| 05:04.37 | MajesticFudgie | Well it may be helpful in finding where it came from and such |
| 05:04.45 | drmessano | Well no |
| 05:04.51 | MajesticFudgie | As right now I'm going off what I saw on the screen and what happened |
| 05:04.51 | drmessano | because the phone wont have any idea |
| 05:05.00 | MajesticFudgie | The phone may have logged attempts though? |
| 05:05.03 | drmessano | "A call hit my listen port.. RING RING" |
| 05:06.24 | drmessano | What kind of router is this? |
| 05:08.11 | MajesticFudgie | BT Home Hub 5 |
| 05:10.02 | tuxd00d | Speaking of routers⦠have you had an issue with an Ubiquity router assigning the same NAT ports to phones? I have one location that has a dozen phones, but in Asterisk they show up as 8 unique ports. But their phones work with Vonage without issue. |
| 05:11.00 | drmessano | MajesticFudgie: HA |
| 05:11.02 | drmessano | Yeah |
| 05:11.04 | drmessano | Well |
| 05:11.24 | drmessano | Im 95% certain we had someone with this same issue |
| 05:11.29 | MajesticFudgie | inb4 router calling me to tell me to pay my bill on time |
| 05:11.32 | drmessano | Due to lack of port randomization |
| 05:12.39 | drmessano | But different in that it was a single phone and indeed it was regged to a hosted Asterisk box |
| 05:12.57 | drmessano | and THAT router used 5060 for the pinhole in the NAT |
| 05:13.29 | tuxd00d | I have not idea how to fix it. Iâm still learning the EdgeMax command line. |
| 05:14.13 | MajesticFudgie | Wierd |
| 05:14.28 | MajesticFudgie | I'll just unplug the phone before I get some sleep and deal with it later |
| 05:15.21 | tuxd00d | I was going to go back and try using TLS. Perhaps it doesnât like UDP. |
| 05:15.35 | drmessano | MajesticFudgie: You can try one thing |
| 05:15.51 | MajesticFudgie | Stabbing it through the heart with a wooden stake? |
| 05:15.54 | tuxd00d | MajesticFudgie: I missed the issue you are having with your phone. |
| 05:16.03 | tuxd00d | MajesticFudgie: Always a sure way to fix it. |
| 05:16.07 | drmessano | Change the LISTEN port, which is usually just the "Port" setting on the line, to something other than 5060 |
| 05:16.11 | drmessano | Make it 5070 |
| 05:16.15 | MajesticFudgie | In the phone? |
| 05:16.18 | drmessano | Yep |
| 05:16.24 | tuxd00d | I had one location where I had to set each phone to a different listen port |
| 05:16.37 | drmessano | Just move it off of 5060 |
| 05:16.49 | drmessano | 5070 is a common choice |
| 05:16.53 | tuxd00d | That as for a PFSense |
| 05:17.18 | tuxd00d | s/as/was |
| 05:17.59 | MajesticFudgie | I'll see if I can change it |
| 05:18.04 | drmessano | Oh you can |
| 05:18.09 | drmessano | It's in the GUI |
| 05:18.23 | tuxd00d | Which phone? |
| 05:19.09 | MajesticFudgie | It's a Polycom Soundpoint 331 iirc |
| 05:19.25 | drmessano | There's a shiny box for Port |
| 05:19.30 | MajesticFudgie | ah I think I sorted it |
| 05:19.45 | MajesticFudgie | Networking -> NAT -> Signalling port |
| 05:19.55 | drmessano | Cool |
| 05:20.00 | MajesticFudgie | changed it to 5070, nmap doesnt see 5060 anymore on the phone |
| 05:20.19 | drmessano | So that may be a good enough forever band-aid |
| 05:20.38 | MajesticFudgie | yup 5070 is now open instead |
| 05:21.38 | tuxd00d | You can use sngrep to watch the conversation |
| 05:21.46 | *** join/#asterisk s-mutin (~s-mutin@85.234.114.134) |
| 05:22.17 | MajesticFudgie | ah |
| 05:22.21 | MajesticFudgie | I'll look into it tomorrow |
| 05:22.26 | MajesticFudgie | I think for now it should be sorted |
| 05:23.57 | drmessano | Right |
| 05:24.57 | drmessano | Besides, there's nothing to watch |
| 05:24.58 | MajesticFudgie | Anyway I need to get some sleep. Thanks for helping me out guys :) |
| 05:25.03 | drmessano | NP |
| 05:25.05 | drmessano | good luck |
| 05:25.12 | tuxd00d | Good luck |
| 05:26.26 | MajesticFudgie | Ty |
| 05:26.29 | tuxd00d | drmessano: do you know if there is a way to show NAT mappings on the Ubiquity EdgeMax? |
| 05:26.53 | drmessano | No, I literally avoid Ubiquiti gear |
| 05:27.48 | *** join/#asterisk puzzola (~puzzola@unaffiliated/puzzola) |
| 05:28.27 | tuxd00d | Everybody seems to love a their own certain brand⦠and avoid the others. Back in the day, PFSenses was the all the rage. Now itâs the plague. |
| 05:28.50 | tuxd00d | What do you like to use? |
| 05:28.51 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 05:28.57 | drmessano | Mikrotik |
| 05:29.05 | drmessano | Ubiquiti quality has taken a slide |
| 05:29.08 | tuxd00d | Did I ask you that before? |
| 05:29.14 | drmessano | No clue |
| 05:29.37 | tuxd00d | Or are you the second one to suggest Mikrotik? I hear good things about them. |
| 05:30.06 | drmessano | I have Ubiquiti APs at work, which I bought before I discovered MT a few years ago |
| 05:30.12 | drmessano | Not only are they awful |
| 05:30.23 | drmessano | But I am sad now that I don't have MT stuff there |
| 05:30.31 | drmessano | But all my side work is Mikrotik stuff |
| 05:30.39 | drmessano | Routers and APs |
| 05:30.43 | tuxd00d | UBNT APs donât work well? |
| 05:31.09 | drmessano | The controller is terrible, the firmware has been russian roulette with updates |
| 05:31.15 | tuxd00d | I have a location where the Cisco APs donât work well, and I was thinking of using UBNT⦠but now⦠|
| 05:31.48 | drmessano | I wouldnt even touch a UBNT router.. People keep discovering major bugs that they havent fixed but known about for years |
| 05:31.54 | drmessano | Like this UDP thing I keep hearing about |
| 05:32.15 | drmessano | and |
| 05:32.52 | drmessano | For NOW I am using a WISP for connectivity to a few of my sites.. THEY have UBNT gear.. and even they are pulling their hair out over radios dying all over the place |
| 05:33.02 | drmessano | So apparently they are having issues |
| 05:33.09 | drmessano | and aint nobody got time for that |
| 05:33.34 | tuxd00d | But MT has been good for you? How is availability in US? |
| 05:33.40 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 05:33.51 | drmessano | Availability is great |
| 05:34.00 | drmessano | They have so many distributors |
| 05:34.07 | drmessano | the best distributors |
| 05:34.10 | drmessano | Bigly ones |
| 05:34.14 | drmessano | But really |
| 05:34.21 | drmessano | Availability is not a problem |
| 05:34.35 | drmessano | I know a few guys that buy units by the hundreds.. no issues |
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| 05:35.23 | drmessano | Their product line has evolved MASSIVELY in the last few years, as well |
| 05:35.28 | drmessano | Like nothing I have seen |
| 05:35.43 | drmessano | They're becoming a major player here now |
| 05:35.50 | drmessano | and already are across the world |
| 05:37.23 | detha | drmessano: fwiw, I feel the same, but with the brands swapped around. Tik has been breaking firmware badly lately, SNMP MIBs suddenly changing, funnies with fastpath, etc |
| 05:37.25 | tuxd00d | Pricing looks really attractive |
| 05:38.36 | drmessano | detha: They've been making MASSIVE changes lately too |
| 05:38.50 | detha | True. Too many changes, too fast |
| 05:38.54 | drmessano | Like development has been in overdrive for at least a year |
| 05:39.04 | drmessano | There's a lot of work to be done |
| 05:39.14 | drmessano | Like the new bridging logic |
| 05:39.27 | detha | in maybe two or three years it might stabilize, but for now I stay away from it |
| 05:39.40 | drmessano | Well whatever |
| 05:39.47 | drmessano | I dont see it as unstable at all |
| 05:39.59 | tuxd00d | So ⦠the consensus is⦠all brands have their faults? |
| 05:40.06 | drmessano | I can deal with minor regressions |
| 05:40.10 | drmessano | What I cant deal with |
| 05:40.17 | drmessano | Is installing new controller software |
| 05:40.20 | drmessano | and new firmware |
| 05:40.27 | drmessano | and performance going in the shitter |
| 05:40.37 | detha | Either brand will work, if you get a working config, and leave it alone. Hardware quality is about the same |
| 05:40.44 | drmessano | SO I have to get on the forums and find a "safe" version for my hardware |
| 05:41.19 | drmessano | UAP AC Pro was the worst $800 I have ever spent... and that was including my 2nd divorce |
| 05:41.21 | *** join/#asterisk bl3nto (~bl3nto@dh207-89-143.xnet.hr) |
| 05:41.47 | tuxd00d | I just need one that can handle VoIP traffic well. VLAN, QoS, PoE, etc.. |
| 05:42.03 | drmessano | MT is fine for that |
| 05:42.12 | tuxd00d | My first divorce cost more than a house in attorneys fees. |
| 05:42.29 | drmessano | Something as simple as an RB750Gr3 can run a medium sized office |
| 05:42.35 | drmessano | and not even break a sweat |
| 05:43.09 | drmessano | Hell I have an RB750Gr3 at work for just network monitoring |
| 05:43.15 | drmessano | Can run The Dude on it |
| 05:43.35 | tuxd00d | For most locations, they want new hardware⦠they are using running 10/100 non-qos dinosaurs. So new router, switch and APs. |
| 05:43.53 | tuxd00d | What is The Dude.. I was just noticing that in their product listing. |
| 05:43.54 | drmessano | Well |
| 05:44.05 | drmessano | It's a network monitoring app |
| 05:44.15 | drmessano | But well integrated into ROS too |
| 05:44.17 | drmessano | For example |
| 05:44.25 | drmessano | You can put a MT at a branch office |
| 05:44.38 | drmessano | and it becomes a relay for a central Dude Server |
| 05:44.55 | drmessano | So it actually acts as the monitor and reports back to the central Dude |
| 05:45.32 | drmessano | But to be clear |
| 05:45.44 | drmessano | It's not OpenNMS or Zabbix or something |
| 05:46.17 | drmessano | It's basically built for monitoring your network devices |
| 05:46.24 | drmessano | APs, Switches, routers |
| 05:46.46 | drmessano | But hell it does have some nice SNMP capabilities |
| 05:47.05 | drmessano | and will monitor common services like NetBIOS ports, FTP, HTTP, etc |
| 05:47.34 | drmessano | So I monitor 50 or so workstations, all my switches, routers, APs, and some other appliances |
| 05:47.54 | drmessano | I've not taken a deep dive into the SNMP capabilities, but it's all there |
| 05:48.09 | tuxd00d | Awesome, I was looking for a product line that I can centrally manage. |
| 05:48.25 | drmessano | So my suggestion |
| 05:48.34 | drmessano | Get an RB750Gr# |
| 05:48.35 | drmessano | Get an RB750Gr3 |
| 05:48.41 | drmessano | Get a HAP Lite AC |
| 05:48.48 | drmessano | Thats basically a MT lab in a box |
| 05:48.53 | drmessano | For $100 |
| 05:48.58 | drmessano | Later on |
| 05:49.16 | drmessano | You can upgrade the RB750Gr3 to a 3011 for The Dude |
| 05:49.36 | drmessano | or even the big day $300 2011 Dude Edition |
| 05:49.40 | drmessano | or even the big daddy $300 2011 Dude Edition |
| 05:50.05 | drmessano | the WAP ACs are nice for APs |
| 05:50.18 | drmessano | They have something called CAPSMAN for central AP configuration |
| 05:50.28 | tuxd00d | Iâm having trouble locating the RB750GR3 on their site. |
| 05:50.32 | drmessano | So you can manage all of a sites APs from one place |
| 05:50.58 | drmessano | https://mikrotik.com/product/RB750Gr3 |
| 05:51.01 | drmessano | Its the Hex |
| 05:51.12 | tuxd00d | Ohh, awesome. |
| 05:51.44 | drmessano | Look at the specs on that little bastard |
| 05:51.44 | tuxd00d | Most locations have about 100 devices, 40-50 of which are phones. |
| 05:52.00 | drmessano | RB750 should be perfect for that |
| 05:52.23 | drmessano | If you look at the specs |
| 05:52.26 | drmessano | It's a small box |
| 05:52.32 | drmessano | But packs a LOT of punch |
| 05:52.43 | tuxd00d | Thatâs what I hear. |
| 05:52.43 | drmessano | As in |
| 05:52.48 | drmessano | Look at the other boxes |
| 05:52.51 | drmessano | Like the HAP AC |
| 05:53.02 | drmessano | Which IS a wireless router |
| 05:53.05 | drmessano | and quite nice |
| 05:53.18 | drmessano | But the specs as far as being a "router", not so much |
| 05:53.31 | drmessano | It's a good branch or home device though... I have several |
| 05:53.33 | tuxd00d | I donât like the APâs built into the router. Most of the time, the router is located in a closet away from the office. |
| 05:53.51 | drmessano | Ok, so the RB750Gr3 is gonna be your best bet |
| 05:53.56 | tuxd00d | And a jack of all trades is a master of none. |
| 05:54.31 | tuxd00d | And I can centrally monitor the RB750GR3, or do I need a larger unit? |
| 05:54.41 | drmessano | You can |
| 05:54.58 | drmessano | All of the boxes have a full ROS L4 license |
| 05:55.02 | drmessano | So they do it all out of the box |
| 05:55.22 | drmessano | From the $20 box to the couple thousand dollar boxes |
| 05:55.27 | drmessano | Same ROS |
| 05:55.29 | drmessano | Same license |
| 05:56.11 | drmessano | If you want a pretty rack unit for your central monitoring, I would upgrade to a 3011 at some point |
| 05:56.18 | drmessano | But like I said |
| 05:56.30 | drmessano | I run an RB750 for all my network monitoring |
| 05:57.13 | drmessano | I have 100 or so endpoints in there now |
| 05:57.13 | drmessano | and it doesn't crack 10% CPU |
| 05:58.21 | tuxd00d | A really appreciate the help drmessano. |
| 06:01.39 | *** join/#asterisk Cory (~Cory@unaffiliated/cory) |
| 06:12.49 | tuxd00d | Have any of you used a Peplink Max Cellular router? |
| 06:14.13 | tuxd00d | Theyâre suppose to be good with VoIP traffic. |
| 06:14.36 | tuxd00d | At least they advertise as such. |
| 06:15.11 | drmessano | So I dont want to sound like I own stock in the company |
| 06:15.13 | drmessano | But |
| 06:15.21 | drmessano | First, do you need the Wifi part? |
| 06:15.34 | drmessano | I googled for the box |
| 06:15.49 | drmessano | https://3gstore.com/product/5618_pepwave_max_br1_3g_4g.html?gclid=EAIaIQobChMIqPec2KqD1gIVGlYNCh3mWQQiEAYYASABEgIz8PD_BwE |
| 06:15.52 | drmessano | Is that it? |
| 06:16.12 | tuxd00d | Nope, just the SD-WAN with a cellular link part |
| 06:16.38 | tuxd00d | Yep, that is one version of the MAX line. |
| 06:16.41 | drmessano | Well back to the stock in the company part |
| 06:16.57 | drmessano | RB750Gr3 + a USB Aircard |
| 06:17.01 | drmessano | Works great lol |
| 06:17.13 | tuxd00d | Does it do SD-WAN? |
| 06:18.19 | drmessano | I've only heard of SD-WAN as a broader term.. What is the specific need? |
| 06:18.42 | tuxd00d | I was originally going to do the USB air card route⦠but the Peplink looks really nice as it can route only VoIP traffic over the SD-WANâ¦. for âunbreakableâ VoIP .. aka âNo dropped callsâ |
| 06:19.05 | tuxd00d | Keeping the cellular costs down. |
| 06:20.05 | drmessano | Are you saying it alleges handoff across network changes? |
| 06:20.05 | tuxd00d | It keeps a connection open on both the broadband and the cellular, and sends packets over cellular when the broadband is gone or lagging. |
| 06:20.55 | tuxd00d | You have to have an aggregator on the other end, which can simple be a AWS EC2, for example. |
| 06:21.05 | drmessano | I cant even begin to see how that works with VoIP. The path is going to change. |
| 06:21.26 | tuxd00d | To Asterisk the IP is always the same. |
| 06:21.48 | tuxd00d | Kind of like a VPN. |
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| 07:22.55 | [sID] | It is possible for my own connection status to be issued? |
| 07:23.21 | [sID] | eg: 501 "max limit" |
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| 07:42.01 | tuxd00d | [sID]: Are you asking if you can set the SIP response code? |
| 07:48.18 | [sID] | yes |
| 07:48.38 | tuxd00d | core show application Congestion |
| 07:49.22 | tuxd00d | I think that is as close as you can get with Asterisk. Kamailio would typically be used to set custom response codes. |
| 07:51.21 | *** join/#asterisk sekil (~sekil@cable-89-216-223-237.dynamic.sbb.rs) |
| 07:54.48 | [sID] | ok thx |
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| 12:41.07 | DanQuinney | dan_j: initial observations seem to point our issue with dropped packets to a set of MySQL backups running on the * servers |
| 12:43.29 | dan_j | DanQuinney: ok, sounds like a different issue to mine then |
| 12:43.52 | dan_j | i'm not dropping packets, it just taking asterisk 10 seconds to respond to REGISTERs and INVITEs every so often |
| 12:43.57 | dan_j | on multiple servers |
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| 13:00.36 | dan_j | thanks for the update though |
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| 13:56.50 | Samot | 8:43:48 AM <dan_j> i'm not dropping packets, it just taking asterisk 10 seconds to respond to REGISTERs and INVITEs every so often <-- If they're not dropping packets and taking that long to respond, what is the load like on those systems? |
| 13:58.35 | Samot | And when you say 10 seconds to respond, do you mean it takes 10 seconds from the moment the device is up for the REGISTER to hit Asterisk? Or are you saying once Asterisk gets the REGISTER it takes 10 seconds for the 401 challenge to be issued? |
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| 14:09.55 | DanQuinney | Samot: if I can jump in? We're seeing similar to dan_j |
| 14:10.04 | DanQuinney | https://usercontent.irccloud-cdn.com/file/TL2l7gYE/options.pcap |
| 14:10.14 | Samot | OK |
| 14:10.19 | Samot | So the questions apply to you |
| 14:10.54 | Samot | Where is the delay? |
| 14:10.56 | DanQuinney | yes, * receives the OPTIONS request, then 40 seconds later sends the 200 OK |
| 14:11.14 | Samot | And what is the load like? |
| 14:11.20 | dan_j | 10 seconds from the moment the packet hits the server for asterisk to process it and output a reply. |
| 14:11.38 | dan_j | Low load |
| 14:11.44 | DanQuinney | minimal load Samot |
| 14:11.51 | dan_j | Friday is always quieter but its still continuing. |
| 14:11.52 | Samot | "low", "minimal" |
| 14:11.56 | Samot | These are vague. |
| 14:12.00 | DanQuinney | let me get the graphs |
| 14:12.01 | Samot | What is the load like? |
| 14:13.08 | dan_j | gtjoseph: has suggested i do a coredump during an outage to see if anything is locked up. I need to wait till the weekend to recompile and perform a dump. Problem is performing a core dump will result in a longer outage. |
| 14:13.17 | DanQuinney | 0.10 - 0.43 |
| 14:13.52 | Samot | On how many cores? |
| 14:14.12 | dan_j | Samot: only 20 channels were open at the last outage. Ive supported over 100 with the same config. And its happening on all my asterisk boxes. |
| 14:14.17 | dan_j | 4 core |
| 14:14.20 | Samot | OK |
| 14:14.22 | Samot | Guys. |
| 14:14.30 | dan_j | 20 to 30% cpu |
| 14:14.39 | Samot | I'm going to give you some serious professional advice. |
| 14:14.45 | Samot | Get off PJSIP. |
| 14:14.51 | Samot | It's not prime time for providers. |
| 14:15.30 | dan_j | I thought about that but its not practical to switch back to chansip as that doesnt support binding to multiple specific ip addresses while leaving others out. |
| 14:15.44 | Samot | OK. |
| 14:15.59 | Samot | If that is really, really, really a show stopper for you.... |
| 14:16.08 | Samot | Then "somewhat stable" is your status. |
| 14:16.29 | Samot | Your customers can sit through you making issue reports and hammering out PJSIP issues with Digium. |
| 14:17.08 | dan_j | But i get where you are coming from. Its just annoying that ignoring the 'qualify' issues, its been very stable. This is a new issue that appeared and i'd like to understand it rather than just working around it. |
| 14:18.13 | dan_j | And there is no guarantee that chansip will resolve this. |
| 14:20.33 | Samot | *cough*me*cough* |
| 14:20.39 | Samot | I AM running chan_SIP |
| 14:23.27 | Samot | Between my network and another ITSP's network I manage, I've got over 100+ Asterisk/FreePBX servers going. |
| 14:23.32 | Samot | All running Chan_SIP |
| 14:23.45 | Samot | I am not seeing these issues. |
| 14:24.51 | dan_j | I understand that you arent having issues but if my issue is a network one then chansip may not help |
| 14:25.16 | Samot | They are hitting the servers |
| 14:25.27 | Samot | Once they are on the server, the server takes X seconds to process request |
| 14:25.43 | dan_j | You are going under the assumption that its a pjsip bug. It could be a dns issue or something |
| 14:25.43 | Samot | This is like driving to the corner store |
| 14:25.51 | Samot | Round trip is should take 10 minutes |
| 14:26.00 | Samot | But once you are in the store and there is a huge line... |
| 14:26.03 | Samot | It takes 20 minutes |
| 14:26.11 | Samot | The traffic part was fast |
| 14:26.16 | Samot | The processing part was slooooow |
| 14:26.35 | Samot | Are you see the servers have any other slow responses? |
| 14:26.37 | Samot | STMP? |
| 14:26.41 | Samot | SMTP |
| 14:26.44 | Samot | MySQL? |
| 14:26.56 | Samot | Anything else on the server running with delays? |
| 14:27.01 | Samot | Or just Asterisk/SIP? |
| 14:27.13 | *** join/#asterisk markit (~marco@88-149-177-66.v4.ngi.it) |
| 14:28.03 | dan_j | Just asterisk. All the other servers in that datacentre seem to be running normally. However that could be that they aren't as time sensitive. |
| 14:31.33 | Samot | I don't mean other servers. |
| 14:31.41 | Samot | I mean other services on the Asterisk box. |
| 14:31.48 | Samot | It's sending emails I'm sure |
| 14:31.57 | Samot | And doing other things for stuff running. |
| 14:32.00 | Samot | Are those having issues?! |
| 14:32.12 | Samot | Do you see a delay with SMTP communications? |
| 14:32.40 | Samot | Do you see a delay in performance any where else on that server outside of the Asterisk service? |
| 14:33.37 | Samot | When Asterisk has its periodic fits, does the rest of the system take a piss as well? |
| 14:35.40 | dan_j | I wouldnt notice a 10 second smtp delay |
| 14:35.52 | Samot | Have you looked? |
| 14:36.02 | Samot | You're having delays on the system.. |
| 14:36.08 | dan_j | No other issues showing in the logs. Pacemaker is able to run sipsak without a problem |
| 14:36.09 | Samot | Have you confirmed they are not system wide? |
| 14:36.29 | dan_j | System wide on multiple machines would be highly unusual. |
| 14:37.10 | Samot | So yeah, you're right.. |
| 14:37.12 | Samot | It could be DNS |
| 14:37.15 | Samot | It could be the network |
| 14:37.22 | Samot | But so far everything that's been done.. |
| 14:37.26 | Samot | Doesn't point to those. |
| 14:37.31 | dan_j | Yep |
| 14:37.34 | Samot | It points to an issue with Asterisk. |
| 14:37.39 | Samot | And most likely PJSIP. |
| 14:37.59 | Samot | Even file said yesterday PJSIP hasn't been exposed to certain environments that much |
| 14:38.05 | *** join/#asterisk stefan27 (~stefan27@static-212-247-4-149.cust.tele2.se) |
| 14:38.08 | Samot | So yeah, you could be discovering issues that no one else would have. |
| 14:39.43 | dan_j | Ok. I surrender. I'll move back to chansip if I can. |
| 14:40.57 | Samot | Well.. |
| 14:41.06 | Samot | I'm not trying to beat you into submission. |
| 14:41.22 | Samot | I'm just giving the counter arguments |
| 14:49.05 | dan_j | Truth is, if the coredump reveals a bug, it might take months for a fix. |
| 14:49.35 | dan_j | I need this fixing today, so the simplest option for me is chansip as i can do that |
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| 14:58.50 | Samot | Sucks but probably the best option. |
| 14:59.44 | Samot | I'm not like "yay I won" on this.. |
| 14:59.55 | Samot | It sucks, it's going to be a lot of work and headache |
| 14:59.59 | Samot | I feel for ya. |
| 15:01.39 | DanQuinney | managed to get a lock file from when the "pause" happens - I'll open up an issue shortly https://www.irccloud.com/pastebin/gI5COq7H/ |
| 15:02.14 | *** join/#asterisk cervajs2 (~cervajs2@178.148.broadband4.iol.cz) |
| 15:05.24 | cervajs2 | what is good way to pass json i.e. { "sales":"1", "support":"2" } as argument to AGI? if i use AGI(agi://localhost/jsonParse,${web_query}) its not complete in agi_arg1 . howto escape arg? |
| 15:07.57 | hdon | hi all :) is it possible to add custom headers to SIP requests and responses? is there a way to get that data into asterisk's CDR? |
| 15:08.33 | hdon | I understand that I can customize the fields in the CDR and use dialplan code to assign what goes into each field. So I guess my question is actually, is there a way to retrieve SIP headers from dialplan? |
| 15:08.55 | Samot | You can get them from the INVITE |
| 15:09.00 | Samot | But you can't get them after that |
| 15:09.22 | hdon | in the dialplan? |
| 15:09.27 | Samot | Yes. |
| 15:09.36 | hdon | awesome :) how? |
| 15:10.11 | Samot | Seriously? |
| 15:10.14 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_SIP_HEADER |
| 15:10.15 | hdon | is it possible for dialplan to customize SIP headers when sending an INVITE? |
| 15:10.17 | Samot | Come on. |
| 15:10.20 | hdon | ahhhh thanks :) |
| 15:11.37 | reveal | How easy is it to move extensions from 2 digits to 4 digits by adding leading 00's? |
| 15:11.37 | hdon | SipAddHeader might do it |
| 15:11.52 | *** join/#asterisk karelk (~karel@31.10.154.117) |
| 15:11.52 | hdon | reveal, are you using an ordinary configuration file you edit by hand? |
| 15:12.22 | hdon | reveal, would you accept an easy way to edit your configuration file the way you describe? |
| 15:12.35 | Samot | Well are you using Chan_SIP or PJSIP |
| 15:12.55 | Samot | SIPAddHeader won't apply to PJSIP |
| 15:13.01 | reveal | hdon: yes if it can be limited by specific extensions, and can update via console or gui |
| 15:13.02 | hdon | reveal, something like sed -i 's/^\([0-9]{2}\)\>/\100/' might do the trick if that's what you want |
| 15:13.38 | reveal | hdon: this would also affect the users who login with a softphone and would need to be changed to reflect that right |
| 15:13.45 | hdon | Samot, i'm not using PJSIP. good to know this limitation. |
| 15:14.26 | hdon | reveal, sorry i was only thinking of extensions.conf -- for things like spi.conf and voicemail the same precise command won't apply |
| 15:14.55 | hdon | reveal, "affect users who login with a softphone" -- are you talking about provisioning configuration to your phones? my solution doesn't affect that. |
| 15:15.21 | hdon | reveal, do you run a TFTP server that provisions configuration for your phones? |
| 15:15.28 | reveal | well if user 20 was changed to 0020 i am going on a limb they won't be able to login to the softphone |
| 15:15.37 | reveal | xlite |
| 15:16.32 | hdon | Samot, what about custom sip headers for the response to the INVITE? is that possible? |
| 15:18.06 | hdon | reveal, if you change your sip.conf file and change extensions there, peers trying to register with you will have to be changed as well |
| 15:18.25 | reveal | will this also screw with voicemail.conf? |
| 15:18.49 | hdon | reveal, mabye an option for you is to leave those phones on their existing extensions, and simply direct calls to their four digit counterparts to the corresponding two digit extensions? |
| 15:19.09 | hdon | reveal, what's motivating this endeavor? are you just trying to renumber your dialplan? |
| 15:19.19 | Samot | hdon: You cannot do a response. |
| 15:19.25 | hdon | Samot, ok thanks :( |
| 15:19.28 | Samot | hdon: That's not what Asterisk does. |
| 15:19.40 | reveal | currently expanding so we didnt prepare with just 2 digits and didnt realize how big we got |
| 15:19.49 | Samot | If you want to customize SIP messages during the transaction, you need a SIP router. |
| 15:19.50 | reveal | so we thought moving to 4 would give more than enough room for growth |
| 15:20.13 | reveal | is it simeple to keep the 2 digits there and create a forwarder for 4 digits? |
| 15:20.40 | hdon | reveal, okay so maybe you should just add to your dialplan something like, e.g. for extension 11, add to your dialplan: exten => 1100 s,1,GoTo(whatever) and redirect to extension 11 |
| 15:21.05 | hdon | Samot, hmm... okay thanks |
| 15:22.45 | Samot | You can set SIPHeaders for the INVITE of an outbound call |
| 15:22.55 | reveal | hdon: thanks |
| 15:23.04 | Samot | And you can get the SIPHeaders from the INVITE of an inbound call |
| 15:23.09 | Samot | That's it. |
| 15:23.38 | Samot | You can't see the response, you can see new messages in the transaction... |
| 15:23.41 | hdon | reveal, yes that's what i'm describing now. just forward these 4 digit extensions to your 2 digit extensions. actually you could do it with only one line i believe. exten => _XX00,1,Goto(${CONTEXT},${EXTEN,0,2},1) ; IIUC |
| 15:23.41 | Samot | er can't. |
| 15:23.59 | hdon | Samot, does my dialplan recommend for reveal look good? |
| 15:24.50 | Samot | Yeah |
| 15:24.57 | Samot | No. |
| 15:25.01 | hdon | :c |
| 15:25.03 | Samot | Wait.. |
| 15:25.08 | Samot | Those should be : |
| 15:25.13 | hdon | ohh right |
| 15:25.18 | Samot | If you're trying to slice the extension. |
| 15:25.29 | hdon | reveal, exten => _XX00,1,Goto(${CONTEXT},${EXTEN:0:2},1) |
| 15:25.37 | hdon | Samot, yeah i was :) |
| 15:26.02 | reveal | I see this effectively will answer if someone dialed 22 or 0022 |
| 15:26.49 | hdon | reveal, sorry i did it for 2200 not 0022. you'll need to make a small adjustment |
| 15:27.08 | hdon | reveal, but yeah that should do it. just make sure you put this in the right context. |
| 15:27.14 | reveal | yeah |
| 15:27.15 | Samot | I'll never get the desire for leading zero extensions. |
| 15:27.25 | reveal | me either |
| 15:27.29 | hdon | reveal, or put it in its own context and include it from any context that needs this rule |
| 15:27.37 | reveal | thats a great idea |
| 15:28.04 | hdon | :) |
| 15:39.41 | hdon | when i create a bridge with ARI, i can specify an ID for the bridge, right? |
| 15:47.07 | Samot | Dude. |
| 15:47.10 | Samot | Seriously. |
| 15:47.12 | Samot | WIKI |
| 15:47.28 | Samot | You are asking questions that are easily answered by reading the ARI section. |
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| 16:01.57 | hdon | :3 |
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| 16:26.27 | Bhakimi | hi guys! i was wondering what the best way to connect a dialplan that executes chanspy to another dialplan that plays a audio file |
| 16:26.39 | Bhakimi | my goal is to use chanspy to play a specific audio file to a extension |
| 16:27.10 | Bhakimi | so while the user is talking on the phone, i can play a prerecorded message to them on demand |
| 16:27.24 | Bhakimi | is that possible ? |
| 16:28.04 | salviadud | on demand, by some script maybe? |
| 16:28.15 | Bhakimi | yea, i already have the script |
| 16:28.21 | Bhakimi | im jsut not use how to do this |
| 16:28.41 | Bhakimi | i use AMI for this |
| 16:28.45 | Bhakimi | here is a example ami web request: channel=Local/98765555-10-SIP-382@convoso&timeout=10&callerid=8304 <8304>&priority=1&context=100192&exten=8304&action=Originate |
| 16:29.10 | Bhakimi | 98765555-10-SIP-382 is the dialplan, this takes the string and splits it and connects to it via chanspy |
| 16:29.12 | Bhakimi | i know it works |
| 16:29.39 | Bhakimi | but i dont think i can call it like a local channel and connect it to extension 8304 which plays a audio recording |
| 16:30.45 | [TK]D-Fender | ? |
| 16:31.14 | [TK]D-Fender | "connect to extension" is not a clear thing |
| 16:31.28 | [TK]D-Fender | please rephrase your request |
| 16:32.11 | Bhakimi | ok |
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| 16:32.41 | Bhakimi | i mean to connect it to a sip extension thats currently in a call |
| 16:34.54 | Bhakimi | let me try to explain better: so there is a sip extension say 382 that is currently in a call, i like to wispier to them a prerecorded message |
| 16:38.29 | [TK]D-Fender | BRIDGE is what you wnt to do |
| 16:38.41 | Bhakimi | yea |
| 16:38.41 | [TK]D-Fender | you want to BRIDGE to an CHANNEL already in progress |
| 16:38.43 | Bhakimi | how can i do that using ami? |
| 16:38.47 | Bhakimi | correct |
| 16:38.49 | [TK]D-Fender | Bridge <- |
| 16:39.06 | Bhakimi | im using astersik 11 |
| 16:39.18 | [TK]D-Fender | Or actually ... technically chanspy does still dump audio to an existing call |
| 16:39.24 | [TK]D-Fender | both work |
| 16:39.37 | [TK]D-Fender | <Bhakimi> hi guys! i was wondering what the best way to connect a dialplan that executes chanspy to another dialplan that plays a audio file <- this already works |
| 16:39.52 | Bhakimi | i tried but it doesnt =( |
| 16:40.00 | [TK]D-Fender | the idea works. |
| 16:40.04 | Bhakimi | yea i figured |
| 16:40.10 | [TK]D-Fender | maybe your implementation doesn't, but the idea does |
| 16:40.11 | Bhakimi | i think the issue is here: channel=Local/98765555-10-SIP-382@convoso&timeout=10&callerid=8304 <8304>&priority=1&context=100192&exten=8304&action=Originate |
| 16:40.17 | Bhakimi | so this is a ami request |
| 16:40.19 | [TK]D-Fender | and you haven;'t shown us what yuo are actually doing |
| 16:40.31 | [TK]D-Fender | that AMI doesn't prove what the call DOES |
| 16:40.40 | Bhakimi | 98765555-10-SIP-382 is a dialplan and so it 8304 |
| 16:40.52 | Bhakimi | let me give you both dialplans |
| 16:41.52 | Bhakimi | https://pastebin.com/K5XwGcuG |
| 16:42.23 | Bhakimi | i basically want to send the 8304 dialpklan to the chanspy dialplan in order to acomplish this |
| 16:44.15 | Bhakimi | [TK]D-Fender: does that explain better? |
| 16:44.41 | Bhakimi | sorry im the worst at explaining |
| 16:47.52 | [TK]D-Fender | Show us the call |
| 16:48.44 | Bhakimi | how can i show you the call? |
| 16:48.50 | Bhakimi | i dont get what you mean by that? |
| 16:49.16 | [TK]D-Fender | DO IT |
| 16:49.21 | Bhakimi | lol |
| 16:49.25 | [TK]D-Fender | Place the call. Show us your attempt to DO THIS |
| 16:49.29 | Bhakimi | ok |
| 16:49.30 | Bhakimi | 1 sec |
| 16:49.43 | Bhakimi | when i do this i dont see anything in asterislk which is the issue |
| 16:49.52 | Bhakimi | this request above that is |
| 16:49.58 | Bhakimi | but the original sip call i can show, 1 moment |
| 16:50.13 | [TK]D-Fender | If you see nothing then the call isn't hitting the dialplan which means you have an AMI issue |
| 16:51.24 | Bhakimi | https://pastebin.com/6fRAcrUr |
| 16:51.38 | Bhakimi | this is the original call |
| 16:51.50 | [TK]D-Fender | all I see is some AGI |
| 16:51.59 | Bhakimi | the idea is to send the 8304 dialplan which plays a audio file to the sip extension 382 |
| 16:51.59 | [TK]D-Fender | you showed some kind of WEB request before |
| 16:52.10 | [TK]D-Fender | this doesn;'t show may ANYTHING useful that lokos like it's even related |
| 16:52.18 | Bhakimi | do you want the response of that request? |
| 16:52.36 | [TK]D-Fender | I don't see your ORIGINATE getting accepted anywhere |
| 16:52.43 | Bhakimi | ok 1 moment |
| 16:52.49 | [TK]D-Fender | Where do I see the CHANNEL get called? |
| 16:52.57 | [TK]D-Fender | that should create a local channel and thus dialplan execution |
| 16:52.59 | [TK]D-Fender | it's not there |
| 16:56.29 | Bhakimi | channel=Local/98765555-10-SIP-382@convoso |
| 16:56.41 | Bhakimi | there ? |
| 16:57.00 | [TK]D-Fender | Where in taht pastebin are you EXECUTING your request? |
| 16:57.11 | [TK]D-Fender | https://pastebin.com/6fRAcrUr |
| 16:57.25 | [TK]D-Fender | I see no DIALPLAN executing there for the spy code you showed |
| 16:57.29 | Bhakimi | found the issue |
| 16:57.31 | Bhakimi | thank you!! |
| 16:57.36 | Bhakimi | ami was not authenticating |
| 16:57.39 | [TK]D-Fender | And in the diaplan part you did show I don't see the CONETXT name so I can't even prove where it exists |
| 16:59.17 | Bhakimi | @convoso |
| 16:59.21 | Bhakimi | thats the context |
| 16:59.37 | Bhakimi | ohh |
| 16:59.39 | Bhakimi | dialplan |
| 16:59.44 | Bhakimi | that was jsut a snippet |
| 17:00.00 | Bhakimi | but you helped me figure it out, thank you for making me look for the ami response |
| 17:00.01 | Bhakimi | ;) |
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| 21:32.56 | Bhakimi | hi guys!!! is there a different way to play a audio file to a extension while its connected somewhere else besides using chanspy? |
| 21:33.05 | Bhakimi | im worried chanspy may cause high loads if called often |
| 21:33.13 | Bhakimi | or possibly cause asterisk to have issues |
| 21:36.36 | [TK]D-Fender | You fear this because .... ? |
| 21:37.37 | Bhakimi | we have thousands of calls that we will do this with throughout the day maybe 20 a second and adding chanspy to do this with that volume concerns me |
| 21:38.45 | [TK]D-Fender | No other method is going to be almost any different |
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| 21:39.11 | [TK]D-Fender | What is the trigger for you to begin those playbacks? |
| 21:40.18 | Bhakimi | we have a call center platform and we send call to the agents confbridge they sit in, confbridge has a enter and leave sound which we used but when calls get connected they caller we connect into the confbridge sometimes hears this sound and knows its a telemarketer |
| 21:40.57 | Bhakimi | so what i did is have the script that throws these calls into confbridge send the call using a silent confbridge profile and at the same time i use chanspy to notify the agent with a alert so they know a call came in |
| 21:42.15 | Bhakimi | im worried that addtional canspy for every call we connect to the agents can cause load or unexpected asterisk issues which may resutl in a crash/deadlock or other unknown issues |
| 21:45.21 | [TK]D-Fender | You can't go on the theory of "unexpected random nameless issues" |
| 21:45.32 | [TK]D-Fender | Go try and see if your load goes up or things start crashing |
| 21:45.46 | [TK]D-Fender | you have no evidence of failure or reasonable basis, and no provided details. |
| 21:45.48 | [TK]D-Fender | this is FUD |
| 21:46.20 | [TK]D-Fender | Bridge is the only other tool and I don't see it being much different |
| 21:46.28 | [TK]D-Fender | and you could clearly test and compare for yourself |
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| 21:50.36 | Bhakimi | whats FUD ? |
| 21:52.02 | rmudgett | Fear Uncertainty and Doubt |
| 21:52.42 | rmudgett | Usually spread by telemarkerters. :) |
| 21:55.38 | [TK]D-Fender | Or anybody who say "I'm afraid I'll have nameless problems." with no basis |
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| 22:03.23 | Bhakimi | agreed ;) |
| 22:03.46 | Bhakimi | [TK]D-Fender: may i pm you? |
| 22:05.14 | [TK]D-Fender | For? |
| 22:05.49 | Bhakimi | was wondering if i can add you on skype or icq? |
| 22:05.56 | Bhakimi | if you have either |
| 22:10.19 | [TK]D-Fender | no. private contact is only for actual clients |
| 22:13.34 | Bhakimi | how can i become a client? |
| 22:14.03 | Bhakimi | im looking for asterisk support |
| 22:14.30 | [TK]D-Fender | depends what your needs are exactly. |
| 22:14.35 | [TK]D-Fender | This you can PM me for. |
| 22:14.39 | Bhakimi | k thx |
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| 22:54.56 | Bhakimi | hi guys! so my server is a load of 2.0 with 32 cores, and as bandwidth increases int he box meaning we get more calls connected i seem to be getting a crazy amount of messages like these: chan_iax2.c:9983 socket_process_meta: Received trunked frame before first full voice frame |
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| 22:55.19 | Bhakimi | i was wondering if anybody can give me some direction? i searched google but wasnt able to find anything that can help me fiure out the cause |
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| 22:55.45 | Bhakimi | i wonder if the server is hitting some kind of load limit? |
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