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| 10:16.32 | MaryJohanna | Hi, I have been working with asterisk a few years ago, but my knowledge about it has been vanished over the time. This time I am wondering if I could use asterisk as a "proxy" or middleware between a SIP server access I am paying for and me. I am looking for a documentation where these flags are described to setup a proxy. |
| 10:17.15 | MaryJohanna | basically right now I can connect to this SIP server, but I want to setup a voicemail thing and to create waiting music and whatsoever on client side which is what I would use asterisk for |
| 10:17.49 | MaryJohanna | so if I would receive a call it would come from the 3rd party SIP server -> my asterisk server -> Zoiper on my phone |
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| 10:48.18 | MaryJohanna | Okay, I figured. What is the bindaddr directive in the sip.conf? |
| 10:48.26 | *** join/#asterisk gavlee (~gav@unaffiliated/gavlee) |
| 10:48.38 | MaryJohanna | there I can define the IP used to reach the sip server I am defining? |
| 11:15.05 | Ravenheart | i had to deal with NAT and asterisk recently |
| 11:15.09 | Ravenheart | its not pleasant |
| 11:15.27 | Ravenheart | ended up running a VPN between asterisk and the SIP phone |
| 11:15.56 | Ravenheart | and using sip.conf (CHAN SIP) is discouraged, its deprecated and you should move to use PJSIP |
| 11:20.33 | MaryJohanna | I am not the one running the SIP provider, so the chance for me to run a VPN there is impossible |
| 11:21.07 | MaryJohanna | anyway thanks for the input |
| 11:21.19 | MaryJohanna | I will take a look how easy it is to hook in pjsip to asterisk |
| 11:22.26 | Ravenheart | pjsip has a million sections and endpoints to configure :) |
| 11:23.10 | MaryJohanna | the interesting part is that I don't even have a pjsip.conf file at /etc/asterisk |
| 11:24.11 | MaryJohanna | nvm I didn't have it installed |
| 11:24.12 | Ravenheart | i built from source and the included example files had it |
| 11:38.57 | *** join/#asterisk MillerBoss (~shit@your.mom.loves.millerboss.com) |
| 11:40.39 | MaryJohanna | Okay, I set it up so far. But how do I specify an IP it should use to connect to the server? |
| 11:40.49 | MaryJohanna | I have multiple interfaces |
| 11:43.30 | Ravenheart | https://hatebin.com/ncqyhzzmwb |
| 11:44.27 | MaryJohanna | Yeah I have a slightly modified version of a DT one I found online atm: https://telekomhilft.telekom.de/t5/Telefonie-Internet/Asterisk-pjsip-conf-fuer-tel-t-online-de-hinter-NAT/td-p/3125047 |
| 11:44.44 | MaryJohanna | but I don't see where this would help me to use a specific IP for the exit traffic |
| 11:45.47 | *** part/#asterisk MillerBoss (~shit@your.mom.loves.millerboss.com) |
| 11:46.11 | MaryJohanna | I have a feeling that the software doesn't support that out of the box so I might as well just set the default gw for the asterisk user to the one interface I need for the connection |
| 11:46.33 | MaryJohanna | because the sip server I am connecting to has my IP whitelisted on that interface and exit IP, but its not my default one |
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| 12:43.31 | TandyUK | MaryJohanna: I think you should look at kamailio |
| 12:43.58 | MaryJohanna | in what way it is better? |
| 12:44.10 | TandyUK | well it sounds like you want to proxy SIP |
| 12:44.21 | charrit69 | Hi, in pjsip. Is there a way to add a header without a predial handler? |
| 12:44.22 | TandyUK | which is precisely what kamailio was invented for |
| 12:44.57 | charrit69 | in chan_sip you could call SIPAddHeader prior to channel establishment, then if a SIP channel was established... it adds the header |
| 12:45.19 | TandyUK | you could also setup a vps somewhere near your providers server (whether using asterisk or anything), and VPN your phone to that |
| 12:46.06 | TandyUK | we notice major issues with packet loss over certain 3/4g providers in the UK, which a vpn (and relevant codecs, ie opus) tend to handle a lot better |
| 12:47.21 | TandyUK | the vpn also means you can quite safely run your SIP over unecrypted UDP (its travelling ove a vpn anyway..), but still have the leg from your server to your provider using TLS/etc |
| 12:48.12 | TandyUK | just note that statement becomes invalid if its not YOUR vpn (ie you dont control both ends, like if you used something like nordvpn) |
| 12:54.55 | TandyUK | ah sorry missed your second line about wanting to add voicemail etc, that you would need asterisk for |
| 12:55.22 | TandyUK | or you could just use a provider who offers the features you want? |
| 12:55.36 | MaryJohanna | sadly I can't |
| 12:55.47 | MaryJohanna | its a work provided service |
| 12:55.53 | TandyUK | ah gotcha |
| 12:55.53 | MaryJohanna | I just want to make it smarter |
| 12:56.09 | MaryJohanna | until now I used it always with Zoiper android |
| 12:56.15 | MaryJohanna | it worked well, but lacked many features |
| 12:56.31 | MaryJohanna | that's why I am thinking about making a middleware asterisk |
| 13:10.51 | MaryJohanna | Okay, I set it up. Now while trying to start it, I get: Loading res_pjsip.so. |
| 13:10.55 | MaryJohanna | that hangs forever |
| 13:11.10 | MaryJohanna | I have to SIGKILL it, otherwise it hangs forever, not even Ctrl+C works |
| 13:11.15 | *** join/#asterisk spatel (~spatel@pool-96-237-230-175.bstnma.fios.verizon.net) |
| 13:18.22 | MaryJohanna | Oh, seemingly it hangs forever waiting for the DNS resolving |
| 13:25.33 | charrit69 | MaryJohanna: yes, it happens from v16, if you don't have internet access or you cannot resolve dns it get's stuck. In v.18 I never had this problem |
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| 13:26.24 | charrit69 | In v18 happens too, I wanted to write that in v1.8 never happens |
| 13:27.04 | charrit69 | If you wait a long time it boots but take too much time |
| 13:27.06 | MaryJohanna | well I am on the latest available in the fedora repo: Asterisk 17.4.0 |
| 13:27.40 | charrit69 | I always compile it from source |
| 13:27.50 | MaryJohanna | I might will do that to |
| 13:28.49 | Samot | Fedora doesn't have 16? |
| 13:28.58 | Samot | 17 is SFO. |
| 13:28.58 | MaryJohanna | At the moment I am trying to figure out why it cannot connect to the sip server. Seemingly the routes are correct now |
| 13:29.14 | MaryJohanna | I just installed it and did asterisk -V |
| 13:29.18 | MaryJohanna | no clue what SFO is |
| 13:29.28 | Samot | Security Fixes Only. |
| 13:29.30 | MaryJohanna | oh |
| 13:29.43 | Samot | 17 will no longer receive bug fixes/updates. It's a dead version. |
| 13:29.53 | Samot | 16 and 18 are the Long Term Support. versions. |
| 13:29.58 | MaryJohanna | yeah well I am on fedora server 32 |
| 13:30.03 | MaryJohanna | that's what comes from the repos |
| 13:30.09 | Samot | Do they have 16 in the repos? |
| 13:30.25 | Samot | Because that's what you should be on. |
| 13:30.26 | Ravenheart | ubuntu 20 had v16 |
| 13:30.27 | MaryJohanna | let me check |
| 13:30.50 | Ravenheart | but it was an old version, so i build from source |
| 13:31.27 | MaryJohanna | https://pastebin.com/raw/89Hu5wVA nope |
| 13:31.44 | MaryJohanna | that's on 32 Server Edition |
| 13:31.48 | charrit69 | v16 is a LTS, you can use it without problem |
| 13:33.58 | MaryJohanna | even the latest release of fedora, the 33 one is on 17 lol. Maybe the packager got confused about the versions: https://fedora.pkgs.org/33/fedora-x86_64/asterisk-17.5.0-2.fc33.3.x86_64.rpm.html |
| 13:34.15 | MaryJohanna | f33 was released not long ago |
| 13:35.29 | Samot | Well 17 isn't a current version. |
| 13:35.30 | MaryJohanna | anyway I am trying to register myself still to the sip server using pjsip. I enabled logging and I see this header: "Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=<somelonghash>" |
| 13:35.35 | MaryJohanna | any way to change that? |
| 13:35.52 | Samot | Why is there a local host IP in there? |
| 13:37.06 | MaryJohanna | I assume its inherited from my global config |
| 13:37.08 | MaryJohanna | let me paste it in |
| 13:38.18 | MaryJohanna | https://pastebin.com/raw/1WjGeiXq |
| 13:39.15 | Samot | OK, let me ask that question again. |
| 13:39.20 | Samot | Why is there a localhost IP? |
| 13:39.53 | MaryJohanna | I have no clue why its there. And in the config, I couldn't think of any better address |
| 13:40.10 | Samot | The actual routable address of the box? |
| 13:40.21 | Samot | Localhost is not a routable address on the network. |
| 13:41.01 | MaryJohanna | then I am not sure what that bind directive is for |
| 13:41.12 | Samot | For the IP bound to Asterisk. |
| 13:41.20 | MaryJohanna | If I would change it to 0.0.0.0, would that work? |
| 13:41.23 | Samot | Doesn't this system have an IP from your network? |
| 13:41.42 | MaryJohanna | yeah but the IP is dynamic |
| 13:41.50 | Samot | You have it set to a dynamic IP? |
| 13:41.53 | Samot | No.\ |
| 13:41.56 | MaryJohanna | nope |
| 13:41.59 | Samot | Give it a static IP. |
| 13:42.01 | MaryJohanna | I have a dynamic dns |
| 13:42.09 | Samot | OK |
| 13:42.11 | MaryJohanna | Wait now I am confused |
| 13:42.21 | Samot | Does it have an IP from your LAN?! |
| 13:42.27 | MaryJohanna | so the issue is with that line? bind = 127.0.0.1:5060 right? |
| 13:42.38 | Samot | Well part of it. |
| 13:42.43 | MaryJohanna | that's where the Via header gets populated |
| 13:42.44 | MaryJohanna | ok |
| 13:42.55 | Samot | Does the PBX box have an IP from your LAN? |
| 13:43.07 | Samot | If you're LAN is 10.0.6.x does it have an IP from that? |
| 13:43.15 | MaryJohanna | No, its on WAN, but let me explain the whole setup |
| 13:43.25 | Samot | It's not on WAN |
| 13:44.36 | MaryJohanna | I have SIP server I am connecting to. It likes to call itself ims. That's my company's server and the one I am trying to connect to using pjsip. But while I can reach the server from WAN, there is a whitelist so I have to be on a company IP for it to work |
| 13:44.46 | MaryJohanna | So I created a VPN at work, specially openvpn |
| 13:45.01 | MaryJohanna | that's where the 10.6.0.6 comes from |
| 13:45.34 | MaryJohanna | that's the IP of the server I have my very own asterisk server set up rn and 10.6.0.6 is its IP on that interface |
| 13:45.39 | Samot | So this is your companies Asterisk server? |
| 13:45.57 | MaryJohanna | I have no clue what do they use whether its asterisk or not |
| 13:46.22 | Samot | OK so you want the PBX to have an IP of the VPN range? |
| 13:46.29 | MaryJohanna | indeed |
| 13:46.42 | MaryJohanna | but I also want to be able to access the PBX outside of the VPN range as well |
| 13:46.46 | Samot | So that IP is bound on one of the interfaces of the system? |
| 13:47.08 | MaryJohanna | lets say that I actually want to bind it to 0.0.0.0/0 |
| 13:47.13 | MaryJohanna | and then let the firewall do the rest |
| 13:47.34 | MaryJohanna | sorry if I wasn't clear the first time |
| 13:47.53 | charrit69 | bind = 0.0.0.0:5060 |
| 13:47.59 | Samot | OK so set the bind to ^^^ |
| 13:48.10 | Samot | Then change the local network to 10.6.0.0/24 |
| 13:48.23 | charrit69 | and remove domain and external_* or put it to the right addresses |
| 13:48.26 | Samot | Because the local network is for devices/phones that will connect to the PBX locally. |
| 13:49.00 | MaryJohanna | alright |
| 13:52.03 | MaryJohanna | Okay, it definitely did something new |
| 13:52.21 | MaryJohanna | How do I confirm that it indeed connected now? |
| 13:52.30 | MaryJohanna | <PROTECTED> |
| 13:52.59 | Samot | Show your configs |
| 13:53.50 | MaryJohanna | I am literally using this with some minor modifications to the company's domain: https://telekomhilft.telekom.de/t5/Telefonie-Internet/Asterisk-pjsip-conf-fuer-tel-t-online-de-hinter-NAT/td-p/3125047 |
| 13:53.58 | MaryJohanna | as this config seems to use outbound proxies too |
| 13:54.40 | MaryJohanna | but I can paste in mine too with some masking if necessary. Though its literally the same now except the external_* things, the global fields (what you guys told me to change) and the account details |
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| 13:59.03 | Samot | OK so the work PBX is setup for this? |
| 13:59.10 | MaryJohanna | yeah |
| 13:59.22 | MaryJohanna | oh it looks like my bad |
| 13:59.22 | MaryJohanna | Could not find option suitable for category 'global' named 'language' at line 1410 of |
| 14:03.13 | *** join/#asterisk sa02irc (~mbax@155-079-043-212.ip-addr.inexio.net) |
| 14:03.24 | Samot | So the work PBX is setup to accept your Asterisk box as another PBX connected to it? |
| 14:10.19 | MaryJohanna | Oh well it isn't I assume. I can connect to it using Zoiper however and the provided address and outbound proxy address + user credentials. Therefore I assumed I can get it to work with asterisk too |
| 14:10.40 | MaryJohanna | However now I am getting back "SIP/2.0 400 Bad Request" |
| 14:10.45 | Samot | From where? |
| 14:10.50 | Samot | Getting it back from where? |
| 14:11.05 | MaryJohanna | this is the response my server gets |
| 14:11.13 | MaryJohanna | when I launch asterisk and it attempts to connect |
| 14:11.17 | Samot | OK so they don't like your request. |
| 14:12.16 | Samot | MaryJohanna: Why do you even need Asterisk? |
| 14:18.51 | *** join/#asterisk bford (uid283514@gateway/web/irccloud.com/x-hwfkslzleoxlcjgu) |
| 14:18.52 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:42.00 | MaryJohanna | Samot: I got sent home into home office and lost all the ability my lovely cisco phone could do for me, such as voicemail, waiting music, blacklists etc |
| 14:42.33 | MaryJohanna | I wanted asterisk to be a middleware that runs on my server 7/24 and I can connect to it instead of using Zoiper to connect to work directly |
| 14:43.30 | Samot | The cisco phone doesn't do waiting music, blacklists, voicemail. |
| 14:43.40 | Samot | The *phone system* does that. The phone is just a device on it. |
| 14:43.40 | MaryJohanna | well it did |
| 14:43.45 | MaryJohanna | it also had a scripting layer |
| 14:43.50 | Samot | OK |
| 14:44.03 | MaryJohanna | it connected the same way to the server |
| 14:44.09 | Samot | Well having Asterisk isn't going to solve that. |
| 14:44.18 | Samot | Those are PBX level features you are referring to. |
| 14:44.24 | Samot | They will not pass to Asterisk like you think. |
| 14:44.39 | Samot | Your voicemail account is *not on the phone* |
| 14:45.13 | MaryJohanna | Well, I had a PBX system at home a long time ago. It worked the same way. I could play a waiting music, direct calls from here and there etc |
| 14:45.25 | Samot | Yeah when it's the ONLY pbX |
| 14:45.27 | MaryJohanna | that's exactly what I want to achieve and I know its possible somehow |
| 14:45.33 | Samot | They sent you home to use a softphone that connects to their PBX |
| 14:45.47 | Samot | So you can access your voicemail, etc. |
| 14:45.55 | Samot | Because that's on the office PBX. |
| 14:45.57 | MaryJohanna | I would like to have pjsip set up so if I have an incoming call, asterisk gets it in the extension and does whatever I programmed it to |
| 14:46.07 | MaryJohanna | like ringing a number on my own pbx |
| 14:46.14 | MaryJohanna | meanwhile playing a waiting music |
| 14:46.20 | Samot | Well work knows your putting a PBX on your work extension? |
| 14:46.22 | MaryJohanna | and if necessary, record what they say |
| 14:46.27 | MaryJohanna | yeah they do |
| 14:46.35 | Samot | Doesn't your work PBX already do that stuff? |
| 14:46.36 | MaryJohanna | I asked the admin and he said good luck |
| 14:46.39 | Samot | Music on hold... |
| 14:46.41 | MaryJohanna | they dont |
| 14:46.51 | MaryJohanna | its a pretty poor system |
| 14:46.59 | MaryJohanna | I have no idea what software do they use even |
| 14:47.13 | MaryJohanna | I just got credientals and an openvpn to a whitelisted exit IP |
| 14:47.16 | Samot | Well if they are using Cisco phones that all off those working features, a Cisco system. |
| 14:47.17 | MaryJohanna | and a good luck message |
| 14:47.26 | Samot | Since Cisco phones are designed for Cisco systems. |
| 14:47.36 | MaryJohanna | could be |
| 14:48.02 | MaryJohanna | I just don't understand why do I get a 400 now with asterisk |
| 14:48.07 | Samot | Well you having issues connecting to your work PBX is an thing you and the work PBX admin need to deal with. |
| 14:48.30 | MaryJohanna | also this: Warning: <ipwashere> "someid[somenumber] Register request having user info" |
| 14:48.45 | MaryJohanna | oh and a 399 before the ip |
| 14:49.06 | Samot | Show it, don't talk about it. |
| 14:49.17 | MaryJohanna | that's all |
| 14:49.27 | MaryJohanna | I just masked the someid and somenumber |
| 14:49.34 | MaryJohanna | as I am not sure if they are private ids |
| 14:51.17 | Samot | Well you're going to need to show actual attempts so the full picture can be seen. |
| 14:51.36 | MaryJohanna | okay, let me paste in the register request and response |
| 14:51.40 | Samot | Showing one line of redacted output and giving commentary on it isn't helpful. |
| 14:53.55 | MaryJohanna | https://pastebin.com/raw/6F53JYP0 |
| 14:55.42 | MaryJohanna | I will just see what zoiper does first and if I can see any difference in the registration process |
| 14:56.15 | file | they probably don't want username in the server_uri for the outbound registration |
| 14:56.54 | Samot | Also why do you have a +mynumber |
| 14:57.01 | Samot | Wouldn't that be your extension number? |
| 14:57.13 | MaryJohanna | well, that's the username they gave to me |
| 14:57.23 | MaryJohanna | its a real world number |
| 14:57.38 | MaryJohanna | so + and the country code. In my case +36 |
| 14:59.09 | MaryJohanna | indeed that was it file, thanks |
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| 15:41.27 | MaryJohanna | Okay, it works now perfectly. However right now if I try to connect to my asterisk server using my softphone, it always keeps trying to authenticate using my company's PBX. I don't want that. I want the sip connection to my company as a simple softphone connection. And I want my PBX with my own numbers and rules. |
| 15:41.34 | MaryJohanna | Is that even possible with asterisk? |
| 15:42.09 | MaryJohanna | sadly most tutorials and docs are still written for the old sip.conf & user.conf duo and I can't seem to find much docs on the pjsip one |
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| 16:01.55 | Samot | Did you change the settings in the softphone? |
| 16:20.11 | MaryJohanna | yes |
| 16:20.43 | MaryJohanna | I see it now in my asterisk server log that everytime my softphone tries to connect my asterisk server tries to forward the register request to the work pbx |
| 16:20.47 | MaryJohanna | which sends back a 401 |
| 16:21.04 | MaryJohanna | I don't want that, I want my own numbers I have configured to work as well |
| 16:21.32 | MaryJohanna | I created a bob user based on the tutorial to see if it works and logged in using that |
| 16:22.00 | Samot | It's forwarding the request? |
| 16:22.08 | Samot | You have some really misconfigured. |
| 16:22.17 | Samot | something* |
| 16:22.54 | MaryJohanna | yeah it does |
| 16:23.18 | Samot | Show it. |
| 16:23.23 | Samot | Show logs of this happening. |
| 16:27.41 | MaryJohanna | https://pastebin.com/raw/chNF5cmL |
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| 16:31.06 | DannyA | hey all. is there a synchronous programmatic way to read from astdb? |
| 16:31.16 | DannyA | when using AMI, i hate that i have to wait for a separate dbresposne callback |
| 16:35.35 | Samot | MaryJohanna: That doesn't show me anything. It shows Zoiper making a register request and YOUR Asterisk box challenging it. That's normal. |
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| 17:27.34 | darkdrgn2k | any one happen to know where to find 3111-40000-001.bootrom.ld for apolycom 7000 ip? |
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