SIP connection through Proxy

[arnauos]

Hi, my internet service provider requires the user of a Proxy Server (10.31.255.134:5070) in order to connect to the landline SIP account (connection goes through a different vlan than regular internet traffic).

dic 16 15:53:19 FX1 callblockerd[1049]: INFO:   starting callblockerd 0.11.1
dic 16 15:53:19 FX1 callblockerd[1049]: error: XDG_RUNTIME_DIR not set in the environment.
dic 16 15:53:19 FX1 callblockerd[1049]: getaddrinfo: Success
dic 16 15:53:19 FX1 callblockerd[1049]: resolving hostname telefonica.net to IPv4 address failed

Is there a way to make this work with callblocker?

Thanks

[pamapa]

using directly the IP address and not the URL might help

[arnauos]

Did some more tests.

Setting the proxy address and port on the domain field returns:

dic 16 20:11:45 FX1 callblockerd[27886]: INFO:   starting callblockerd 0.11.1
dic 16 20:11:45 FX1 callblockerd[27886]: error: XDG_RUNTIME_DIR not set in the environment.
dic 16 20:11:45 FX1 callblockerd[27886]: getaddrinfo: Success
dic 16 20:11:45 FX1 callblockerd[27886]: resolving hostname 10.31.255.134:5070 to IPv4 address failed

Setting only the IP address returns a 408 when registering (timeout?)

dic 16 20:12:24 FX1 callblockerd[28087]: INFO:   starting callblockerd 0.11.1
dic 16 20:12:24 FX1 callblockerd[28087]: error: XDG_RUNTIME_DIR not set in the environment.
dic 16 20:12:56 FX1 callblockerd[28087]: registration on domain '10.31.255.134' with username 'XXXXXXXXX' failed (code=408)

Adding the "@telefonica.net" after the username, returns the following error:

dic 16 20:12:03 FX1 callblockerd[27976]: INFO:   starting callblockerd 0.11.1
dic 16 20:12:04 FX1 callblockerd[27976]: error: XDG_RUNTIME_DIR not set in the environment.
dic 16 20:12:04 FX1 callblockerd[27976]: pjsua_acc_add() failed (Invalid URI (PJSIP_EINVALIDURI))

[pamapa]

hard to say what the problem is!

you can increase the log level with "log_level" to "debug" as described here https://github.com/pamapa/callblocker/tree/master/etc/callblocker to see more. In case of SIP "pjsip_log_level" to "5".

[pamapa]

ok, i looked into the code, for SIP the pjsip library is used and no proxy support used yet.

In "SipAccount SipAccount::add" can you add something additionally like this?
cfg.reg_use_proxy=3;
// cfg.require_100rel=1; // not sure its needed
cfg.proxy_cnt=1;
cfg.proxy[0]=pj_str("sip:;lr"); // outbound_proxy

I have no proxy to try out. Maybe this site helps you:
https://wiki.asterisk.org/wiki/display/AST/PJSIP+with+Proxies

[arnauos]

Good news, I have added the lines you said after this one:

cfg.cred_info[0].data = pj_str((char*)m_settings.password.c_str());
Manually specifying the proxy address (as per notation I saw in documentation):

  cfg.reg_use_proxy=3;
  // cfg.require_100rel=1; // not sure its needed
  cfg.proxy_cnt=1;
  cfg.proxy[0]=pj_str("sip:10.31.255.134:5070\;lr"); // outbound_proxy

I also created a hosts entry file for my provider domain (telefonica.net) pointing to the proxy address to bypass the resolve error.

Now callblocker registers succesfully and I can see on the log the incoming calls. I can set them to block but I can't see callblocker hanging up the call.

Will do some more testing tomorrow.

Thanks! :D

[arnauos]

Ok, did some more testing.

• Receiving "registration on domain 'telefonica.net' with username 'XXXXXXXXX' failed (code=408)" on the logs but it does register
• Calls are being correctly logged by the app
• Calls are identified when present on the blacklist but they are not always blocked (from 10 test, got one time blocked only)
• When blocking, call is cut on the caller but the rest of the phones continue ringing (landline phones connected to a CISCO SPA-112 ATA) -> how is the call blocked? if it's with an answer+hangup the rest of the phones should stop ringing...

[pamapa]

The call is actually blocked by a simply answer+hangup.

i think we are getting closer to make it work.
Can you try with un-commenting "cfg.require_100rel=1;"?:

cfg.reg_use_proxy=3;
cfg.require_100rel=1; // this might help
cfg.proxy_cnt=1;
cfg.proxy[0]=pj_str("sip:10.31.255.134:5070\;lr"); // outbound_proxy

[arnauos]

I observe the same behaviour with require_100rel added.

Today it seems to be cutting calls successfully, but landline phones still continue ringing. Just guessing, maybe the answer+hangup is too fast for the ATA to acknowledge that the call has been taken elsewhere?

[pamapa]

yes, i was also think about "too fast". Can you try adding a sleep? either after detecting its a spam or just before the hangup.

[pamapa]

also remember in case of analog phones the caller information (CID) is transmitted between the first and second ring. so one ring is always expected for analog phones.

[arnauos]

Yeah, one or two rings does not matter, but call ending is a must.

Will try adding the sleep and do some more testing in few weeks as I'll have some free time.

I'll report back then. Thanks!

[pamapa]

what is also very interesting is what happens in the function "void SipAccount::onCallState(pjsua_call_id call_id, pjsip_event* e)".

what is logged in the cases where the hangup was not done?

• Logger::debug("[%s] call state changed to %s", number.c_str(), state.c_str());
  Does "Logger::debug("hangup...");" occur?

[pamapa]

what is the state here, does proxy work for you?

[arnauos]

Sorry, one thing led to another and in the end, I totally forgot 😱

I will continue the tests in a few days and report back.

[arnauos]

Hi, I did a new install and some more testing, I think the issues are down to my sip provider as I can't get consistent results during testing.

I also had a sip client open and overall, the same behaviour was observed.

• connection through proxy = seems to work after defining the hosts entry for the domain and adding the outbound_proxy in settings ("sip:x.x.x.x:yyyy;lr"), this is mostly due to my providers setup
• however, I still get some registration failed code=408 messages
• call blocking works correctly sometimes
• tried adding sleep before the call block, also changing pjsua_call_hangup to pjsua_call_answer with diferent status codes (486, 600, 603...) with the same results observed, sometimes blocked, sometimes the ATA and the SIP client keep ringing

So, overall, I think the issue might be that sometimes the client can't register (code=408) and call blocking will not work correctly until it succesfully connects.

Not sure on how to fix the code=408 error.