The Goal
I'm trying to make a call to a telephone number. I'd like to be able to make a call from the raspberry pi, and also make a call to my voip.ms phone number and be able to answer or auto-answer and play some generic .wav file.
My current understanding of things
This maybe should be titled "My current misunderstanding of things". I'm new to sip and pjsip, and I think I must be missing some part of the process I don't understand. I was under the impression that, if I register with voip.ms, when I make a call it would route to voip.ms and they would do a lookup on the number/address, and then respond with an address that I would then begin to communicate with.
What I've done so far
Compiled
I've compiled Pjsip on a Raspberry Pi 3B+ properly, for what I can tell. I can include pjsua2.hpp in my c++ applications. I've roughly followed this tutorial
Tested compilation with pjsua binary && demo.cpp
I'm running into identical problems running a modified pjsua2_demo.cpp and the binary included in the pjsip build. For the sake of simplicity, I'll ask about the binary located (for me) at <project-path>/pjproject-2.8/pjsip-apps/bin/pjsua-armv7l-unknown-linux-gnueabihf.
Successfully registered with voip.ms
I have an account and phone number with Voip.ms and can become registered with voip.ms by executing the following script:
call_and_auto_answer.sh
./pjsua2-cpp/pjproject-2.8/pjsip-apps/bin/pjsua-armv7l-unknown-linux-gnueabihf \
--play-file ~/CantinaBand60.wav \
--local-port=5060 \
--auto-answer 200 \
--auto-play \
--auto-loop \
--max-calls 5 \
--config-file ./sip.cfg
Where the config looks like:
sip.cfg
#
# Logging options:
#
--log-level 5
--app-log-level 4
#
# Account 0:
#
--id sip:<my-subaccount-username>#sip.voip.ms
--registrar sip:<server-location>.voip.ms
--reg-timeout 300
--realm *
--username <my-subaccount-username>
--password <my-subaccount-password>
--use-timer 1
#
# Network settings:
#
--local-port 5060
#
# Media settings:
#
--srtp-keying 0
--auto-play
--auto-loop
--play-file /home/pi/CantinaBand60.wav
--snd-auto-close 1
#using default --clock-rate 16000
#using default --quality 8
#using default --ec-tail 200
#using default --ilbc-mode 30
--rtp-port 4000
#
# User agent:
#
--auto-answer 200
--max-calls 5
#
# SIP extensions:
#
--use-timer 1
When I enter the cli, I see for my account list:
Account list:
[ 0] <sip:192.168.1.49:5060>: does not register
Online status: Online
[ 1] <sip:192.168.1.49:5060;transport=TCP>: does not register
Online status: Online
*[ 2] sip:<my-subaccount-username>#sip.voip.ms: 200/OK (expires=285)
Online status: Online
Buddy list:
-none-
Voip.ms shows I've registered on their website.
The problem
I'm trying to call my personal cell phone from my pi (I assume using the registered voip.ms phone number), and call my pi from my personal cell phone. While calling out I'm typically getting either 408 Request Timeout errors or 502 gethostbyname errors.
Different destinations, different errors
Depending on the destination for my call from the pi, I get one of two different errors most of the time
Timeout Error
I get an error that says,
18:19:19.757 pjsua_app.c ....Call 4 is DISCONNECTED [reason=408 (Request Timeout)]
18:19:19.757 pjsua_app_comm ....
[DISCONNCTD] To: <destination-sip-address>
where is any of the following:
sip:
sip:
sip:thetestcall#sip2sip.info
sip:thetestcall#iptel.org
sip:201#ideasip.com
and the phone numbers are formatted like: 3035551234, though I've tried prepending a 1 and a +1 just to check.
Lookup Error
I get an error that says,
19:09:45.435 sip_resolve.c ....Failed to resolve '<destination-sip-address>'. Err=70018 (gethostbyname() has returned error (PJ_ERESOLVE))
19:09:45.435 tsx0x18520dc ....Failed to send Request msg INVITE/cseq=10722 (tdta0x185012c)! err=70018 (gethostbyname() has returned error (PJ_ERESOLVE))
19:09:45.435 pjsua_app.c .......Call 4 is DISCONNECTED [reason=502 (gethostbyname() has returned error (PJ_ERESOLVE))]
19:09:45.435 pjsua_app_comm .......
[DISCONNCTD] To: sip:<destination-sip-address>
where is any of the following:
sip:
sip:
sip:abcd1234
Possible Successes
I'm getting what looks like a success while calling:
sip:**12340#ideasip.com
It confirms the call and has a bunch of messages, notable including:
19:16:17.550 pjsua_core.c ....TX 1300 bytes Request msg INVITE/cseq=13899 (tdta0x15c263c) to UDP 208.97.25.11:5060:
...
19:16:17.551 pjsua_app.c .......Call 4 state changed to CALLING
...
>>> 19:16:17.606 pjsua_core.c .RX 575 bytes Response msg 100/INVITE/cseq=13899 (rdata0x6d7008a4) from UDP 208.97.25.11:5060:
...
19:16:17.609 pjsua_core.c .RX 946 bytes Response msg 200/INVITE/cseq=13899 (rdata0x6d7008a4) from UDP 208.97.25.11:5060:
...
19:16:17.609 pjsua_app.c .....Call 4 state changed to CONNECTING
...
19:16:17.610 pjsua_app.c .....Call 4 state changed to CONFIRMED
...
19:16:17.676 pjsua_core.c .RX 594 bytes Response msg 100/INVITE/cseq=13900 (rdata0x6d7008a4) from UDP 208.97.25.11:5060:
...
19:16:17.678 conference.c ......Port 5 (sip:**12340#ideasip.com) transmitting to port 5 (sip:**12340#ideasip.com)
...
19:16:17.678 conference.c ......Port 1 (/home/pi/CantinaBand60.wav) transmitting to port 5 (sip:**12340#ideasip.com)
...
19:16:36.931 pjsua_app.c ......Call 4 is DISCONNECTED [reason=200 (Normal call clearing)]
Same Network
Additionally, if I set up a second pjsip client on the same network, I can call it from pi1 and answer the call on pi2.
Incoming calls
When I register with voip.ms, then try to call my voip.ms phone number from my personal cell phone, the call fails with a message on my iPhone that says, User Busy. This makes me think I'm messing something up with the registration, or that I'm missing some component, like a subscribe or link with that voip.ms account.
Final thoughts
I'm not sure what I'm missing here. I've read through a ton of the pjsip and pjsua docs, and I can't find anything I'm missing. Does anybody have insight into how to make a call to a phone number and allow for incoming calls? This has been quite a few days of solid work.
So I figured out the answer to my question. Here's the skinny:
Voip.ms registration
My registration with voip.ms wasn't configured properly. I was given credentials by a coworker, but upon further inspection of the sip endpoint, I found that the DiD number purchased for the account wasn't associated with the subaccount my coworker created for me. So, depending on the recipient's phone carrier, I was given different errors. Additionally, when I was testing inbound calls and receiving the error, User Busy, this was because the account I registered wasn't associated with the phone number.
To fix this, on voip.ms I associated the DiD number to my subaccount, and then went to my subaccount information and set the callerId number to be my DiD number, though I think you can override this value via pjsip.
Outbound sip calls
Secondly, to call a phone number, outbound calls should follow the following format:
sip:<phoneNumber>#<endpoint>
So for me, this looked very much like:
sip:5551234567#newyork.voip.ms
Related
I am using Fiddler 5.0 to capture https requests of mobile app. It works fine except one app. Below is logs of Fiddler.
21:31:59:7072 HTTPSLint> Warning: ClientHello record was 508 bytes long. Some servers have problems with ClientHello's greater than 255 bytes. https://github.com/ssllabs/research/wiki/Long-Handshake-Intolerance
21:31:59:7072 !SecureClientPipeDirect failed: System.Security.Authentication.AuthenticationException A call to SSPI failed, see inner exception. < 处理证书时，出现了一个未知错误。 for pipe (CN=ci.migudm.cn, O=DO_NOT_TRUST, OU=Created by http://www.fiddler2.com).
21:31:59:7412 !SecureClientPipeDirect failed: System.Security.Authentication.AuthenticationException A call to SSPI failed, see inner exception. < 处理证书时，出现了一个未知错误。 for pipe (CN=ci.migudm.cn, O=DO_NOT_TRUST, OU=Created by http://www.fiddler2.com).
21:31:59:7862 HTTPSLint> Warning: ClientHello record was 508 bytes long. Some servers have problems with ClientHello's greater than 255 bytes. https://github.com/ssllabs/research/wiki/Long-Handshake-Intolerance
Have you checked out the link provided by the logs? There's a bunch of very relevant information there that I think might help you get what you need:
Some web servers are unable to process Client Hello messages longer than 255 bytes, even though such messages are allowed by the protocol. Although most Client Hello messages are shorter than this limit, with the use of Server Name Indications (SNI), Elliptic Curve extensions, and the increase in the number of supported suites, the size can easily go over.
Browsers are thought not to be affected, but some other tools may be. In particular, OpenSSL 1.0.1+ is known to be affected. If you are using the s_client tool, try to reduce the size of the handshake by disabling some of the suites offered by OpenSSL (this can be done by specifying the desired suites using the -cipher switch).
OpenSSL Bug #2771: Openssl 1.0.1 times out when connecting to Outlook Exchange 2007; sign in with guest/guest.
F5 tracks this problem under bug #376483 and it is fixed in BIG-IP LTM 10.2.4 (see [SOL14758: SSL client connections may fail if the ClientHello message is greater than 255 bytes and uses TLS 1.1 or 1.2] (https://support.f5.com/kb/en-us/solutions/public/14000/700/sol14758.html) for more details).
Michael Tschannen says:
The issue is fixed according to F5 in 10.2.4, they are however not mentioned in any release notes (I have >just checked until 11.x)
To debug the issue on an F5, the log level of "SSH" needs to be at least "Informational".
Example:
Mar 22 09:44:21 local/tmm info tmm[4696]: 01260013:6: SSL Handshake failed for TCP from x.x.x.x:443 to x.x.x.x:49549
Source: https://github.com/ssllabs/research/wiki/Long-Handshake-Intolerance
If you've been through all of this information, could you please provide the F5 version, OpenSSL version, and log information from OpenSSL?
i have a Ubuntu 16.04.5 server with Vesta CP.
I checked the server on https://dnsflagday.net, and I got this report:
domain.cl. #123.456.78.90 (ns1.domain.cl.): dns=ok edns=ok edns1=ok edns#512=ok ednsopt=ok edns1opt=ok do=ok ednsflags=ok docookie=ok edns512tcp=timeout optlist=ok
domain.cl. #123.456.78.90 (ns2.domain.cl.): dns=ok edns=ok edns1=ok edns#512=ok ednsopt=ok edns1opt=ok do=ok ednsflags=ok docookie=ok edns512tcp=timeout optlist=ok
I do not know what edns512tcp = timeout means and I have not had much luck looking for a solution on internet.
Can someone help me? thanks
For that tool, any kind of "timeout" error is a problem, it means some server did not reply or the message (either query or reply) was eaten by some active element on the path, so it needs to be fixed.
edns512tcp is when the testing software does an EDNS query with a buffer of 512 bytes and over TCP.
If you go to https://ednscomp.isc.org/ednscomp/ for your domain you will have the full test results.
For that specific error it is:
EDNS - over TCP Response (edns#512tcp)
dig +vc +nocookie +norec +noad +edns +dnssec +bufsize=512 dnskey zone #server
expect: NOERROR
expect: OPT record with version set to 0
See RFC5966 and See RFC6891
So you can see which DNS query was done with dig, that you can reproduce it (+vc is an old flag name that is an alias for +tcp). The test expects to get a NOERROR code back and an OPT record. Your servers did not reply at all, so the test failed.
It seems that your servers did not reply to that at all, which is wrong. Maybe they do not reply to TCP queries at all which is even more wrong. In all cases you will need to contact the entity responsible for maintaining those servers and point it to the test results so that they start to fix the problem.
thanks for your help.
I read more about it and I could detect that port 53 was being blocked by the firewall, I added the rule to the firewall to allow TCP connections on port 53.
Everything it's fine now
I have the Telit LE920-EUG 4G LTE module. I am trying to execute GET and POST http requests to a remote server. Though the PDP context is activating properly and I have internet access on the SIM that I'm using, I can't seem to be able to connect to a remote server and execute HTTP requests (both POST and GET) from the module.
I have tried two ways, one through direct HTTP commands supported by the module(All commands mentioned in the LE9x0 AT command reference guide), the commands sequence for which is mentioned below, but +CME ERROR: 100 occurs, and it's same for every http command(AT#HHTPQRY, AT#HTTPRCV) that I try to execute.
AT#SGACT=1,1
#SGACT: 31.81.208.1
OK
AT#HTTPCFG=0,"httpbin.org",80,0,,,0,120,1
+CME ERROR: 100
//No configuration details
AT#HTTPCFG?
+CME ERROR: 100
AT#HTTPCFG=?
+CME ERROR: 100
I have also tried the GET and POST commands after socket dialing. The socket connects but they are not receiving any data from the server or posting anything onto the server, the connection closes with a NO CARRIER. The command sequence that I'm using is given below
//Socket Dial
AT#SD=1,0,80,www.m2msupport.net
CONNECT
//GET commands sequence
GET /m2msupport/http_get_test.php HTTP/1.1
Host:www.m2msupport.net
Connection:keep-alive
//Connection closes with No Response
NO CARRIER
//Socket info shows the bytes sent
at#si=1
#SI: 1,86,0,0,0
OK
I am using pjsua as command line SIP client and it works just fine and is capable to auto answer call automatically but i would like to somehow use auto answer delay. Configuration I have works fine but pjsua client answer the call immediately. The thing i need is some delay before call is answered with 200 OK.
Any help would be welcome
My config file contains:
# dont use hosts audio device
--null-audio
# Logging options:
--log-level=3
--app-log-level=3
--log-file=LOG-File.log
# Account 0:
--id=sip:100#test-lab.org
# SIP Signal IP addr
--registrar=sip:test-lab.org
--reg-timeout=3600
--proxy=sip:10.20.21.254
--realm=test-lab.org
--username=100#test-lab.org
--password=12345678
--use-timer=1
--use-ims
--auto-update-nat=1
# Network settings:
--local-port=5065
--ip-addr=10.100.100.25
--bound-addr=10.100.100.25
--no-tcp
--outbound=sip:10.20.21.254;lr
# Media settings:
--ptime=20
--rtp-port=8000
#
--snd-auto-close=1
# User agent:
--max-calls 6
--add-codec=pcma
--clock-rate=8000
--auto-play
--duration=300
--play-file=music.wav
--auto-answer=200
--no-vad
It seems that mod_logxml module is not compatible with ejabberd 13.12 version. Ejabberd fails to start when this module is installed.
My question is - is there another way how to log (file, database, etc) all the XMPP packets send and received by ejabberd?
Yes, you can make your own log module if you cannot find anything that fits your needs.
You can use a module I developed as a reference. This module which would intercept stanza with the type "chat" or "groupchat" and send a acknowledgement back to the sender.
You can modify this module by removing the type so it intercepts all stanza types and log whatever you want into a DB instead of sending a message back to the sender.
https://github.com/Mingism/ejabberd-stanza-ack
I've changed mod_logxml.erl that it will work with eJabberd 13.12. You have to put in
ejabberd/src/
Configs are for ejabberd.yml format:
mod_logxml:
stanza: [message, other]
direction: [external]
orientation: [send, recv]
logdir: "/var/jabber/logs/"
timezone: universal
rotate_days: 1
rotate_megs: 100
rotate_kpackets: no
check_rotate_kpackets: 1
Repository