is there a discussion list for openrcs? I couldn't find a mailing list.
thanks,
Thufir
Hello,
On 08.06.17 00:38, Thufir Hawat wrote:
is there a discussion list for openrcs? I couldn't find a mailing list.
are you referring to openrcs.com sip service?
If yes, you can ask here. It is a service running kamailio dev version to ensure basic features are working.
Cheers, Daniel
What are the connection parameters? When I try to connect the Jitsi log shows:
01:24:37.309 INFO: [476] impl.protocol.sip.SipLogger.logInfo().185 Info from the JAIN-SIP stack: the sip stack timer gov.nist.javax.sip.stack.timers.DefaultSipTimer has been started 01:24:48.007 INFO: [488] impl.protocol.sip.SipLogger.logInfo().185 Info from the JAIN-SIP stack: the sip stack timer gov.nist.javax.sip.stack.timers.DefaultSipTimer has been stopped 01:24:49.009 INFO: [488] impl.protocol.sip.SipLogger.logInfo().185 Info from the JAIN-SIP stack: the sip stack timer gov.nist.javax.sip.stack.timers.DefaultSipTimer has been stopped
which doesn't give me much info. If there's a different client which works better out of the box, do let me know.
thanks,
Thufir
On Thu, 8 Jun 2017, Daniel-Constantin Mierla wrote:
Hello,
On 08.06.17 00:38, Thufir Hawat wrote:
is there a discussion list for openrcs? I couldn't find a mailing list.
are you referring to openrcs.com sip service?
If yes, you can ask here. It is a service running kamailio dev version to ensure basic features are working.
Cheers, Daniel
-- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com
On 08.06.17 10:26, Thufir Hawat wrote:
What are the connection parameters? When I try to connect the Jitsi log shows:
01:24:37.309 INFO: [476] impl.protocol.sip.SipLogger.logInfo().185 Info from the JAIN-SIP stack: the sip stack timer gov.nist.javax.sip.stack.timers.DefaultSipTimer has been started 01:24:48.007 INFO: [488] impl.protocol.sip.SipLogger.logInfo().185 Info from the JAIN-SIP stack: the sip stack timer gov.nist.javax.sip.stack.timers.DefaultSipTimer has been stopped 01:24:49.009 INFO: [488] impl.protocol.sip.SipLogger.logInfo().185 Info from the JAIN-SIP stack: the sip stack timer gov.nist.javax.sip.stack.timers.DefaultSipTimer has been stopped
which doesn't give me much info. If there's a different client which works better out of the box, do let me know.
It should work by setting domain or proxy to openrcs.com. I used Jitsi over the time and all was fine. Maybe your ISP is blocking SIP/VoIP. Try with transport TLS (port 5061). For UDP, in addition to default SIP port 5060, Kamailio is listening on 5062.
Cheers, Daniel
thanks,
Thufir
On Thu, 8 Jun 2017, Daniel-Constantin Mierla wrote:
Hello,
On 08.06.17 00:38, Thufir Hawat wrote:
is there a discussion list for openrcs? I couldn't find a mailing list.
are you referring to openrcs.com sip service?
If yes, you can ask here. It is a service running kamailio dev version to ensure basic features are working.
Cheers, Daniel
-- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com
On Thu, 8 Jun 2017, Daniel-Constantin Mierla wrote:
On 08.06.17 10:26, Thufir Hawat wrote:
..
It should work by setting domain or proxy to openrcs.com. I used Jitsi over the time and all was fine. Maybe your ISP is blocking SIP/VoIP. Try with transport TLS (port 5061). For UDP, in addition to default SIP port 5060, Kamailio is listening on 5062.
Cheers, Daniel
thanks,
..
ok, good to know that openrcs works with Jitsi, as hoped and expected. I don't think that the ISP is blocking SIP because I'm able to use anveo with an old Linksys IP hard phone, and that uses SIP.
Now, I was able to get, so far as I can tell, thufir@sip2sip.info working on a Grandstream 1625 but would like to use openrcs because kamailio intrigues me.
Each SIP account needs its own port number? Starting at 5060 and going up. Why does kamailio listen on 5062? This means that the *outbound* settings need to specify 5062, but inbound might be most anything?
thanks,
Thufir
On 09.06.17 07:18, Thufir Hawat wrote:
On Thu, 8 Jun 2017, Daniel-Constantin Mierla wrote:
On 08.06.17 10:26, Thufir Hawat wrote:
..
It should work by setting domain or proxy to openrcs.com. I used Jitsi over the time and all was fine. Maybe your ISP is blocking SIP/VoIP. Try with transport TLS (port 5061). For UDP, in addition to default SIP port 5060, Kamailio is listening on 5062.
Cheers, Daniel
thanks,
..
ok, good to know that openrcs works with Jitsi, as hoped and expected. I don't think that the ISP is blocking SIP because I'm able to use anveo with an old Linksys IP hard phone, and that uses SIP.
Now, I was able to get, so far as I can tell, thufir@sip2sip.info working on a Grandstream 1625 but would like to use openrcs because kamailio intrigues me.
Each SIP account needs its own port number? Starting at 5060 and going up. Why does kamailio listen on 5062? This means that the *outbound* settings need to specify 5062, but inbound might be most anything?
No, it is no need for own port for each sip account. Kamailio on openrcs.com listens on 5062 as well as on the default 5060. The reason for 5062 is that some corporate/isp companies block port 5060, or there are some broken SIP ALGs on home routers mangling traffic for port 5060, messing it.
So you can use default port 5060.
What operating system are you using for Jitsi? Can you sniff network traffic on port 5060? If you are on linux, try:
ngrep -d any -qt -W byline port 5060
On windows you can use wireshark.
The output from the sniffer is good to see what happens on the network.
Also, be aware that openrcs.com has a different password for SIP account than the one used for web interface. You have to go to the SIP account page and see/generate it there.
Cheers, Daniel
On Fri, 9 Jun 2017, Daniel-Constantin Mierla wrote:
On 09.06.17 07:18, Thufir Hawat wrote:
On Thu, 8 Jun 2017, Daniel-Constantin Mierla wrote:
On 08.06.17 10:26, Thufir Hawat wrote:
..
Each SIP account needs its own port number? Starting at 5060 and going up. Why does kamailio listen on 5062? This means that the *outbound* settings need to specify 5062, but inbound might be most anything?
No, it is no need for own port for each sip account. Kamailio on openrcs.com listens on 5062 as well as on the default 5060. The reason for 5062 is that some corporate/isp companies block port 5060, or there are some broken SIP ALGs on home routers mangling traffic for port 5060, messing it.
So you can use default port 5060.
What operating system are you using for Jitsi? Can you sniff network traffic on port 5060? If you are on linux, try:
ngrep -d any -qt -W byline port 5060
On windows you can use wireshark.
The output from the sniffer is good to see what happens on the network.
Also, be aware that openrcs.com has a different password for SIP account than the one used for web interface. You have to go to the SIP account page and see/generate it there.
perfect. Please check this reasoning: *because* anveo works fine from an IP phone *therefore* the SIP ALG on my home router isn't mangling traffic on 5060. (Anveo uses SIP.)
Yes, I saw that openrcs has commendable security on the password. I'll take a look with ngrep from Ubuntu.
thanks,
Thufir
On 09.06.17 08:58, Thufir Hawat wrote:
On Fri, 9 Jun 2017, Daniel-Constantin Mierla wrote:
On 09.06.17 07:18, Thufir Hawat wrote:
On Thu, 8 Jun 2017, Daniel-Constantin Mierla wrote:
On 08.06.17 10:26, Thufir Hawat wrote:
..
Each SIP account needs its own port number? Starting at 5060 and going up. Why does kamailio listen on 5062? This means that the *outbound* settings need to specify 5062, but inbound might be most anything?
No, it is no need for own port for each sip account. Kamailio on openrcs.com listens on 5062 as well as on the default 5060. The reason for 5062 is that some corporate/isp companies block port 5060, or there are some broken SIP ALGs on home routers mangling traffic for port 5060, messing it.
So you can use default port 5060.
What operating system are you using for Jitsi? Can you sniff network traffic on port 5060? If you are on linux, try:
ngrep -d any -qt -W byline port 5060
On windows you can use wireshark.
The output from the sniffer is good to see what happens on the network.
Also, be aware that openrcs.com has a different password for SIP account than the one used for web interface. You have to go to the SIP account page and see/generate it there.
perfect. Please check this reasoning: *because* anveo works fine from an IP phone *therefore* the SIP ALG on my home router isn't mangling traffic on 5060. (Anveo uses SIP.)
Yes, I saw that openrcs has commendable security on the password. I'll take a look with ngrep from Ubuntu.
I just downloaded latest Jitsi, started it, and set the SIP account to username@openrcs.com and put in there the password. It registered from the first attempt, with the default settings (udp on port 5060). So vary likely something is misconfigured in your jitsi or in the network, if it doesn't work for you.
Cheers, Daniel
On Fri, 9 Jun 2017, Daniel-Constantin Mierla wrote:
On 09.06.17 08:58, Thufir Hawat wrote:
On Fri, 9 Jun 2017, Daniel-Constantin Mierla wrote:
On 09.06.17 07:18, Thufir Hawat wrote:
On Thu, 8 Jun 2017, Daniel-Constantin Mierla wrote:
On 08.06.17 10:26, Thufir Hawat wrote:
..
Each SIP account needs its own port number? Starting at 5060 and going up. Why does kamailio listen on 5062? This means that the *outbound* settings need to specify 5062, but inbound might be most anything?
No, it is no need for own port for each sip account. Kamailio on openrcs.com listens on 5062 as well as on the default 5060. The reason for 5062 is that some corporate/isp companies block port 5060, or there are some broken SIP ALGs on home routers mangling traffic for port 5060, messing it.
So you can use default port 5060.
What operating system are you using for Jitsi? Can you sniff network traffic on port 5060? If you are on linux, try:
ngrep -d any -qt -W byline port 5060
On windows you can use wireshark.
The output from the sniffer is good to see what happens on the network.
Also, be aware that openrcs.com has a different password for SIP account than the one used for web interface. You have to go to the SIP account page and see/generate it there.
perfect. Please check this reasoning: *because* anveo works fine from an IP phone *therefore* the SIP ALG on my home router isn't mangling traffic on 5060. (Anveo uses SIP.)
Yes, I saw that openrcs has commendable security on the password. I'll take a look with ngrep from Ubuntu.
I just downloaded latest Jitsi, started it, and set the SIP account to username@openrcs.com and put in there the password. It registered from the first attempt, with the default settings (udp on port 5060). So vary likely something is misconfigured in your jitsi or in the network, if it doesn't work for you.
Cheers, Daniel
-- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com
thufir@doge:~$ thufir@doge:~$ sudo ngrep -d any -qt -W byline port 5060 interface: any filter: (ip or ip6) and ( port 5060 )
U 2017/06/09 00:33:13.016586 192.168.1.7:5060 -> 78.47.51.102:5060 REGISTER sip:openrcs.com SIP/2.0. Call-ID: a73d7a44c5ca9d9ed23d347e38a560cd@0:0:0:0:0:0:0:0. CSeq: 1 REGISTER. From: "thufir" sip:thufir@openrcs.com;tag=e6261e66. To: "thufir" sip:thufir@openrcs.com. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-393333-d483503bcdbe2ba54859c2878d3cddfd. Max-Forwards: 70. User-Agent: Jitsi2.8.5426Linux. Expires: 600. Contact: "thufir" sip:thufir@192.168.1.7:5060;transport=udp;registering_acc=openrcs_com;expires=600. Content-Length: 0. .
U 2017/06/09 00:33:13.207083 78.47.51.102:5060 -> 192.168.1.7:5060 SIP/2.0 401 Unauthorized. Call-ID: a73d7a44c5ca9d9ed23d347e38a560cd@0:0:0:0:0:0:0:0. CSeq: 1 REGISTER. From: "thufir" sip:thufir@openrcs.com;tag=e6261e66. To: "thufir" sip:thufir@openrcs.com;tag=897398a4e4993a34f7061d7cde36a9dd.747f. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-393333-d483503bcdbe2ba54859c2878d3cddfd;rport=49204;received=192.157.119.2. WWW-Authenticate: Digest realm="openrcs.com", nonce="WTpQZVk6TzlB3R+ar4G+iNBsiVv/CCYb". Server: kamailio (5.1.0-dev3 (x86_64/linux)). Content-Length: 0. .
U 2017/06/09 00:33:13.214224 192.168.1.7:5060 -> 78.47.51.102:5060 REGISTER sip:openrcs.com SIP/2.0. Call-ID: a73d7a44c5ca9d9ed23d347e38a560cd@0:0:0:0:0:0:0:0. CSeq: 2 REGISTER. From: "thufir" sip:thufir@openrcs.com;tag=e6261e66. To: "thufir" sip:thufir@openrcs.com. Max-Forwards: 70. User-Agent: Jitsi2.8.5426Linux. Expires: 600. Contact: "thufir" sip:thufir@192.168.1.7:5060;transport=udp;registering_acc=openrcs_com;expires=600. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-393333-895d23a7b018122554dfc31aa33b18e8. Authorization: Digest username="thufir",realm="openrcs.com",nonce="WTpQZVk6TzlB3R+ar4G+iNBsiVv/CCYb",uri="sip:openrcs.com",response="04179dd6d394ab270525e8b349a3b33d". Content-Length: 0. .
U 2017/06/09 00:33:13.410068 78.47.51.102:5060 -> 192.168.1.7:5060 SIP/2.0 401 Unauthorized. Call-ID: a73d7a44c5ca9d9ed23d347e38a560cd@0:0:0:0:0:0:0:0. CSeq: 2 REGISTER. From: "thufir" sip:thufir@openrcs.com;tag=e6261e66. To: "thufir" sip:thufir@openrcs.com;tag=897398a4e4993a34f7061d7cde36a9dd.3d80. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-393333-895d23a7b018122554dfc31aa33b18e8;rport=49204;received=192.157.119.2. WWW-Authenticate: Digest realm="openrcs.com", nonce="WTpQZVk6TzlB3R+ar4G+iNBsiVv/CCYb". Server: kamailio (5.1.0-dev3 (x86_64/linux)). Content-Length: 0. .
^Cthufir@doge:~$ thufir@doge:~$
The config is like so:
https://pasteboard.co/gnodqc0dg.png
I'm able to use ekiga from Android, but that's using STUN or something behind the scenes with CSipSimple. Also, not only does anveo work on the hardphone, but so does sip2sip.info insofar as I'm able to call and get sound from Ekiga (on Android) to the IP phone.
The inference I draw, if not conclusion, is that the network is ok.
but, the 401 unauthorized looks like password problem. All I can say is that copy/pasted from the generated password...
thanks,
Thufir
On 09.06.17 09:39, Thufir Hawat wrote:
[...]
I'm able to use ekiga from Android, but that's using STUN or something behind the scenes with CSipSimple. Also, not only does anveo work on the hardphone, but so does sip2sip.info insofar as I'm able to call and get sound from Ekiga (on Android) to the IP phone.
The inference I draw, if not conclusion, is that the network is ok.
but, the 401 unauthorized looks like password problem. All I can say is that copy/pasted from the generated password...
Paste the password first in an editor, where you can see it and proof-check -- it happens when copying from web pages that an additional character is at the end (e.g., space, tab, end of line).
Cheers, Daniel
thanks for taking a look. I'm kinda confused in that everything except kamailio, through openrcs, works. I just punched in default and sip.antisip.com worked like a charm.
I'll re-do the password for openrcs.
here's a chat in SIP between Android running CSipSimple and Jitsi on a laptop running Ubuntu:
thufir@doge:~$ thufir@doge:~$ thufir@doge:~$ sudo ngrep -d any -qt -W byline port 5060 interface: any filter: (ip or ip6) and ( port 5060 )
U 2017/06/09 01:51:39.142529 192.168.1.7:5060 -> 91.121.30.149:5060 OPTIONS sip:sip.antisip.com SIP/2.0. Call-ID: 6ab3725733b7b1b508491b0525b0c3ff@0:0:0:0:0:0:0:0. CSeq: 8 OPTIONS. From: "thufir" sip:thufir@sip.antisip.com;tag=b9cad9a. To: "thufir" sip:thufir@sip.antisip.com. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-373433-17e33965a20de956f1138f9c1b8db537. Max-Forwards: 70. Contact: "thufir" sip:thufir@192.168.1.7:5060;transport=udp;registering_acc=sip_antisip_com. User-Agent: Jitsi2.8.5426Linux. Allow: INFO,UPDATE,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE. Allow-Events: refer,conference,remote-control,presence,presence.winfo,message-summary. Content-Length: 0. .
U 2017/06/09 01:51:39.309161 91.121.30.149:5060 -> 192.168.1.7:5060 SIP/2.0 407 Proxy Authentication Required. Call-ID: 6ab3725733b7b1b508491b0525b0c3ff@0:0:0:0:0:0:0:0. CSeq: 8 OPTIONS. From: "thufir" sip:thufir@sip.antisip.com;tag=b9cad9a. To: "thufir" sip:thufir@sip.antisip.com;tag=7b539f556d5094779f74e99f9427a6d9.e11d. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-373433-17e33965a20de956f1138f9c1b8db537;rport=49206;received=192.157.119.2. Proxy-Authenticate: Digest realm="sip.antisip.com", nonce="WTpnXlk6Y3afhlswO/DZX1TDOsrOwoNo". Server: kamailio (4.4.2 (x86_64/linux)). Content-Length: 0. .
U 2017/06/09 01:51:43.859671 91.121.30.149:5060 -> 192.168.1.7:5060 ....
U 2017/06/09 01:51:50.360157 91.121.30.149:5060 -> 192.168.1.7:5060 MESSAGE sip:thufir@192.157.119.2;transport=udp;registering_acc=sip_antisip_com SIP/2.0. Via: SIP/2.0/UDP 91.121.30.149;branch=z9hG4bK0334.3aa3285bee183976abfa903b2d789f50.0. Via: SIP/2.0/UDP 86.64.162.35;rport=5060;branch=z9hG4bK0334.22757207.0. Via: SIP/2.0/UDP 192.157.119.2:51882;received=192.157.119.2;rport=51882;branch=z9hG4bKPjlm1ig621AznsUz9pYXE-Gv9Vf5epajBv. Max-Forwards: 68. From: sip:thufir@ekiga.net;tag=mJtCqK2oo-M2Xi5ZruW0tpcIwfCCjuX.. To: sip:thufir@sip.antisip.com. Call-ID: jDEDD8ksx1ZWrtqcipKyegl.i8HCRjHj. CSeq: 18100 MESSAGE. Accept: text/plain, application/im-iscomposing+xml. User-Agent: CSipSimple_RCT6873W42M-23/r2457. Content-Type: text/plain. Content-Length: 23. .
From Android csipsimple
U 2017/06/09 01:51:50.364688 192.168.1.7:5060 -> 91.121.30.149:5060 SIP/2.0 200 OK. CSeq: 18100 MESSAGE. Call-ID: jDEDD8ksx1ZWrtqcipKyegl.i8HCRjHj. From: sip:thufir@ekiga.net;tag=mJtCqK2oo-M2Xi5ZruW0tpcIwfCCjuX.. To: sip:thufir@sip.antisip.com;tag=429cf21b. Via: SIP/2.0/UDP 91.121.30.149;branch=z9hG4bK0334.3aa3285bee183976abfa903b2d789f50.0,SIP/2.0/UDP 86.64.162.35;rport=5060;branch=z9hG4bK0334.22757207.0,SIP/2.0/UDP 192.157.119.2:51882;received=192.157.119.2;rport=51882;branch=z9hG4bKPjlm1ig621AznsUz9pYXE-Gv9Vf5epajBv. Contact: "thufir" sip:thufir@192.168.1.7:5060;transport=udp;registering_acc=sip_antisip_com. User-Agent: Jitsi2.8.5426Linux. Content-Length: 0. .
U 2017/06/09 01:51:59.809748 192.168.1.7:5060 -> 91.121.30.149:5060 MESSAGE sip:thufir@ekiga.net SIP/2.0. Call-ID: 7434a2326d67c3572d6bb6c8b080ff02@0:0:0:0:0:0:0:0. CSeq: 1622093140 MESSAGE. From: "thufir" sip:thufir@sip.antisip.com;tag=ecf886d. To: sip:thufir@ekiga.net. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-373433-c324d9b8e3e84d56d039709f8beb88f7. Max-Forwards: 70. Content-Type: text/plain. Contact: "thufir" sip:thufir@192.168.1.7:5060;transport=udp;registering_acc=sip_antisip_com. User-Agent: Jitsi2.8.5426Linux. Content-Length: 15. . back from jitsi
U 2017/06/09 01:52:00.038338 91.121.30.149:5060 -> 192.168.1.7:5060 SIP/2.0 407 Proxy Authentication Required. Call-ID: 7434a2326d67c3572d6bb6c8b080ff02@0:0:0:0:0:0:0:0. CSeq: 1622093140 MESSAGE. From: "thufir" sip:thufir@sip.antisip.com;tag=ecf886d. To: sip:thufir@ekiga.net;tag=7b539f556d5094779f74e99f9427a6d9.5f66. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-373433-c324d9b8e3e84d56d039709f8beb88f7;rport=49206;received=192.157.119.2. Proxy-Authenticate: Digest realm="sip.antisip.com", nonce="WTpnc1k6Y4twt/ASz+2Wsb1IBT9ilPBQ". Server: kamailio (4.4.2 (x86_64/linux)). Content-Length: 0. .
U 2017/06/09 01:52:00.046779 192.168.1.7:5060 -> 91.121.30.149:5060 MESSAGE sip:thufir@ekiga.net SIP/2.0. Call-ID: 7434a2326d67c3572d6bb6c8b080ff02@0:0:0:0:0:0:0:0. CSeq: 1622093141 MESSAGE. From: "thufir" sip:thufir@sip.antisip.com;tag=ecf886d. To: sip:thufir@ekiga.net. Max-Forwards: 70. Content-Type: text/plain. Contact: "thufir" sip:thufir@192.168.1.7:5060;transport=udp;registering_acc=sip_antisip_com. User-Agent: Jitsi2.8.5426Linux. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-373433-7a83dc96fd78f8ee9782376e8d1277dd. Proxy-Authorization: Digest username="thufir",realm="sip.antisip.com",nonce="WTpnc1k6Y4twt/ASz+2Wsb1IBT9ilPBQ",uri="sip:thufir@ekiga.net",response="26b00e4135b1fec9cc32a9596544c175". Content-Length: 15. . back from jitsi
U 2017/06/09 01:52:00.548661 192.168.1.7:5060 -> 91.121.30.149:5060 MESSAGE sip:thufir@ekiga.net SIP/2.0. Call-ID: 7434a2326d67c3572d6bb6c8b080ff02@0:0:0:0:0:0:0:0. CSeq: 1622093141 MESSAGE. From: "thufir" sip:thufir@sip.antisip.com;tag=ecf886d. To: sip:thufir@ekiga.net. Max-Forwards: 70. Content-Type: text/plain. Contact: "thufir" sip:thufir@192.168.1.7:5060;transport=udp;registering_acc=sip_antisip_com. User-Agent: Jitsi2.8.5426Linux. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-373433-7a83dc96fd78f8ee9782376e8d1277dd. Proxy-Authorization: Digest username="thufir",realm="sip.antisip.com",nonce="WTpnc1k6Y4twt/ASz+2Wsb1IBT9ilPBQ",uri="sip:thufir@ekiga.net",response="26b00e4135b1fec9cc32a9596544c175". Content-Length: 15. . back from jitsi
U 2017/06/09 01:52:01.548112 192.168.1.7:5060 -> 91.121.30.149:5060 MESSAGE sip:thufir@ekiga.net SIP/2.0. Call-ID: 7434a2326d67c3572d6bb6c8b080ff02@0:0:0:0:0:0:0:0. CSeq: 1622093141 MESSAGE. From: "thufir" sip:thufir@sip.antisip.com;tag=ecf886d. To: sip:thufir@ekiga.net. Max-Forwards: 70. Content-Type: text/plain. Contact: "thufir" sip:thufir@192.168.1.7:5060;transport=udp;registering_acc=sip_antisip_com. User-Agent: Jitsi2.8.5426Linux. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-373433-7a83dc96fd78f8ee9782376e8d1277dd. Proxy-Authorization: Digest username="thufir",realm="sip.antisip.com",nonce="WTpnc1k6Y4twt/ASz+2Wsb1IBT9ilPBQ",uri="sip:thufir@ekiga.net",response="26b00e4135b1fec9cc32a9596544c175". Content-Length: 15. . back from jitsi
U 2017/06/09 01:52:01.812779 91.121.30.149:5060 -> 192.168.1.7:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.7:5060;rport=49206;received=192.157.119.2;branch=z9hG4bK-373433-7a83dc96fd78f8ee9782376e8d1277dd. Call-ID: 7434a2326d67c3572d6bb6c8b080ff02@0:0:0:0:0:0:0:0. From: "thufir" sip:thufir@sip.antisip.com;tag=ecf886d. To: sip:thufir@ekiga.net;tag=z9hG4bK78c4.4306f927.0. CSeq: 1622093141 MESSAGE. Content-Length: 0. .
U 2017/06/09 01:52:04.141737 192.168.1.7:5060 -> 91.121.30.149:5060 OPTIONS sip:sip.antisip.com SIP/2.0. Call-ID: 9646c19414d902937c87a32e346fb590@0:0:0:0:0:0:0:0. CSeq: 9 OPTIONS. From: "thufir" sip:thufir@sip.antisip.com;tag=dc854c2d. To: "thufir" sip:thufir@sip.antisip.com. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-373433-132f0dd271d8b29cc2f047eaa61923cc. Max-Forwards: 70. Contact: "thufir" sip:thufir@192.168.1.7:5060;transport=udp;registering_acc=sip_antisip_com. User-Agent: Jitsi2.8.5426Linux. Allow: INFO,UPDATE,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE. Allow-Events: refer,conference,remote-control,presence,presence.winfo,message-summary. Content-Length: 0. .
U 2017/06/09 01:52:04.311070 91.121.30.149:5060 -> 192.168.1.7:5060 SIP/2.0 407 Proxy Authentication Required. Call-ID: 9646c19414d902937c87a32e346fb590@0:0:0:0:0:0:0:0. CSeq: 9 OPTIONS. From: "thufir" sip:thufir@sip.antisip.com;tag=dc854c2d. To: "thufir" sip:thufir@sip.antisip.com;tag=7b539f556d5094779f74e99f9427a6d9.2127. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-373433-132f0dd271d8b29cc2f047eaa61923cc;rport=49206;received=192.157.119.2. Proxy-Authenticate: Digest realm="sip.antisip.com", nonce="WTpnd1k6Y49Ts68nvP/bOHuNe4SSm8ZR". Server: kamailio (4.4.2 (x86_64/linux)). Content-Length: 0. .
^Cthufir@doge:~$ thufir@doge:~$ thufir@doge:~$
On 09.06.17 10:58, Thufir Hawat wrote:
thanks for taking a look. I'm kinda confused in that everything except kamailio, through openrcs, works. I just punched in default and sip.antisip.com worked like a charm.
I'll re-do the password for openrcs.
[...]
The 401 from your previous email is due to authentication failure, so the password used there was wrong. Try again and be sure the password is introduced correctly.
Cheers, Daniel
-
Daniel-Constantin Mierla -
Thufir Hawat