Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP:192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" sip:1300@networklab.loc;tag=as0a530a8d To: sip:192.168.100.145;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.100.145:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP:192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" sip:1300@networklab.loc;tag=as0a530a8d To: sip:192.168.100.145;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.100.145:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello Pedro, So test case is simple. asterisk is play voicemail role and kamailio is gateway. Asterisk peer is setup and working. I think message SIP/2.0 484 Address Incomplete is some from kamailio
When Asterisk send reply back this message show up because is no $rU in header line.
I don't know how to correct it that it will full uri format line.
voice 192.168.10.120 Auto (No) No A 5060 OK (2 ms) 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline]
<--- SIP read from UDP:192.168.10.120:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK4c787422 From: "asterisk" sip:1300@networklab.loc;tag=as5c659db3 To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2 Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.10.120:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
<-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc' Method: OPTIONS
<--- SIP read from UDP:192.168.10.120:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK4c787422 From: "asterisk" sip:1300@networklab.loc;tag=as5c659db3 To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2 Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.10.120:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; }
Slava.
----- Original Message -----
From: "Pedro Niño" nino.pedro@gmail.com To: "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org Sent: Sunday, March 30, 2014 8:30:56 PM Subject: Re: [SR-Users] message 484
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" < volga629@networklab.ca > escribió:
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP: 192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" sip:1300@networklab.loc;tag=as0a530a8d To: sip:192.168.100.145;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.100.145:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Ok, I wonder....
If this is a message you're seeing at the asterisk server, it may be related to the qualify=yes or qualify=Number parameter in the peer, at sip.conf.
If it's right, then you can modify at 2 places: one, by disabling qualify. (qualify=no) at asterisk, or the other by configuring Kamailio to answer a 200 "OK" message when the message comes from the asterisk server.
If not, can you explain when are you seeing such behavior? And can run a 'sngrep host <asterisk_IP>' at the Kamailio server?
Keep me posted. El mar 30, 2014 8:26 PM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Pedro, So test case is simple. asterisk is play voicemail role and kamailio is gateway. Asterisk peer is setup and working. I think message SIP/2.0 484 Address Incomplete is some from kamailio
When Asterisk send reply back this message show up because is no $rU in header line.
I don't know how to correct it that it will full uri format line.
voice 192.168.10.120 Auto (No) No A 5060 OK (2 ms) 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline]
<--- SIP read from UDP:192.168.10.120:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK4c787422 From: "asterisk" sip:1300@networklab.loc;tag=as5c659db3 To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2 Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.10.120:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
<-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc' Method: OPTIONS
<--- SIP read from UDP:192.168.10.120:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK4c787422 From: "asterisk" sip:1300@networklab.loc;tag=as5c659db3 To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2 Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.10.120:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; }Slava.
*From: *"Pedro Niño" nino.pedro@gmail.com *To: *"Kamailio (SER) - Users Mailing List" <sr-users@lists.sip-router.org
*Sent: *Sunday, March 30, 2014 8:30:56 PM *Subject: *Re: [SR-Users] message 484
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP:192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" sip:1300@networklab.loc;tag=as0a530a8d To: sip:192.168.100.145;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.100.145:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
By the way, I'm attaching a excellent response from another people at this mailing list, about using qualify and modding Kamailio config to make it answer.
Check it, may be your case
------------
On Monday 24 March 2014 15:23:31 Alexandr Usov wrote:
Peers (from Internet behind NAT) registered on Kamailio (local ip 192.168.182.1), calls from/to routed via Asterisk (192.168.182.24).
Can't use qualify info:
<--- SIP read from UDP:192.168.182.1:5060 ---> SIP/2.0 484 Address Incomplete
...
<------------->
Check your kamailio.cfg. In the "default" config OPTIONS get this response due to: if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; }
You'll have to response with a 200 yourself, eg:
route[REQINIT]{ ... if($si=="192.168.182.24" && is_method("OPTIONS")) { sl_send_reply("200","Up and running"); exit; }
It's up to you to decide to which OPTIONS requests to response with what code. El mar 30, 2014 9:34 PM, "Pedro Niño" nino.pedro@gmail.com escribió:
Ok, I wonder....
If this is a message you're seeing at the asterisk server, it may be related to the qualify=yes or qualify=Number parameter in the peer, at sip.conf.
If it's right, then you can modify at 2 places: one, by disabling qualify. (qualify=no) at asterisk, or the other by configuring Kamailio to answer a 200 "OK" message when the message comes from the asterisk server.
If not, can you explain when are you seeing such behavior? And can run a 'sngrep host <asterisk_IP>' at the Kamailio server?
Keep me posted. El mar 30, 2014 8:26 PM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Pedro, So test case is simple. asterisk is play voicemail role and kamailio is gateway. Asterisk peer is setup and working. I think message SIP/2.0 484 Address Incomplete is some from kamailio
When Asterisk send reply back this message show up because is no $rU in header line.
I don't know how to correct it that it will full uri format line.
voice 192.168.10.120 Auto (No) No A 5060 OK (2 ms) 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline]
<--- SIP read from UDP:192.168.10.120:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK4c787422 From: "asterisk" sip:1300@networklab.loc;tag=as5c659db3 To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2 Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.10.120:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
<-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc' Method: OPTIONS
<--- SIP read from UDP:192.168.10.120:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK4c787422 From: "asterisk" sip:1300@networklab.loc;tag=as5c659db3 To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2 Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.10.120:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; }Slava.
*From: *"Pedro Niño" nino.pedro@gmail.com *To: *"Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org> *Sent: *Sunday, March 30, 2014 8:30:56 PM *Subject: *Re: [SR-Users] message 484
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP:192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" sip:1300@networklab.loc;tag=as0a530a8d To: sip:192.168.100.145;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.100.145:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello Pedro, I set in main routing section and it I see 200 OK right now. Than you for help.
# account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting }
if (is_method("OPTIONS")) { sl_send_reply("200", "OK"); exit; }
U 2014/03/30 22:32:47.139646 192.168.10.120:5062 -> 192.168.10.120:5060 OPTIONS sip:192.168.10.120 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK6f8354a0. Max-Forwards: 70. From: "asterisk" sip:1300@networklab.loc;tag=as2cbae229. To: sip:192.168.10.120. Contact: sip:1300@192.168.10.120:5062. Call-ID: 456f80c06d85d9b34027ccc533855f72@networklab.loc. CSeq: 102 OPTIONS. User-Agent: Asterisk PBX 12.0.0. Date: Mon, 31 Mar 2014 02:32:47 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Length: 0. .
U 2014/03/30 22:32:47.140301 192.168.10.120:5060 -> 192.168.10.120:5062 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK6f8354a0. From: "asterisk" sip:1300@networklab.loc;tag=as2cbae229. To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.93f4. Call-ID: 456f80c06d85d9b34027ccc533855f72@networklab.loc. CSeq: 102 OPTIONS. Server: kamailio (4.1.2 (x86_64/linux)). Content-Length: 0.
Slava.
----- Original Message -----
From: "Pedro Niño" nino.pedro@gmail.com To: "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org Sent: Sunday, March 30, 2014 10:04:30 PM Subject: Re: [SR-Users] message 484
Ok, I wonder....
If this is a message you're seeing at the asterisk server, it may be related to the qualify=yes or qualify=Number parameter in the peer, at sip.conf.
If it's right, then you can modify at 2 places: one, by disabling qualify. (qualify=no) at asterisk, or the other by configuring Kamailio to answer a 200 "OK" message when the message comes from the asterisk server.
If not, can you explain when are you seeing such behavior? And can run a 'sngrep host <asterisk_IP>' at the Kamailio server?
Keep me posted. El mar 30, 2014 8:26 PM, "Slava Bendersky" < volga629@networklab.ca > escribió:
Hello Pedro, So test case is simple. asterisk is play voicemail role and kamailio is gateway. Asterisk peer is setup and working. I think message SIP/2.0 484 Address Incomplete is some from kamailio
When Asterisk send reply back this message show up because is no $rU in header line.
I don't know how to correct it that it will full uri format line.
voice 192.168.10.120 Auto (No) No A 5060 OK (2 ms) 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline]
<--- SIP read from UDP: 192.168.10.120:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK4c787422 From: "asterisk" sip:1300@networklab.loc;tag=as5c659db3 To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2 Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.10.120:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
<-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc' Method: OPTIONS
<--- SIP read from UDP: 192.168.10.120:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK4c787422 From: "asterisk" sip:1300@networklab.loc;tag=as5c659db3 To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2 Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.10.120:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; }
Slava.
From: "Pedro Niño" < nino.pedro@gmail.com > To: "Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org > Sent: Sunday, March 30, 2014 8:30:56 PM Subject: Re: [SR-Users] message 484
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" < volga629@networklab.ca > escribió:
<blockquote>
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP: 192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" sip:1300@networklab.loc;tag=as0a530a8d To: sip:192.168.100.145;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.100.145:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Just be aware, with this any options message from anybody will be answered with ok.
I would prefer to make it specific, at least to be sure that only happens when the source is the asterisk server, and in the right case, because you could get strange behaviors.
Keep testing, and be sure is the right behavior that you need. El mar 30, 2014 10:04 PM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Pedro, I set in main routing section and it I see 200 OK right now. Than you for help.
# account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if (is_method("OPTIONS")) { sl_send_reply("200", "OK"); exit; }U 2014/03/30 22:32:47.139646 192.168.10.120:5062 -> 192.168.10.120:5060 OPTIONS sip:192.168.10.120 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK6f8354a0. Max-Forwards: 70. From: "asterisk" sip:1300@networklab.loc;tag=as2cbae229. To: sip:192.168.10.120. Contact: sip:1300@192.168.10.120:5062. Call-ID: 456f80c06d85d9b34027ccc533855f72@networklab.loc. CSeq: 102 OPTIONS. User-Agent: Asterisk PBX 12.0.0. Date: Mon, 31 Mar 2014 02:32:47 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Length: 0. .
U 2014/03/30 22:32:47.140301 192.168.10.120:5060 -> 192.168.10.120:5062 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK6f8354a0. From: "asterisk" sip:1300@networklab.loc;tag=as2cbae229. To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.93f4. Call-ID: 456f80c06d85d9b34027ccc533855f72@networklab.loc. CSeq: 102 OPTIONS. Server: kamailio (4.1.2 (x86_64/linux)). Content-Length: 0.
Slava.
*From: *"Pedro Niño" nino.pedro@gmail.com *To: *"Kamailio (SER) - Users Mailing List" <sr-users@lists.sip-router.org
*Sent: *Sunday, March 30, 2014 10:04:30 PM *Subject: *Re: [SR-Users] message 484
Ok, I wonder....
If this is a message you're seeing at the asterisk server, it may be related to the qualify=yes or qualify=Number parameter in the peer, at sip.conf.
If it's right, then you can modify at 2 places: one, by disabling qualify. (qualify=no) at asterisk, or the other by configuring Kamailio to answer a 200 "OK" message when the message comes from the asterisk server.
If not, can you explain when are you seeing such behavior? And can run a 'sngrep host <asterisk_IP>' at the Kamailio server?
Keep me posted. El mar 30, 2014 8:26 PM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Pedro, So test case is simple. asterisk is play voicemail role and kamailio is gateway. Asterisk peer is setup and working. I think message SIP/2.0 484 Address Incomplete is some from kamailio
When Asterisk send reply back this message show up because is no $rU in header line.
I don't know how to correct it that it will full uri format line.
voice 192.168.10.120 Auto (No) No A 5060 OK (2 ms) 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline]
<--- SIP read from UDP:192.168.10.120:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK4c787422 From: "asterisk" sip:1300@networklab.loc;tag=as5c659db3 To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2 Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.10.120:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
<-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc' Method: OPTIONS
<--- SIP read from UDP:192.168.10.120:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK4c787422 From: "asterisk" sip:1300@networklab.loc;tag=as5c659db3 To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2 Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.10.120:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; }Slava.
*From: *"Pedro Niño" nino.pedro@gmail.com *To: *"Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org> *Sent: *Sunday, March 30, 2014 8:30:56 PM *Subject: *Re: [SR-Users] message 484
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP:192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" sip:1300@networklab.loc;tag=as0a530a8d To: sip:192.168.100.145;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.100.145:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello Pedro, I modified this section to
if (is_method("OPTIONS") || allow_trusted("$si","$(proto)")) { sl_send_reply("200", "OK"); exit; }
I am maintaining ACL so should be more tight.
Slava.
----- Original Message -----
From: "Pedro Niño" nino.pedro@gmail.com To: "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org Sent: Sunday, March 30, 2014 11:20:50 PM Subject: Re: [SR-Users] message 484
Just be aware, with this any options message from anybody will be answered with ok.
I would prefer to make it specific, at least to be sure that only happens when the source is the asterisk server, and in the right case, because you could get strange behaviors.
Keep testing, and be sure is the right behavior that you need. El mar 30, 2014 10:04 PM, "Slava Bendersky" < volga629@networklab.ca > escribió:
Hello Pedro, I set in main routing section and it I see 200 OK right now. Than you for help.
# account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting }
if (is_method("OPTIONS")) { sl_send_reply("200", "OK"); exit; }
U 2014/03/30 22:32:47.139646 192.168.10.120:5062 -> 192.168.10.120:5060 OPTIONS sip:192.168.10.120 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK6f8354a0. Max-Forwards: 70. From: "asterisk" sip:1300@networklab.loc;tag=as2cbae229. To: sip:192.168.10.120. Contact: < sip:1300@192.168.10.120:5062 >. Call-ID: 456f80c06d85d9b34027ccc533855f72@networklab.loc. CSeq: 102 OPTIONS. User-Agent: Asterisk PBX 12.0.0. Date: Mon, 31 Mar 2014 02:32:47 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Length: 0. .
U 2014/03/30 22:32:47.140301 192.168.10.120:5060 -> 192.168.10.120:5062 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK6f8354a0. From: "asterisk" sip:1300@networklab.loc;tag=as2cbae229. To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.93f4. Call-ID: 456f80c06d85d9b34027ccc533855f72@networklab.loc. CSeq: 102 OPTIONS. Server: kamailio (4.1.2 (x86_64/linux)). Content-Length: 0.
Slava.
From: "Pedro Niño" < nino.pedro@gmail.com > To: "Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org > Sent: Sunday, March 30, 2014 10:04:30 PM Subject: Re: [SR-Users] message 484
Ok, I wonder....
If this is a message you're seeing at the asterisk server, it may be related to the qualify=yes or qualify=Number parameter in the peer, at sip.conf.
If it's right, then you can modify at 2 places: one, by disabling qualify. (qualify=no) at asterisk, or the other by configuring Kamailio to answer a 200 "OK" message when the message comes from the asterisk server.
If not, can you explain when are you seeing such behavior? And can run a 'sngrep host <asterisk_IP>' at the Kamailio server?
Keep me posted. El mar 30, 2014 8:26 PM, "Slava Bendersky" < volga629@networklab.ca > escribió:
<blockquote>
Hello Pedro, So test case is simple. asterisk is play voicemail role and kamailio is gateway. Asterisk peer is setup and working. I think message SIP/2.0 484 Address Incomplete is some from kamailio
When Asterisk send reply back this message show up because is no $rU in header line.
I don't know how to correct it that it will full uri format line.
voice 192.168.10.120 Auto (No) No A 5060 OK (2 ms) 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline]
<--- SIP read from UDP: 192.168.10.120:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK4c787422 From: "asterisk" sip:1300@networklab.loc;tag=as5c659db3 To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2 Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.10.120:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
<-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc' Method: OPTIONS
<--- SIP read from UDP: 192.168.10.120:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK4c787422 From: "asterisk" sip:1300@networklab.loc;tag=as5c659db3 To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2 Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.10.120:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; }
Slava.
From: "Pedro Niño" < nino.pedro@gmail.com > To: "Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org > Sent: Sunday, March 30, 2014 8:30:56 PM Subject: Re: [SR-Users] message 484
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" < volga629@networklab.ca > escribió:
<blockquote>
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP: 192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" sip:1300@networklab.loc;tag=as0a530a8d To: sip:192.168.100.145;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.100.145:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi! I guess this is a poorly configured Asterisk server that has "Allowoverlap" enabled. A 484 is used for overlap dialing. The server says "I need more digits to complete this call".
/O
On 31 Mar 2014, at 02:30, Pedro Niño nino.pedro@gmail.com wrote:
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help.
El mar 29, 2014 1:59 AM, "Slava Bendersky" volga629@networklab.ca escribió: Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP:192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" sip:1300@networklab.loc;tag=as0a530a8d To: sip:192.168.100.145;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.100.145:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello Olle, Overlap is disabled on asterisk. I more wonder about this message.
Mar 31 08:40:20 dsm01 /usr/sbin/kamailio[6101]: WARNING: sanity [sanity.c:833]: check_parse_uris(): sanity_check(): check_parse_uris(): failed to parse From uri
Because from direct connected network, call failing to voicemail.
Slva. ----- Original Message -----
From: "Olle E. Johansson" oej@edvina.net To: "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org Sent: Monday, March 31, 2014 3:33:11 AM Subject: Re: [SR-Users] message 484
Hi! I guess this is a poorly configured Asterisk server that has "Allowoverlap" enabled. A 484 is used for overlap dialing. The server says "I need more digits to complete this call".
/O
On 31 Mar 2014, at 02:30, Pedro Niño < nino.pedro@gmail.com > wrote:
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" < volga629@networklab.ca > escribió:
<blockquote>
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP: 192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" < sip:1300@networklab.loc >;tag=as0a530a8d To: < sip:192.168.100.145 >;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.100.145:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
So, the problem is that calls made from a direct connected user, falls to voicemail? Even if the other user is online?
All the users are on the same asterisk server? Or using a trunk outside?
As a test, tried to register to the asterisk server directly and test the call?
That's why I was asking to elaborate, and show a bit more about the call flow behavior... A small text diagram and desired behavior would be useful
El mar 31, 2014 8:13 AM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Olle, Overlap is disabled on asterisk. I more wonder about this message.
Mar 31 08:40:20 dsm01 /usr/sbin/kamailio[6101]: WARNING: sanity [sanity.c:833]: check_parse_uris(): sanity_check(): check_parse_uris(): failed to parse From uri
Because from direct connected network, call failing to voicemail.
Slva.
*From: *"Olle E. Johansson" oej@edvina.net *To: *"Kamailio (SER) - Users Mailing List" <sr-users@lists.sip-router.org
*Sent: *Monday, March 31, 2014 3:33:11 AM *Subject: *Re: [SR-Users] message 484
Hi! I guess this is a poorly configured Asterisk server that has "Allowoverlap" enabled. A 484 is used for overlap dialing. The server says "I need more digits to complete this call".
/O
On 31 Mar 2014, at 02:30, Pedro Niño nino.pedro@gmail.com wrote:
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP:192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" sip:1300@networklab.loc;tag=as0a530a8d To: sip:192.168.100.145;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.100.145:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello Pedro, Asterisk is used only for voicemail. All extension are terminates and registrar on kamailio gateway. Kamailio used ldap and registration working ok. I can come back to lab soon after meeting and I post some debug.
Slava.
----- Original Message -----
From: "Pedro Niño" nino.pedro@gmail.com To: "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org Sent: Monday, March 31, 2014 9:51:11 AM Subject: Re: [SR-Users] message 484
So, the problem is that calls made from a direct connected user, falls to voicemail? Even if the other user is online?
All the users are on the same asterisk server? Or using a trunk outside?
As a test, tried to register to the asterisk server directly and test the call?
That's why I was asking to elaborate, and show a bit more about the call flow behavior... A small text diagram and desired behavior would be useful
El mar 31, 2014 8:13 AM, "Slava Bendersky" < volga629@networklab.ca > escribió:
Hello Olle, Overlap is disabled on asterisk. I more wonder about this message.
Mar 31 08:40:20 dsm01 /usr/sbin/kamailio[6101]: WARNING: sanity [sanity.c:833]: check_parse_uris(): sanity_check(): check_parse_uris(): failed to parse From uri
Because from direct connected network, call failing to voicemail.
Slva.
From: "Olle E. Johansson" < oej@edvina.net > To: "Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org > Sent: Monday, March 31, 2014 3:33:11 AM Subject: Re: [SR-Users] message 484
Hi! I guess this is a poorly configured Asterisk server that has "Allowoverlap" enabled. A 484 is used for overlap dialing. The server says "I need more digits to complete this call".
/O
On 31 Mar 2014, at 02:30, Pedro Niño < nino.pedro@gmail.com > wrote:
<blockquote>
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" < volga629@networklab.ca > escribió:
<blockquote>
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP: 192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" < sip:1300@networklab.loc >;tag=as0a530a8d To: < sip:192.168.100.145 >;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.100.145:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello Pedro,
Here SDP from asterisk. Asterisk it just don't know where to send traffic. Sip peer on asterisk connects no issue.
[voice] type=peer host=kamailio ip defaultuser=1300 fromuser=1300 user=1300 secret=test permit=local subnet disallow=all allow=ulaw dtmfmode=rfc2833 context=voicemailbox canreinvite=no insecure=port,invite qualify=yes directrtpsetup=no
-- Incorrect password '' for user '1200' (context = default) -- <SIP/1200-00000004> Playing 'vm-incorrect-mailbox.gsm' (language 'en') Retransmitting #9 (no NAT) to 10.237.236.207:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 Via: SIP/2.0/UDP 10.237.236.212:64609;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- Record-Route: sip:10.237.236.207;lr=on From: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 To: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 2 INVITE Server: Asterisk PBX 12.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:120@10.237.236.207:5062 Content-Type: application/sdp Require: timer Content-Length: 183
v=0 o=root 1990993471 1990993471 IN IP4 10.237.236.207 s=Asterisk PBX 12.0.0 c=IN IP4 10.237.236.207 t=0 0 m=audio 15070 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv
--- Retransmitting #10 (no NAT) to 10.237.236.207:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 Via: SIP/2.0/UDP 10.237.236.212:64609;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- Record-Route: sip:10.237.236.207;lr=on From: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 To: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 2 INVITE Server: Asterisk PBX 12.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:120@10.237.236.207:5062 Content-Type: application/sdp Require: timer Content-Length: 183
v=0 o=root 1990993471 1990993471 IN IP4 10.237.236.207 s=Asterisk PBX 12.0.0 c=IN IP4 10.237.236.207 t=0 0 m=audio 15070 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv
--- [Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4259 retrans_pkt: Retransmission timeout reached on transmission YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4288 retrans_pkt: Hanging up call YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Mar 31 14:44:25] WARNING[2801][C-0000000e]: app_voicemail.c:10590 vm_authenticate: Couldn't read username Scheduling destruction of SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' in 32000 ms (Method: INVITE) set_destination: Parsing sip:10.237.236.207;lr=on for address/port to send to set_destination: set destination to 10.237.236.207:5060 Reliably Transmitting (no NAT) to 10.237.236.207:5060: BYE sip:1200@10.237.236.212:64609;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 Route: sip:10.237.236.207;lr=on Max-Forwards: 70 From: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae To: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 102 BYE User-Agent: Asterisk PBX 12.0.0 X-Asterisk-HangupCause: No user responding X-Asterisk-HangupCauseCode: 18 Content-Length: 0
---
<--- SIP read from UDP:10.237.236.207:5060 ---> SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 To: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 From: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 102 BYE Accept-Language: en Content-Length: 0
<-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' Method: INVITE Reliably Transmitting (no NAT) to 10.237.236.207:5060: OPTIONS sip:10.237.236.207 SIP/2.0 Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK51a7f1ef Max-Forwards: 70 From: "asterisk" sip:1300@networklab.loc;tag=as7232ca20 To: sip:10.237.236.207 Contact: sip:1300@10.237.236.207:5062 Call-ID: 46ea55704ee7005705c98d9106904470@networklab.loc CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.0.0 Date: Mon, 31 Mar 2014 18:44:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0
Slava.
----- Original Message -----
From: "Pedro Niño" nino.pedro@gmail.com To: "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org Sent: Monday, March 31, 2014 9:51:11 AM Subject: Re: [SR-Users] message 484
So, the problem is that calls made from a direct connected user, falls to voicemail? Even if the other user is online?
All the users are on the same asterisk server? Or using a trunk outside?
As a test, tried to register to the asterisk server directly and test the call?
That's why I was asking to elaborate, and show a bit more about the call flow behavior... A small text diagram and desired behavior would be useful
El mar 31, 2014 8:13 AM, "Slava Bendersky" < volga629@networklab.ca > escribió:
Hello Olle, Overlap is disabled on asterisk. I more wonder about this message.
Mar 31 08:40:20 dsm01 /usr/sbin/kamailio[6101]: WARNING: sanity [sanity.c:833]: check_parse_uris(): sanity_check(): check_parse_uris(): failed to parse From uri
Because from direct connected network, call failing to voicemail.
Slva.
From: "Olle E. Johansson" < oej@edvina.net > To: "Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org > Sent: Monday, March 31, 2014 3:33:11 AM Subject: Re: [SR-Users] message 484
Hi! I guess this is a poorly configured Asterisk server that has "Allowoverlap" enabled. A 484 is used for overlap dialing. The server says "I need more digits to complete this call".
/O
On 31 Mar 2014, at 02:30, Pedro Niño < nino.pedro@gmail.com > wrote:
<blockquote>
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" < volga629@networklab.ca > escribió:
<blockquote>
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP: 192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" < sip:1300@networklab.loc >;tag=as0a530a8d To: < sip:192.168.100.145 >;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.100.145:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Sorry, I was out for a while. Still have this issue?
From what I am seeing, asterisk is expecting for the password. Is the
voicemail configured ? Check username and password.
Somewhere there it says that couldn't read username and password from the voicemail. Have the extensions.conf at asterisk dialplan configured properly? El mar 31, 2014 2:25 PM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Pedro,
Here SDP from asterisk. Asterisk it just don't know where to send traffic. Sip peer on asterisk connects no issue.
[voice] type=peer host=kamailio ip defaultuser=1300 fromuser=1300 user=1300 secret=test permit=local subnet disallow=all allow=ulaw dtmfmode=rfc2833 context=voicemailbox canreinvite=no insecure=port,invite qualify=yes directrtpsetup=no
-- Incorrect password '' for user '1200' (context = default) -- <SIP/1200-00000004> Playing 'vm-incorrect-mailbox.gsm' (language'en') Retransmitting #9 (no NAT) to 10.237.236.207:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 Via: SIP/2.0/UDP 10.237.236.212:64609 ;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- Record-Route: sip:10.237.236.207;lr=on From: "Slava Bendersky"sip:1200@networklab.loc ;transport=UDP;tag=6358d712 To: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 2 INVITE Server: Asterisk PBX 12.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:120@10.237.236.207:5062 Content-Type: application/sdp Require: timer Content-Length: 183
v=0 o=root 1990993471 1990993471 IN IP4 10.237.236.207 s=Asterisk PBX 12.0.0 c=IN IP4 10.237.236.207 t=0 0 m=audio 15070 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv
Retransmitting #10 (no NAT) to 10.237.236.207:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 Via: SIP/2.0/UDP 10.237.236.212:64609 ;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- Record-Route: sip:10.237.236.207;lr=on From: "Slava Bendersky"sip:1200@networklab.loc ;transport=UDP;tag=6358d712 To: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 2 INVITE Server: Asterisk PBX 12.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:120@10.237.236.207:5062 Content-Type: application/sdp Require: timer Content-Length: 183
v=0 o=root 1990993471 1990993471 IN IP4 10.237.236.207 s=Asterisk PBX 12.0.0 c=IN IP4 10.237.236.207 t=0 0 m=audio 15070 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv
[Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4259 retrans_pkt: Retransmission timeout reached on transmission YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4288 retrans_pkt: Hanging up call YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Mar 31 14:44:25] WARNING[2801][C-0000000e]: app_voicemail.c:10590 vm_authenticate: Couldn't read username Scheduling destruction of SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' in 32000 ms (Method: INVITE) set_destination: Parsing sip:10.237.236.207;lr=on for address/port to send to set_destination: set destination to 10.237.236.207:5060 Reliably Transmitting (no NAT) to 10.237.236.207:5060: BYE sip:1200@10.237.236.212:64609;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 Route: sip:10.237.236.207;lr=on Max-Forwards: 70 From: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae To: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 102 BYE User-Agent: Asterisk PBX 12.0.0 X-Asterisk-HangupCause: No user responding X-Asterisk-HangupCauseCode: 18 Content-Length: 0
<--- SIP read from UDP:10.237.236.207:5060 ---> SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 To: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 From: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 102 BYE Accept-Language: en Content-Length: 0
<-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' Method: INVITE Reliably Transmitting (no NAT) to 10.237.236.207:5060: OPTIONS sip:10.237.236.207 SIP/2.0 Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK51a7f1ef Max-Forwards: 70 From: "asterisk" sip:1300@networklab.loc;tag=as7232ca20 To: sip:10.237.236.207 Contact: sip:1300@10.237.236.207:5062 Call-ID: 46ea55704ee7005705c98d9106904470@networklab.loc CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.0.0 Date: Mon, 31 Mar 2014 18:44:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0
Slava.
*From: *"Pedro Niño" nino.pedro@gmail.com *To: *"Kamailio (SER) - Users Mailing List" <sr-users@lists.sip-router.org
*Sent: *Monday, March 31, 2014 9:51:11 AM *Subject: *Re: [SR-Users] message 484
So, the problem is that calls made from a direct connected user, falls to voicemail? Even if the other user is online?
All the users are on the same asterisk server? Or using a trunk outside?
As a test, tried to register to the asterisk server directly and test the call?
That's why I was asking to elaborate, and show a bit more about the call flow behavior... A small text diagram and desired behavior would be useful
El mar 31, 2014 8:13 AM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Olle, Overlap is disabled on asterisk. I more wonder about this message.
Mar 31 08:40:20 dsm01 /usr/sbin/kamailio[6101]: WARNING: sanity [sanity.c:833]: check_parse_uris(): sanity_check(): check_parse_uris(): failed to parse From uri
Because from direct connected network, call failing to voicemail.
Slva.
*From: *"Olle E. Johansson" oej@edvina.net *To: *"Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org> *Sent: *Monday, March 31, 2014 3:33:11 AM *Subject: *Re: [SR-Users] message 484
Hi! I guess this is a poorly configured Asterisk server that has "Allowoverlap" enabled. A 484 is used for overlap dialing. The server says "I need more digits to complete this call".
/O
On 31 Mar 2014, at 02:30, Pedro Niño nino.pedro@gmail.com wrote:
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP:192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" sip:1300@networklab.loc;tag=as0a530a8d To: sip:192.168.100.145;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.100.145:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
I think you should remove this section: or comment it, its behavior is not the one we want at this moment.
-------
if (is_method("OPTIONS") || allow_trusted("$si","$(proto)")) { #if (is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "OK"); }
----- El abr 1, 2014 7:58 PM, "Pedro Niño" nino.pedro@gmail.com escribió:
Sorry, I was out for a while. Still have this issue?
From what I am seeing, asterisk is expecting for the password. Is the voicemail configured ? Check username and password.
Somewhere there it says that couldn't read username and password from the voicemail. Have the extensions.conf at asterisk dialplan configured properly? El mar 31, 2014 2:25 PM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Pedro,
Here SDP from asterisk. Asterisk it just don't know where to send traffic. Sip peer on asterisk connects no issue.
[voice] type=peer host=kamailio ip defaultuser=1300 fromuser=1300 user=1300 secret=test permit=local subnet disallow=all allow=ulaw dtmfmode=rfc2833 context=voicemailbox canreinvite=no insecure=port,invite qualify=yes directrtpsetup=no
-- Incorrect password '' for user '1200' (context = default) -- <SIP/1200-00000004> Playing 'vm-incorrect-mailbox.gsm' (language'en') Retransmitting #9 (no NAT) to 10.237.236.207:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 Via: SIP/2.0/UDP 10.237.236.212:64609 ;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- Record-Route: sip:10.237.236.207;lr=on From: "Slava Bendersky"sip:1200@networklab.loc ;transport=UDP;tag=6358d712 To: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 2 INVITE Server: Asterisk PBX 12.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:120@10.237.236.207:5062 Content-Type: application/sdp Require: timer Content-Length: 183
v=0 o=root 1990993471 1990993471 IN IP4 10.237.236.207 s=Asterisk PBX 12.0.0 c=IN IP4 10.237.236.207 t=0 0 m=audio 15070 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv
Retransmitting #10 (no NAT) to 10.237.236.207:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 Via: SIP/2.0/UDP 10.237.236.212:64609 ;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- Record-Route: sip:10.237.236.207;lr=on From: "Slava Bendersky"sip:1200@networklab.loc ;transport=UDP;tag=6358d712 To: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 2 INVITE Server: Asterisk PBX 12.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:120@10.237.236.207:5062 Content-Type: application/sdp Require: timer Content-Length: 183
v=0 o=root 1990993471 1990993471 IN IP4 10.237.236.207 s=Asterisk PBX 12.0.0 c=IN IP4 10.237.236.207 t=0 0 m=audio 15070 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv
[Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4259 retrans_pkt: Retransmission timeout reached on transmission YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4288 retrans_pkt: Hanging up call YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Mar 31 14:44:25] WARNING[2801][C-0000000e]: app_voicemail.c:10590 vm_authenticate: Couldn't read username Scheduling destruction of SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' in 32000 ms (Method: INVITE) set_destination: Parsing sip:10.237.236.207;lr=on for address/port to send to set_destination: set destination to 10.237.236.207:5060 Reliably Transmitting (no NAT) to 10.237.236.207:5060: BYE sip:1200@10.237.236.212:64609;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 Route: sip:10.237.236.207;lr=on Max-Forwards: 70 From: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae To: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 102 BYE User-Agent: Asterisk PBX 12.0.0 X-Asterisk-HangupCause: No user responding X-Asterisk-HangupCauseCode: 18 Content-Length: 0
<--- SIP read from UDP:10.237.236.207:5060 ---> SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 To: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 From: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 102 BYE Accept-Language: en Content-Length: 0
<-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' Method: INVITE Reliably Transmitting (no NAT) to 10.237.236.207:5060: OPTIONS sip:10.237.236.207 SIP/2.0 Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK51a7f1ef Max-Forwards: 70 From: "asterisk" sip:1300@networklab.loc;tag=as7232ca20 To: sip:10.237.236.207 Contact: sip:1300@10.237.236.207:5062 Call-ID: 46ea55704ee7005705c98d9106904470@networklab.loc CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.0.0 Date: Mon, 31 Mar 2014 18:44:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0
Slava.
*From: *"Pedro Niño" nino.pedro@gmail.com *To: *"Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org> *Sent: *Monday, March 31, 2014 9:51:11 AM *Subject: *Re: [SR-Users] message 484
So, the problem is that calls made from a direct connected user, falls to voicemail? Even if the other user is online?
All the users are on the same asterisk server? Or using a trunk outside?
As a test, tried to register to the asterisk server directly and test the call?
That's why I was asking to elaborate, and show a bit more about the call flow behavior... A small text diagram and desired behavior would be useful
El mar 31, 2014 8:13 AM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Olle, Overlap is disabled on asterisk. I more wonder about this message.
Mar 31 08:40:20 dsm01 /usr/sbin/kamailio[6101]: WARNING: sanity [sanity.c:833]: check_parse_uris(): sanity_check(): check_parse_uris(): failed to parse From uri
Because from direct connected network, call failing to voicemail.
Slva.
*From: *"Olle E. Johansson" oej@edvina.net *To: *"Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org> *Sent: *Monday, March 31, 2014 3:33:11 AM *Subject: *Re: [SR-Users] message 484
Hi! I guess this is a poorly configured Asterisk server that has "Allowoverlap" enabled. A 484 is used for overlap dialing. The server says "I need more digits to complete this call".
/O
On 31 Mar 2014, at 02:30, Pedro Niño nino.pedro@gmail.com wrote:
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP:192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" sip:1300@networklab.loc;tag=as0a530a8d To: sip:192.168.100.145;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.100.145:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello Pedro, I just come back on line. If i remove this line I start getting
----- Original Message -----
From: "Pedro Niño" nino.pedro@gmail.com To: "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org Sent: Tuesday, April 1, 2014 8:40:58 PM Subject: Re: [SR-Users] message 484
I think you should remove this section: or comment it, its behavior is not the one we want at this moment.
-------
if (is_method("OPTIONS") || allow_trusted("$si","$(proto)")) { #if (is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "OK"); }
----- El abr 1, 2014 7:58 PM, "Pedro Niño" < nino.pedro@gmail.com > escribió:
Sorry, I was out for a while. Still have this issue?
From what I am seeing, asterisk is expecting for the password. Is the voicemail configured ? Check username and password.
Somewhere there it says that couldn't read username and password from the voicemail. Have the extensions.conf at asterisk dialplan configured properly? El mar 31, 2014 2:25 PM, "Slava Bendersky" < volga629@networklab.ca > escribió:
<blockquote>
Hello Pedro,
Here SDP from asterisk. Asterisk it just don't know where to send traffic. Sip peer on asterisk connects no issue.
[voice] type=peer host=kamailio ip defaultuser=1300 fromuser=1300 user=1300 secret=test permit=local subnet disallow=all allow=ulaw dtmfmode=rfc2833 context=voicemailbox canreinvite=no insecure=port,invite qualify=yes directrtpsetup=no
-- Incorrect password '' for user '1200' (context = default) -- <SIP/1200-00000004> Playing 'vm-incorrect-mailbox.gsm' (language 'en') Retransmitting #9 (no NAT) to 10.237.236.207:5060 : SIP/2.0 200 OK Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 Via: SIP/2.0/UDP 10.237.236.212:64609;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- Record-Route: sip:10.237.236.207;lr=on From: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 To: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 2 INVITE Server: Asterisk PBX 12.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: < sip:120@10.237.236.207:5062 > Content-Type: application/sdp Require: timer Content-Length: 183
v=0 o=root 1990993471 1990993471 IN IP4 10.237.236.207 s=Asterisk PBX 12.0.0 c=IN IP4 10.237.236.207 t=0 0 m=audio 15070 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv
--- Retransmitting #10 (no NAT) to 10.237.236.207:5060 : SIP/2.0 200 OK Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 Via: SIP/2.0/UDP 10.237.236.212:64609;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- Record-Route: sip:10.237.236.207;lr=on From: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 To: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 2 INVITE Server: Asterisk PBX 12.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: < sip:120@10.237.236.207:5062 > Content-Type: application/sdp Require: timer Content-Length: 183
v=0 o=root 1990993471 1990993471 IN IP4 10.237.236.207 s=Asterisk PBX 12.0.0 c=IN IP4 10.237.236.207 t=0 0 m=audio 15070 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv
--- [Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4259 retrans_pkt: Retransmission timeout reached on transmission YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4288 retrans_pkt: Hanging up call YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions ). [Mar 31 14:44:25] WARNING[2801][C-0000000e]: app_voicemail.c:10590 vm_authenticate: Couldn't read username Scheduling destruction of SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' in 32000 ms (Method: INVITE) set_destination: Parsing sip:10.237.236.207;lr=on for address/port to send to set_destination: set destination to 10.237.236.207:5060 Reliably Transmitting (no NAT) to 10.237.236.207:5060 : BYE sip:1200@10.237.236.212:64609;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 Route: sip:10.237.236.207;lr=on Max-Forwards: 70 From: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae To: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 102 BYE User-Agent: Asterisk PBX 12.0.0 X-Asterisk-HangupCause: No user responding X-Asterisk-HangupCauseCode: 18 Content-Length: 0
---
<--- SIP read from UDP: 10.237.236.207:5060 ---> SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 To: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 From: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 102 BYE Accept-Language: en Content-Length: 0
<-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' Method: INVITE Reliably Transmitting (no NAT) to 10.237.236.207:5060 : OPTIONS sip:10.237.236.207 SIP/2.0 Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK51a7f1ef Max-Forwards: 70 From: "asterisk" sip:1300@networklab.loc;tag=as7232ca20 To: sip:10.237.236.207 Contact: < sip:1300@10.237.236.207:5062 > Call-ID: 46ea55704ee7005705c98d9106904470@networklab.loc CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.0.0 Date: Mon, 31 Mar 2014 18:44:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0
Slava.
From: "Pedro Niño" < nino.pedro@gmail.com > To: "Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org > Sent: Monday, March 31, 2014 9:51:11 AM Subject: Re: [SR-Users] message 484
So, the problem is that calls made from a direct connected user, falls to voicemail? Even if the other user is online?
All the users are on the same asterisk server? Or using a trunk outside?
As a test, tried to register to the asterisk server directly and test the call?
That's why I was asking to elaborate, and show a bit more about the call flow behavior... A small text diagram and desired behavior would be useful
El mar 31, 2014 8:13 AM, "Slava Bendersky" < volga629@networklab.ca > escribió:
<blockquote>
Hello Olle, Overlap is disabled on asterisk. I more wonder about this message.
Mar 31 08:40:20 dsm01 /usr/sbin/kamailio[6101]: WARNING: sanity [sanity.c:833]: check_parse_uris(): sanity_check(): check_parse_uris(): failed to parse From uri
Because from direct connected network, call failing to voicemail.
Slva.
From: "Olle E. Johansson" < oej@edvina.net > To: "Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org > Sent: Monday, March 31, 2014 3:33:11 AM Subject: Re: [SR-Users] message 484
Hi! I guess this is a poorly configured Asterisk server that has "Allowoverlap" enabled. A 484 is used for overlap dialing. The server says "I need more digits to complete this call".
/O
On 31 Mar 2014, at 02:30, Pedro Niño < nino.pedro@gmail.com > wrote:
<blockquote>
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" < volga629@networklab.ca > escribió:
<blockquote>
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP: 192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" < sip:1300@networklab.loc >;tag=as0a530a8d To: < sip:192.168.100.145 >;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.100.145:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Getting....? El abr 7, 2014 1:21 PM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Pedro, I just come back on line. If i remove this line I start getting
*From: *"Pedro Niño" nino.pedro@gmail.com *To: *"Kamailio (SER) - Users Mailing List" <sr-users@lists.sip-router.org
*Sent: *Tuesday, April 1, 2014 8:40:58 PM *Subject: *Re: [SR-Users] message 484
I think you should remove this section: or comment it, its behavior is not the one we want at this moment.
if (is_method("OPTIONS") || allow_trusted("$si","$(proto)")) { #if (is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "OK"); }
El abr 1, 2014 7:58 PM, "Pedro Niño" nino.pedro@gmail.com escribió:
Sorry, I was out for a while. Still have this issue?
From what I am seeing, asterisk is expecting for the password. Is the voicemail configured ? Check username and password.
Somewhere there it says that couldn't read username and password from the voicemail. Have the extensions.conf at asterisk dialplan configured properly? El mar 31, 2014 2:25 PM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Pedro,
Here SDP from asterisk. Asterisk it just don't know where to send traffic. Sip peer on asterisk connects no issue.
[voice] type=peer host=kamailio ip defaultuser=1300 fromuser=1300 user=1300 secret=test permit=local subnet disallow=all allow=ulaw dtmfmode=rfc2833 context=voicemailbox canreinvite=no insecure=port,invite qualify=yes directrtpsetup=no
-- Incorrect password '' for user '1200' (context = default) -- <SIP/1200-00000004> Playing 'vm-incorrect-mailbox.gsm' (language'en') Retransmitting #9 (no NAT) to 10.237.236.207:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 Via: SIP/2.0/UDP 10.237.236.212:64609 ;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- Record-Route: sip:10.237.236.207;lr=on From: "Slava Bendersky"sip:1200@networklab.loc ;transport=UDP;tag=6358d712 To: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 2 INVITE Server: Asterisk PBX 12.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:120@10.237.236.207:5062 Content-Type: application/sdp Require: timer Content-Length: 183
v=0 o=root 1990993471 1990993471 IN IP4 10.237.236.207 s=Asterisk PBX 12.0.0 c=IN IP4 10.237.236.207 t=0 0 m=audio 15070 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv
Retransmitting #10 (no NAT) to 10.237.236.207:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 Via: SIP/2.0/UDP 10.237.236.212:64609 ;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- Record-Route: sip:10.237.236.207;lr=on From: "Slava Bendersky"sip:1200@networklab.loc ;transport=UDP;tag=6358d712 To: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 2 INVITE Server: Asterisk PBX 12.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:120@10.237.236.207:5062 Content-Type: application/sdp Require: timer Content-Length: 183
v=0 o=root 1990993471 1990993471 IN IP4 10.237.236.207 s=Asterisk PBX 12.0.0 c=IN IP4 10.237.236.207 t=0 0 m=audio 15070 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv
[Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4259 retrans_pkt: Retransmission timeout reached on transmission YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4288 retrans_pkt: Hanging up call YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Mar 31 14:44:25] WARNING[2801][C-0000000e]: app_voicemail.c:10590 vm_authenticate: Couldn't read username Scheduling destruction of SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' in 32000 ms (Method: INVITE) set_destination: Parsing sip:10.237.236.207;lr=on for address/port to send to set_destination: set destination to 10.237.236.207:5060 Reliably Transmitting (no NAT) to 10.237.236.207:5060: BYE sip:1200@10.237.236.212:64609;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 Route: sip:10.237.236.207;lr=on Max-Forwards: 70 From: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae To: "Slava Bendersky"sip:1200@networklab.loc ;transport=UDP;tag=6358d712 Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 102 BYE User-Agent: Asterisk PBX 12.0.0 X-Asterisk-HangupCause: No user responding X-Asterisk-HangupCauseCode: 18 Content-Length: 0
<--- SIP read from UDP:10.237.236.207:5060 ---> SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 To: "Slava Bendersky"sip:1200@networklab.loc ;transport=UDP;tag=6358d712 From: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 102 BYE Accept-Language: en Content-Length: 0
<-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' Method: INVITE Reliably Transmitting (no NAT) to 10.237.236.207:5060: OPTIONS sip:10.237.236.207 SIP/2.0 Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK51a7f1ef Max-Forwards: 70 From: "asterisk" sip:1300@networklab.loc;tag=as7232ca20 To: sip:10.237.236.207 Contact: sip:1300@10.237.236.207:5062 Call-ID: 46ea55704ee7005705c98d9106904470@networklab.loc CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.0.0 Date: Mon, 31 Mar 2014 18:44:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0
Slava.
*From: *"Pedro Niño" nino.pedro@gmail.com *To: *"Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org> *Sent: *Monday, March 31, 2014 9:51:11 AM *Subject: *Re: [SR-Users] message 484
So, the problem is that calls made from a direct connected user, falls to voicemail? Even if the other user is online?
All the users are on the same asterisk server? Or using a trunk outside?
As a test, tried to register to the asterisk server directly and test the call?
That's why I was asking to elaborate, and show a bit more about the call flow behavior... A small text diagram and desired behavior would be useful
El mar 31, 2014 8:13 AM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Olle, Overlap is disabled on asterisk. I more wonder about this message.
Mar 31 08:40:20 dsm01 /usr/sbin/kamailio[6101]: WARNING: sanity [sanity.c:833]: check_parse_uris(): sanity_check(): check_parse_uris(): failed to parse From uri
Because from direct connected network, call failing to voicemail.
Slva.
*From: *"Olle E. Johansson" oej@edvina.net *To: *"Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org> *Sent: *Monday, March 31, 2014 3:33:11 AM *Subject: *Re: [SR-Users] message 484
Hi! I guess this is a poorly configured Asterisk server that has "Allowoverlap" enabled. A 484 is used for overlap dialing. The server says "I need more digits to complete this call".
/O
On 31 Mar 2014, at 02:30, Pedro Niño nino.pedro@gmail.com wrote:
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" volga629@networklab.ca escribió:
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP:192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" sip:1300@networklab.loc;tag=as0a530a8d To: sip:192.168.100.145;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: sip:1300@192.168.100.145:5062;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello Pedro, I meant when removing option line kamailio starts getting message 484.
Slava.
----- Original Message -----
From: "Pedro Niño" nino.pedro@gmail.com To: "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org Sent: Wednesday, April 9, 2014 2:35:01 PM Subject: Re: [SR-Users] message 484
Getting....? El abr 7, 2014 1:21 PM, "Slava Bendersky" < volga629@networklab.ca > escribió:
Hello Pedro, I just come back on line. If i remove this line I start getting
From: "Pedro Niño" < nino.pedro@gmail.com > To: "Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org > Sent: Tuesday, April 1, 2014 8:40:58 PM Subject: Re: [SR-Users] message 484
I think you should remove this section: or comment it, its behavior is not the one we want at this moment.
-------
if (is_method("OPTIONS") || allow_trusted("$si","$(proto)")) { #if (is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "OK"); }
----- El abr 1, 2014 7:58 PM, "Pedro Niño" < nino.pedro@gmail.com > escribió:
<blockquote>
Sorry, I was out for a while. Still have this issue?
From what I am seeing, asterisk is expecting for the password. Is the voicemail configured ? Check username and password.
Somewhere there it says that couldn't read username and password from the voicemail. Have the extensions.conf at asterisk dialplan configured properly? El mar 31, 2014 2:25 PM, "Slava Bendersky" < volga629@networklab.ca > escribió:
<blockquote>
Hello Pedro,
Here SDP from asterisk. Asterisk it just don't know where to send traffic. Sip peer on asterisk connects no issue.
[voice] type=peer host=kamailio ip defaultuser=1300 fromuser=1300 user=1300 secret=test permit=local subnet disallow=all allow=ulaw dtmfmode=rfc2833 context=voicemailbox canreinvite=no insecure=port,invite qualify=yes directrtpsetup=no
-- Incorrect password '' for user '1200' (context = default) -- <SIP/1200-00000004> Playing 'vm-incorrect-mailbox.gsm' (language 'en') Retransmitting #9 (no NAT) to 10.237.236.207:5060 : SIP/2.0 200 OK Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 Via: SIP/2.0/UDP 10.237.236.212:64609;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- Record-Route: sip:10.237.236.207;lr=on From: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 To: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 2 INVITE Server: Asterisk PBX 12.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: < sip:120@10.237.236.207:5062 > Content-Type: application/sdp Require: timer Content-Length: 183
v=0 o=root 1990993471 1990993471 IN IP4 10.237.236.207 s=Asterisk PBX 12.0.0 c=IN IP4 10.237.236.207 t=0 0 m=audio 15070 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv
--- Retransmitting #10 (no NAT) to 10.237.236.207:5060 : SIP/2.0 200 OK Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 Via: SIP/2.0/UDP 10.237.236.212:64609;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- Record-Route: sip:10.237.236.207;lr=on From: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 To: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 2 INVITE Server: Asterisk PBX 12.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: < sip:120@10.237.236.207:5062 > Content-Type: application/sdp Require: timer Content-Length: 183
v=0 o=root 1990993471 1990993471 IN IP4 10.237.236.207 s=Asterisk PBX 12.0.0 c=IN IP4 10.237.236.207 t=0 0 m=audio 15070 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv
--- [Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4259 retrans_pkt: Retransmission timeout reached on transmission YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4288 retrans_pkt: Hanging up call YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions ). [Mar 31 14:44:25] WARNING[2801][C-0000000e]: app_voicemail.c:10590 vm_authenticate: Couldn't read username Scheduling destruction of SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' in 32000 ms (Method: INVITE) set_destination: Parsing sip:10.237.236.207;lr=on for address/port to send to set_destination: set destination to 10.237.236.207:5060 Reliably Transmitting (no NAT) to 10.237.236.207:5060 : BYE sip:1200@10.237.236.212:64609;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 Route: sip:10.237.236.207;lr=on Max-Forwards: 70 From: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae To: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 102 BYE User-Agent: Asterisk PBX 12.0.0 X-Asterisk-HangupCause: No user responding X-Asterisk-HangupCauseCode: 18 Content-Length: 0
---
<--- SIP read from UDP: 10.237.236.207:5060 ---> SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 To: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 From: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 102 BYE Accept-Language: en Content-Length: 0
<-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' Method: INVITE Reliably Transmitting (no NAT) to 10.237.236.207:5060 : OPTIONS sip:10.237.236.207 SIP/2.0 Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK51a7f1ef Max-Forwards: 70 From: "asterisk" sip:1300@networklab.loc;tag=as7232ca20 To: sip:10.237.236.207 Contact: < sip:1300@10.237.236.207:5062 > Call-ID: 46ea55704ee7005705c98d9106904470@networklab.loc CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.0.0 Date: Mon, 31 Mar 2014 18:44:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0
Slava.
From: "Pedro Niño" < nino.pedro@gmail.com > To: "Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org > Sent: Monday, March 31, 2014 9:51:11 AM Subject: Re: [SR-Users] message 484
So, the problem is that calls made from a direct connected user, falls to voicemail? Even if the other user is online?
All the users are on the same asterisk server? Or using a trunk outside?
As a test, tried to register to the asterisk server directly and test the call?
That's why I was asking to elaborate, and show a bit more about the call flow behavior... A small text diagram and desired behavior would be useful
El mar 31, 2014 8:13 AM, "Slava Bendersky" < volga629@networklab.ca > escribió:
<blockquote>
Hello Olle, Overlap is disabled on asterisk. I more wonder about this message.
Mar 31 08:40:20 dsm01 /usr/sbin/kamailio[6101]: WARNING: sanity [sanity.c:833]: check_parse_uris(): sanity_check(): check_parse_uris(): failed to parse From uri
Because from direct connected network, call failing to voicemail.
Slva.
From: "Olle E. Johansson" < oej@edvina.net > To: "Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org > Sent: Monday, March 31, 2014 3:33:11 AM Subject: Re: [SR-Users] message 484
Hi! I guess this is a poorly configured Asterisk server that has "Allowoverlap" enabled. A 484 is used for overlap dialing. The server says "I need more digits to complete this call".
/O
On 31 Mar 2014, at 02:30, Pedro Niño < nino.pedro@gmail.com > wrote:
<blockquote>
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" < volga629@networklab.ca > escribió:
<blockquote>
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP: 192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" < sip:1300@networklab.loc >;tag=as0a530a8d To: < sip:192.168.100.145 >;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.100.145:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello Pedro, When I removing this line I starts getting
"484","Address Incomplete"
I tried enable rtp debug on asterisk and look like all re transmissions cause by reinvite.
Slava.
----- Original Message -----
From: "Pedro Niño" nino.pedro@gmail.com To: "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org Sent: Tuesday, April 1, 2014 8:40:58 PM Subject: Re: [SR-Users] message 484
I think you should remove this section: or comment it, its behavior is not the one we want at this moment.
-------
if (is_method("OPTIONS") || allow_trusted("$si","$(proto)")) { #if (is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "OK"); }
----- El abr 1, 2014 7:58 PM, "Pedro Niño" < nino.pedro@gmail.com > escribió:
Sorry, I was out for a while. Still have this issue?
From what I am seeing, asterisk is expecting for the password. Is the voicemail configured ? Check username and password.
Somewhere there it says that couldn't read username and password from the voicemail. Have the extensions.conf at asterisk dialplan configured properly? El mar 31, 2014 2:25 PM, "Slava Bendersky" < volga629@networklab.ca > escribió:
<blockquote>
Hello Pedro,
Here SDP from asterisk. Asterisk it just don't know where to send traffic. Sip peer on asterisk connects no issue.
[voice] type=peer host=kamailio ip defaultuser=1300 fromuser=1300 user=1300 secret=test permit=local subnet disallow=all allow=ulaw dtmfmode=rfc2833 context=voicemailbox canreinvite=no insecure=port,invite qualify=yes directrtpsetup=no
-- Incorrect password '' for user '1200' (context = default) -- <SIP/1200-00000004> Playing 'vm-incorrect-mailbox.gsm' (language 'en') Retransmitting #9 (no NAT) to 10.237.236.207:5060 : SIP/2.0 200 OK Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 Via: SIP/2.0/UDP 10.237.236.212:64609;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- Record-Route: sip:10.237.236.207;lr=on From: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 To: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 2 INVITE Server: Asterisk PBX 12.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: < sip:120@10.237.236.207:5062 > Content-Type: application/sdp Require: timer Content-Length: 183
v=0 o=root 1990993471 1990993471 IN IP4 10.237.236.207 s=Asterisk PBX 12.0.0 c=IN IP4 10.237.236.207 t=0 0 m=audio 15070 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv
--- Retransmitting #10 (no NAT) to 10.237.236.207:5060 : SIP/2.0 200 OK Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 Via: SIP/2.0/UDP 10.237.236.212:64609;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- Record-Route: sip:10.237.236.207;lr=on From: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 To: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 2 INVITE Server: Asterisk PBX 12.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: < sip:120@10.237.236.207:5062 > Content-Type: application/sdp Require: timer Content-Length: 183
v=0 o=root 1990993471 1990993471 IN IP4 10.237.236.207 s=Asterisk PBX 12.0.0 c=IN IP4 10.237.236.207 t=0 0 m=audio 15070 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv
--- [Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4259 retrans_pkt: Retransmission timeout reached on transmission YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4288 retrans_pkt: Hanging up call YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions ). [Mar 31 14:44:25] WARNING[2801][C-0000000e]: app_voicemail.c:10590 vm_authenticate: Couldn't read username Scheduling destruction of SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' in 32000 ms (Method: INVITE) set_destination: Parsing sip:10.237.236.207;lr=on for address/port to send to set_destination: set destination to 10.237.236.207:5060 Reliably Transmitting (no NAT) to 10.237.236.207:5060 : BYE sip:1200@10.237.236.212:64609;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 Route: sip:10.237.236.207;lr=on Max-Forwards: 70 From: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae To: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 102 BYE User-Agent: Asterisk PBX 12.0.0 X-Asterisk-HangupCause: No user responding X-Asterisk-HangupCauseCode: 18 Content-Length: 0
---
<--- SIP read from UDP: 10.237.236.207:5060 ---> SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 To: "Slava Bendersky"sip:1200@networklab.loc;transport=UDP;tag=6358d712 From: sip:120@networklab.loc;transport=UDP;tag=as3b53c4ae Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. CSeq: 102 BYE Accept-Language: en Content-Length: 0
<-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' Method: INVITE Reliably Transmitting (no NAT) to 10.237.236.207:5060 : OPTIONS sip:10.237.236.207 SIP/2.0 Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK51a7f1ef Max-Forwards: 70 From: "asterisk" sip:1300@networklab.loc;tag=as7232ca20 To: sip:10.237.236.207 Contact: < sip:1300@10.237.236.207:5062 > Call-ID: 46ea55704ee7005705c98d9106904470@networklab.loc CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.0.0 Date: Mon, 31 Mar 2014 18:44:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0
Slava.
From: "Pedro Niño" < nino.pedro@gmail.com > To: "Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org > Sent: Monday, March 31, 2014 9:51:11 AM Subject: Re: [SR-Users] message 484
So, the problem is that calls made from a direct connected user, falls to voicemail? Even if the other user is online?
All the users are on the same asterisk server? Or using a trunk outside?
As a test, tried to register to the asterisk server directly and test the call?
That's why I was asking to elaborate, and show a bit more about the call flow behavior... A small text diagram and desired behavior would be useful
El mar 31, 2014 8:13 AM, "Slava Bendersky" < volga629@networklab.ca > escribió:
<blockquote>
Hello Olle, Overlap is disabled on asterisk. I more wonder about this message.
Mar 31 08:40:20 dsm01 /usr/sbin/kamailio[6101]: WARNING: sanity [sanity.c:833]: check_parse_uris(): sanity_check(): check_parse_uris(): failed to parse From uri
Because from direct connected network, call failing to voicemail.
Slva.
From: "Olle E. Johansson" < oej@edvina.net > To: "Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org > Sent: Monday, March 31, 2014 3:33:11 AM Subject: Re: [SR-Users] message 484
Hi! I guess this is a poorly configured Asterisk server that has "Allowoverlap" enabled. A 484 is used for overlap dialing. The server says "I need more digits to complete this call".
/O
On 31 Mar 2014, at 02:30, Pedro Niño < nino.pedro@gmail.com > wrote:
<blockquote>
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" < volga629@networklab.ca > escribió:
<blockquote>
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP: 192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" < sip:1300@networklab.loc >;tag=as0a530a8d To: < sip:192.168.100.145 >;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.100.145:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello Olle, This only one place. I don't see xlog only function error.
if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; }
Slava.
----- Original Message -----
From: "Olle E. Johansson" oej@edvina.net To: "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org Sent: Monday, March 31, 2014 3:33:11 AM Subject: Re: [SR-Users] message 484
Hi! I guess this is a poorly configured Asterisk server that has "Allowoverlap" enabled. A 484 is used for overlap dialing. The server says "I need more digits to complete this call".
/O
On 31 Mar 2014, at 02:30, Pedro Niño < nino.pedro@gmail.com > wrote:
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" < volga629@networklab.ca > escribió:
<blockquote>
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP: 192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" < sip:1300@networklab.loc >;tag=as0a530a8d To: < sip:192.168.100.145 >;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.100.145:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Olle E. Johansson -
Pedro Niño -
Slava Bendersky