[SR-Users] Possible causes of calls being terminated (ACK packet never received)
Alex Villacís Lasso writes: > I am trying to explain the situation to our carrier, but I want to > rule out possible misconfigurations on our side. Are there common > misconfigurations that produce the symptoms described here? Are there > any issues evident from the attached traffic? 200 ok matches invite if from tag matches, call id matches and cseq is the same as in invite. -- juha ___ 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
[SR-Users] Possible causes of calls being terminated (ACK packet never received)
I am trying to diagnose a SIP issue between our carrier and our network. The carrier has a CARRIER_IP and a different CARRIER_MEDIA_IP, and it submits an INVITE packet to MY_PUBLIC_IP using MY_DID. The firewall at our public IP (where the attached traffic sample was taken) redirects it to a particular Kamailio server at 192.168.10.10 inside the LAN, and it in turn routes it to the Asterisk instance in localhost. The issue is that the INVITE is received, then the SIP/2.0 100 Trying and then SIP/2.0 200 OK are routed back (or so I think), and then the expected ACK from is never received, even though the caller already hears the media from the Asterisk IVR. After a timeout, our Asterisk closes the call, in the middle of the conversation. I am trying to explain the situation to our carrier, but I want to rule out possible misconfigurations on our side. Are there common misconfigurations that produce the symptoms described here? Are there any issues evident from the attached traffic? 10:55:02.721271 IP CARRIER_IP.5060 > MY_PUBLIC_IP.5060: SIP, length: 870 EH@.&~.).n..INVITE sip:MY_DID@MY_PUBLIC_IP:5060 SIP/2.0 Max-Forwards: 69 Session-Expires: 3600;Refresher=uac Supported: timer To: "unknown" From: "WIRELESS CALLER" ;tag=3609590102-643763 P-Asserted-Identity:"WIRELESS CALLER" Call-ID: 90553-3609590102-643720@msw1 CSeq: 1 INVITE Via: SIP/2.0/UDP CARRIER_IP:5060;branch=b18bbcc394bb5dae43946b12f5d5fe0e Contact: sip:CALLER_ID@CARRIER_IP:5060 Content-Type: application/sdp Content-Length: 303 v=0 o=carrierhostname 1234 0 IN IP4 CARRIER_MEDIA_IP s=sip call c=IN IP4 CARRIER_MEDIA_IP t=0 0 m=audio 27630 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 10:55:02.749669 IP MY_PUBLIC_IP.5060 > CARRIER_IP.5060: SIP, length: 396 E]..?...&~.)SIP/2.0 100 trying -- your call is important to us To: "unknown" From: "WIRELESS CALLER" ;tag=3609590102-643763 Call-ID: 90553-3609590102-643720@msw1 CSeq: 1 INVITE Via: SIP/2.0/UDP CARRIER_IP:5060;branch=b18bbcc394bb5dae43946b12f5d5fe0e;rport=5060 Server: kamailio (4.1.3 (x86_64/linux)) Content-Length: 0 10:55:03.757740 IP MY_PUBLIC_IP.5060 > CARRIER_IP.5060: SIP, length: 1171 E^..?...&~.)SIP/2.0 200 OK Via: SIP/2.0/UDP CARRIER_IP:5060;rport=5060;branch=b18bbcc394bb5dae43946b12f5d5fe0e Record-Route: Record-Route: From: "WIRELESS CALLER" ;tag=3609590102-643763 To: "unknown" ;tag=as0d5ffe67 Call-ID: 90553-3609590102-643720@msw1 CSeq: 1 INVITE Server: Asterisk PBX 11.8.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: Content-Type: application/sdp Require: timer Content-Length: 281 v=0 o=root 607970891 607970891 IN IP4 MY_PUBLIC_IP s=Asterisk PBX 11.8.1 c=IN IP4 MY_PUBLIC_IP t=0 0 m=audio 17780 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes 10:55:04.257451 IP MY_PUBLIC_IP.5060 > CARRIER_IP.5060: SIP, length: 1171 E_..?...&~.)SIP/2.0 200 OK Via: SIP/2.0/UDP CARRIER_IP:5060;rport=5060;branch=b18bbcc394bb5dae43946b12f5d5fe0e Record-Route: Record-Route: From: "WIRELESS CALLER" ;tag=3609590102-643763 To: "unknown" ;tag=as0d5ffe67 Call-ID: 90553-3609590102-643720@msw1 CSeq: 1 INVITE Server: Asterisk PBX 11.8.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: Content-Type: application/sdp Require: timer Content-Length: 281 v=0 o=root 607970891 607970891 IN IP4 MY_PUBLIC_IP s=Asterisk PBX 11.8.1 c=IN IP4 MY_PUBLIC_IP t=0 0 m=audio 17780 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes 10:55:05.258278 IP MY_PUBLIC_IP.5060 > CARRIER_IP.5060: SIP, length: 1171 E`..?...&~.)SIP/2.0 200 OK Via: SIP/2.0/UDP CARRIER_IP:5060;rport=5060;branch=b18bbcc394bb5dae43946b12f5d5fe0e Record-Route: Record-Route: From: "WIRELESS CALLER" ;tag=3609590102-643763 To: "unknown" ;tag=as0d5ffe67 Call-ID: 90553-3609590102-643720@msw1 CSeq: 1 INVITE Server: Asterisk PBX 11.8.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: Content-Type: application/sdp Require: timer Content-Length: 281 v=0 o=root 607970891 607970891 IN IP4 MY_PUBLIC_IP s=Asterisk PBX 11.8.1 c=IN IP4 MY_PUBLIC_IP t=0 0 m=audio 17780 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes 10:55:07.257412 IP MY_PUBLIC_IP.5060 > CARRIER_IP.5060: SIP, length: 1171 Ea..?...&~.)SIP/2.0 200 OK Via: SIP/2.0/UDP
Re: [SR-Users] Kamailio with Sqlite
Hello,
try with:
modparam("auth_db", "db_url", "sqlite:etc/kamailio/kamailio.db")
Note the four times /.
Cheers,
Daniel
On 20/05/14 16:09, Ashwin Kumar R wrote:
Hi,
I want to configure kamailio to use SQLITE instead of MYSQL.I made
DB_ENGINE=SQLITE in kamctlrc file and loadmodule "db_sqlite.so" in
kamctl.conf file.
But when I changed these parameters
modparam("auth_db", "db_url", "sqlite:///etc/kamailio/kamailio.db")
#!define DBURL "sqlite:///etc/kamailio/kamailio.db"
Im getting the following error
ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start
failed
I checked the log file ,following is the comment in syslog
May 20 19:16:30 r13pc80-desktop /usr/local/sbin/kamailio[10920]:
ERROR: db_sqlite [dbase.c:67]: db_sqlite_new_connection(): failed to
open sqlite database 'etc/kamailio/kamailio.db'
What kamailio.cfg and kamctlrc parameters do I need to change to do
this?
Regards
Ashwin
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[SR-Users] Kamailio with Sqlite
Hi,
I want to configure kamailio to use SQLITE instead of MYSQL.I made
DB_ENGINE=SQLITE in kamctlrc file and loadmodule "db_sqlite.so" in
kamctl.conf file.
But when I changed these parameters
modparam("auth_db", "db_url", "sqlite:///etc/kamailio/kamailio.db")
#!define DBURL "sqlite:///etc/kamailio/kamailio.db"
Im getting the following error
ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start
failed
I checked the log file ,following is the comment in syslog
May 20 19:16:30 r13pc80-desktop /usr/local/sbin/kamailio[10920]: ERROR:
db_sqlite [dbase.c:67]: db_sqlite_new_connection(): failed to open
sqlite database 'etc/kamailio/kamailio.db'
What kamailio.cfg and kamctlrc parameters do I need to change to do
this?
Regards
Ashwin
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Re: [SR-Users] Kamailio and MSILO with TLS AORs
Hi Roberto, Did you ever figure out how to do it? I'd be interested too. Cheers, Peter On Thu, Apr 17, 2014 at 4:04 PM, Roberto Fichera wrote: > On 04/17/2014 10:36 AM, Roberto Fichera wrote: > > Hi All, > > > On 04/16/2014 07:15 PM, Roberto Fichera wrote: > > > > Hi All, > > > >> On 04/16/2014 06:53 PM, Daniel-Constantin Mierla wrote: > >>> Hello, > >> Ciao Daniel, > >> > >>> most likely the messages are looped back, be sure you allow requests > from 'myself' without authentication. > >>> > >>> You can watch loopback interface with ngrep to see if the messages are > sent out. > >> Yep! Allowing requests from myself solved the problem! Thanks! > >> > >> Another thing regarding MSILO, I would like to preserve some custom > header fields > >> for both instant message and notification. How can I do it? > > I was able to solve this issue too, pretty straight I mean, there is the > extra_hdrs_avp param > > which did the trick. BTW, is there a way to send back the notification > to the sender that > > the message has been sent correctly via m_dump(), so to get a 200 about > the given > > m_dump()ed message? > > > > I think that this can be solved in a different way, the idea is to add an > event_route for the msilo > module so that we get notified for each m_dump()ed message about the given > IM status and > then handle it in the cfg file. So, looking at the code within > m_tm_callback() it seems that it's > right place where to add such feature. Since I'm not a kamailio developer > but I've quite long > time experience in C coding, does anyone can show/point me to some example > about > how to do that? > > Thanks in advance, > Roberto Fichera. > > ___ > 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
Re: [SR-Users] Realtime integration: Unregistered clients showing as registered?
Hello, Thanks for your suggestion, unfortunately it had no effect on the outcome. This (using asterisk-kamailio integration with a domain specified for clients) must have been achieved before, I wonder if I'm doing something wrong here, or is this just not doable? Thanks, Olli 2014-05-18 21:29 GMT+03:00 VOIP Tests : > Try updating your /etc/hosts file with the domain 'testers.com'. > > Arun > > > ___ > 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
Re: [SR-Users] Upgrade to version 4.1
Thanks Daniel. On Tue, May 20, 2014 at 9:28 AM, Daniel Tryba wrote: > On Tuesday 20 May 2014 15:35:15 VOIP Tests wrote: > > Hello, I understand that there are some bugs in ver 4.03. Can someone let > > me know how I can update my running version of Kamailio ( 4.03 ) to > version > > 4.1? Will I need to do a fresh install of the new version or is there a > > way I can do an upgrade/update to ver 4.03? > > Seen the migration guide: > > http://www.kamailio.org/wiki/install/upgrade/4.0.x-to-4.1.0 > > You need to update some sql tables if you use them, upgrading myiasm to > innodb > is optional. Can't remember running into any problems. > > -- > > POCOS B.V. - Croy 9c - 5653 LC Eindhoven > Telefoon: 040 293 8661 - Fax: 040 293 8658 > http://www.pocos.nl/ - http://www.sipo.nl/ > K.v.K. Eindhoven 17097024 > > ___ > 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
Re: [SR-Users] Upgrade to version 4.1
On Tuesday 20 May 2014 15:35:15 VOIP Tests wrote: > Hello, I understand that there are some bugs in ver 4.03. Can someone let > me know how I can update my running version of Kamailio ( 4.03 ) to version > 4.1? Will I need to do a fresh install of the new version or is there a > way I can do an upgrade/update to ver 4.03? Seen the migration guide: http://www.kamailio.org/wiki/install/upgrade/4.0.x-to-4.1.0 You need to update some sql tables if you use them, upgrading myiasm to innodb is optional. Can't remember running into any problems. -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024 ___ 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
[SR-Users] Upgrade to version 4.1
Hello, I understand that there are some bugs in ver 4.03. Can someone let me know how I can update my running version of Kamailio ( 4.03 ) to version 4.1? Will I need to do a fresh install of the new version or is there a way I can do an upgrade/update to ver 4.03? Thank you, Arun ___ 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
Re: [SR-Users] [rtpengine] No media from WebRTC UA
On 05/16/14 20:30, Alexey Rybalko wrote: > During the call from Fire I saw a lot of "SRTP output wanted, but no > crypto suite was negotiated" messages from rtpengine. However DTLS is > finally was established. Is that one more issue of Firefox? Some DTLS-SRTP endpoints seem to be slow with starting or accepting the DTLS handshake. What I've seen in my tests was that some "active" endpoints have a delay of a few seconds before they start the DTLS handshake, while some "passive" endpoints only start accepting the DTLS handshake after they've received the signalling answer (while the RFC requires them to accept DTLS right away after sending the offer). Both of these cases somewhat eliminate one of the advantages that DTLS-SRTP has over SDES: it still breaks early media. If your RTP side sends media before the SRTP side has finished the DTLS handshake, you will lose packets and see those messages popping up. But I hope that these issues will disappear as the code matures. > Looking in STUN section of the dump files I wonder why Chrome use more > than 10 binding request (USE-CANDIDATE) for each candidate while > Mozilla does it just once. I've noticed that too and I have no idea why it does that. Doesn't seem to be causing any problems though. cheers ___ 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
[SR-Users] code review tools
Hello, some time ago it was proposed to use a code review tool to analyze patches, the topic was also discussed during the last IRC devel meeting. I cc-ed users mailing list, because people there might have experiences with some of them. I found on wikipedia a short list of available applications for this purpose: - http://en.wikipedia.org/wiki/List_of_tools_for_code_review Personally I haven't used any so far. Iirc, gerrit was proposed before by Victor Seva. From the wikipedia list, Differential would be straightforward to deploy from dependencies point of view, as we have php on our servers. But I have no clue which is the right/best to use, so here we have to decide if worth to use one and which one. Besides the above one, perhaps github has an integrated review tool. I can be considered as an alternative as well. My opinion is that such tool is useful, specially for big patches, to allow easier more people to have a look at them. Hoping to get feedback on benefits of such tools and which would be the right/best to use. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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
Re: [SR-Users] failed to find command redis_cmd
I enhanced the log message to print the number of parameters for the
functions that is searched -- hope that helps in the future.
Cheers,
Daniel
On 20/05/14 04:08, Ryan Brindley wrote:
Yea. I thought the same thing once I figured it out. The error is
confusing but then it does make sense once you understand its looking
for the 2 argument prototype.
Maybe an additional "wrong parameters?" at the end of the error
message could help prevent misdirection debugging.
Thanks Daniel for helping!
Ryan Brindley
Software Development Officer
Stratics Networks, Inc.
1.866.635.6918 x108
On Mon, May 19, 2014 at 4:51 PM, Alex Balashov
mailto:abalas...@evaristesys.com>> wrote:
On 05/19/2014 05:02 PM, Ryan Brindley wrote:
Wow...I found the issue :-/
The example I gave in this thread was not a copy paste and I had
redis_cmd("name", "SET foo bar" "r"); (not the lack of the
second comma).
Yeah, it's somewhat confusing. However, when you invoke a function
according to a prototype that doesn't exist, the fixup functions
will return an error in such a way as to produce a "function
doesn't exist" message in the logs. It does exist, just not in the
two-argument variety. :-)
--
Alex Balashov - Principal
Evariste Systems LLC
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
Please be kind to the English language:
http://www.entrepreneur.com/article/232906
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