Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc
Take a look at the “fingerprint:” line. From: John Nash Sent: Thursday, June 23, 2016 3:42 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc Actually the issue is i hear no audio on either side and just after session progress (I guess when media starts coming from remote media server) i see error "SRTP output wanted, but no crypto suite was negotiated" I had also checked media logs i could see RTP packets being sent from freeswitch to RTPengine IP but there was no packet at all just after that. Ideally after RTP packet from freeswitch to rtpengine, Rtpengine should send that packet to browser using wss? On Fri, Jun 24, 2016 at 1:05 AM, Eric Tamme wrote: So - i dont see a problem here - Chrome is getting UDP/TLS/RTP/SAVPF and Freeswitch is getting RTP/AVP. Freeswitch responded to the offer in the invite with an answer in the 183, and in the 200. What is the failure you are seeing, and where is it happening (in freeswitch? in the browser?) The only thing that looks bad is that you are retransmitting the ACK which FS either ... doesnt like, or is never getting, because it keeps retransmitting the 200, which is why you get a 481 when you send BYE. -Eric On 06/23/2016 01:24 PM, John Nash wrote: OK here is the log https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744 Sorry took me a while to convert wireshark trace to text file. My freeswitch is running on private IP (127.0.0.1) and opensips I run on both public and private so that for outside world opensips is the only public IP they see. In proxy log I pasted Opensips ===> Freeswitch logs and back. On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme wrote: No - it's annoying to look at a trace that's had information removed and try and piece together whats happening. Your paranoid side is wrong, sorry. -Eric On 06/23/2016 01:06 PM, Patrick Wakano wrote: my paranoic side would recommend to hide/change private informations, specially any authentication line that might appear... this is certainly a sort of social engineering threat we should worry... better be safe than sorry On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme wrote: I mean you can use a private gist, but you will be publishing the link in a public email list. In general I personally dont believe revealing ip addresses etc. is any problem - to put my money where my mouth is here is a gist link to an unaltered SIP trace on my server :) https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 -Eric On 06/23/2016 12:23 PM, John Nash wrote: Ok i am ready with logs. About gist may I use private option as traces have our IPs, user On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme wrote: Hey John, Please paste a full UNALTERED sip trace into a gist (gist.github.com) from the proxy servers perspective and provide a link so that we can see what comes in, and what goes out from both sides. EG: ngrep -qtd any -W byline port 5060 This will show us the traffic that is leaving the proxy destined for the Freeswitch box, and what the freeswitch box sends back. Also - you can look in your browsers console log and provide the SIP trace from there in a seperate gist, so that we can see what opensips sends back up to your browser. -Eric Am I using correct sip.js example? I copied it to my server and accessing it using https: (used letsencrypt) On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme wrote: 1. I would suggest using SIP.js - https://github.com/onsip/SIP.js it is a much more active project that sipml5. 2. Im guessing that you are not properly passing flags to RTPEngine. If you want to have DTLS-SRTP between the browser, and plain RTP/AVP between RTPEngine and freeswitch, you need to "offer" rtp/avp to freeswitch, and "answer" dtls-srtp back up to the browser. the offer to freeswitch would be: $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; and the answer back up to the browswer would be: $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; -Eric On 06/23/2016 08:20 AM, John Nash wrote: I am following http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and trying to test a call sipml5 --->Opensips + rtpengine > SIP end point (Freeswitch) But I do not have any audio on both sides. I see this error at rtpengine log "SRTP output wanted, but no crypto suite was negotiated" Anyone tested this scenario positive? __
Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc
Hi, the rtpengine cannot negotiate SRTP between the two points, both must support the same cryptography and protocol. eg; SRTP to SRTP , DTLS/SRTP to DTLS/SRTP cipher 128 to 128 and 256 to 256. You can print the request body ($rb) on the INVITE with “application/sdp” and visually compare the exchange, do this on offer and answer. From: John Nash Sent: Thursday, June 23, 2016 3:42 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc Actually the issue is i hear no audio on either side and just after session progress (I guess when media starts coming from remote media server) i see error "SRTP output wanted, but no crypto suite was negotiated" I had also checked media logs i could see RTP packets being sent from freeswitch to RTPengine IP but there was no packet at all just after that. Ideally after RTP packet from freeswitch to rtpengine, Rtpengine should send that packet to browser using wss? On Fri, Jun 24, 2016 at 1:05 AM, Eric Tamme wrote: So - i dont see a problem here - Chrome is getting UDP/TLS/RTP/SAVPF and Freeswitch is getting RTP/AVP. Freeswitch responded to the offer in the invite with an answer in the 183, and in the 200. What is the failure you are seeing, and where is it happening (in freeswitch? in the browser?) The only thing that looks bad is that you are retransmitting the ACK which FS either ... doesnt like, or is never getting, because it keeps retransmitting the 200, which is why you get a 481 when you send BYE. -Eric On 06/23/2016 01:24 PM, John Nash wrote: OK here is the log https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744 Sorry took me a while to convert wireshark trace to text file. My freeswitch is running on private IP (127.0.0.1) and opensips I run on both public and private so that for outside world opensips is the only public IP they see. In proxy log I pasted Opensips ===> Freeswitch logs and back. On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme wrote: No - it's annoying to look at a trace that's had information removed and try and piece together whats happening. Your paranoid side is wrong, sorry. -Eric On 06/23/2016 01:06 PM, Patrick Wakano wrote: my paranoic side would recommend to hide/change private informations, specially any authentication line that might appear... this is certainly a sort of social engineering threat we should worry... better be safe than sorry On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme wrote: I mean you can use a private gist, but you will be publishing the link in a public email list. In general I personally dont believe revealing ip addresses etc. is any problem - to put my money where my mouth is here is a gist link to an unaltered SIP trace on my server :) https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 -Eric On 06/23/2016 12:23 PM, John Nash wrote: Ok i am ready with logs. About gist may I use private option as traces have our IPs, user On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme wrote: Hey John, Please paste a full UNALTERED sip trace into a gist (gist.github.com) from the proxy servers perspective and provide a link so that we can see what comes in, and what goes out from both sides. EG: ngrep -qtd any -W byline port 5060 This will show us the traffic that is leaving the proxy destined for the Freeswitch box, and what the freeswitch box sends back. Also - you can look in your browsers console log and provide the SIP trace from there in a seperate gist, so that we can see what opensips sends back up to your browser. -Eric Am I using correct sip.js example? I copied it to my server and accessing it using https: (used letsencrypt) On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme wrote: 1. I would suggest using SIP.js - https://github.com/onsip/SIP.js it is a much more active project that sipml5. 2. Im guessing that you are not properly passing flags to RTPEngine. If you want to have DTLS-SRTP between the browser, and plain RTP/AVP between RTPEngine and freeswitch, you need to "offer" rtp/avp to freeswitch, and "answer" dtls-srtp back up to the browser. the offer to freeswitch would be: $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; and the answer back up to the browswer would be: $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; -Eric On 06/23/2016 08:20 AM, John Nash wrote: I am following http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and trying to test a call sipml5 --->Opensips + rtpengin
Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects?
Hi, have you tried/considered running a simple query on the database and
parsing for the information you need?
From: Rodrigo Pimenta Carvalho
Sent: Tuesday, June 21, 2016 10:39 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the
caller, without side effects?
Hi Răzvan.
I have tried that idea. But that didn't work. The SIP INVITE message is being
changed by the OpenSIPS in a wrong way, in my point of view.
Do you know some way to save the entire SIP INVITE message before calling
lookup() and then make the saved message take place after the lookup()
execution?
My original message is:
INVITE sip:6...@mydomain.com.br SIP/2.0
Via: SIP/2.0/TCP 192.168.21.40:5090;rport;branch=z9hG4bK876727215
From: ;tag=179920819
To:
Call-ID: 1410250893
CSeq: 21 INVITE
Contact:
Proxy-Authorization: Digest username="crdphmacl_SPnuV5xqtnSX",
realm="localhost", nonce="5769458c01cc263a7c0d6995dc48d42288ec6f8e4048",
uri="sip:6...@mydomain.com.br", response="0f4c122d2a0a28dea6194c235cd77430",
algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Subject: Phone call
Content-Length: 227
This is being changed to:
INVITE
sip:crdphmacl_SPnuV5xqtnSX@131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81
SIP/2.0
Record-Route:
Via: SIP/2.0/TCP myDomain.com.br:5060;branch=z9hG4bKe2db.49d54587.0;i=1
Via: SIP/2.0/TCP
192.168.21.40:5090;received=xxx.yyy.240.204;rport=60672;branch=z9hG4bK716249970
From: ;tag=12586028
To:
Call-ID: 1106771604
CSeq: 21 INVITE
Contact:
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Subject: Phone call
Content-Length: 224
So, the caller is receiving its own SIP INVITE.
That is why when A calls B, is A that rings, not B.
It is becoming a bit complicated. So, I suspect I'm going to the incorrect
direction
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org em nome
de Răzvan Crainea
Enviado: terça-feira, 21 de junho de 2016 04:24
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the
caller, without side effects?
Hi, Rodrigo!
Have you tried restoring the R-URI after the caller lookup? Something like:
$var(ru) = $ru;
lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I
think is more suitable than from contact uri
$ru = $var(ru);
# continue your processing here
# now do the real lookup for the callee
lookup("location");
Don't do the lookups in the reversed way, because you might loose some contacts.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.comHome — OpenSIPS Solutions
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS
is more than a SIP proxy/router as it includes application-level
functionalities.
On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote:
Dear OpenSIPS-users,
The table location has the column attr where I use to store specific
additional information for each registration.
Whenever A calls B, I have to read this specific information from the A
record and from the B record. That is, I need to get and handle specific
information about the caller and callee.
For the callee, I use to invoke the lookup("location") function that put the
needed information in the attr_avp. That is good and works very well. Then, I
just have to read the attr_avp to get such specific information.
For the caller, I use to invoke:
$var(aorChamador) = $(ct.fields(uri));
lookup("location","","$var(aorChamador)");
However it causes amazing side effect in the SIP signaling. Ex: When A calls
B, B stays quiet and A rings. So A can answer A. Crazy!
According to the documentation, lookup will overwritten the Request-URI. I
guess that is why the SIP signaling become incoherent.
How could I get the caller attr specific information without side effects?
Any hint will be very helpful!!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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Re: [OpenSIPS-Users] Connecting to mongodb Replica set from opensips .
Hi, some modules are not safe to use with mongodb, namely; acc (Cannot write to
missed_call), msilo (Does not delete stored messages after sent), avpops(do not
recall the issue with this one) and there may be others.
To connect to the replica set (Minimum 3 replicas for proper master election):
All the modules (cachedb_mongodb.so, db_cachedb.so etc.) should point to the
same db name in the xdb_url module parameter.
loadmodule "cachedb_mongodb.so"
loadmodule "db_cachedb.so"
modparam("cachedb_mongodb",
"cachedb_url","mongodb:replicaset1://root:password@1.2.3.4:27017,2.3.4.5:27017,5.5.5.5:27017,/jack.db.CallCenter_Info")
modparam("db_cachedb","cachedb_url","mongodb:replicaset1://root:password@1.2.3.4:27017,2.3.4.5:27017,5.5.5.5:27017,/jack.db.CallCenter_Info")
loadmodule "usrloc.so"
modparam("usrloc",
"db_url","mongodb:replicaset1://root:password@1.2.3.4:27017,2.3.4.5:27017,5.5.5.5:27017,/jack.db.CallCenter_Info")
Use the mongo slave_ok,1 parameter to read write via slaves.
For the version table, you will need to create a copy in mongodb with the same
name as the version table in mysql. Use the db_version_table= global parameter
if you want to change the default name of the table. >From mongo shell do this:
use db
db.version.insert({ table_version : NumberInt(1009), table_name : "location"})
db.version.insert({ table_version : NumberInt(10), table_name : "dialog"})
From: Sasmita Panda
Sent: Wednesday, June 01, 2016 6:17 AM
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] Connecting to mongodb Replica set from opensips .
HI Guys ,
i am trying to use mongodb in replica set . I have tested my opensips-1.11
with a stand alone mongodb instance . Its working fine .
Now I am trying to connect to a replica set . But somehow I amnot able
to do this . Opensips is getting connected to the replica set but its not able
to run raw query .
My opensips configuration is like bellow :
loadmodule "cachedb_mongodb.so"
loadmodule "db_cachedb.so"
modparam("cachedb_mongodb",
"cachedb_url","mongodb:replicaset1://root:password@1.2.3.4:27017,2.3.4.5:27017/jack.db.CallCenter_Info")
modparam("db_cachedb","cachedb_url","mongodb:replicaset1://root:password@1.2.3.4:27017,2.3.4.5:27017/jack.db.test")
modparam("acc", "db_url", "cachedb://mongodb:replicaset1")
modparam("acc", "db_flag", 2)
modparam("acc", "log_flag", 2)
modparam("acc", "cdr_flag", 1)
modparam("acc", "log_facility", "LOG_LOCAL7")
Opensips is not able to execute version check query for acc module . Bellow is
the logs of opensips .
INFO:cachedb_mongodb:mongo_new_connection: Connected at server
52.71.216.67:27017,23.21.65.168:27017 with version 3.0.12 , to db db.test
DBG:db_cachedb:db_cachedb_init: Succesfully initiated connection to
[mongodb:replicaset1]
DBG:cachedb_mongodb:mongo_db_query_trans: Running raw mongo query on table
db.my_version_table
ERROR:cachedb_mongodb:mongo_db_query_trans: Failed to run query. Err = 0, 0 , 0
DBG:cachedb_mongodb:mongo_db_query_trans: Fetched key lastOp
DBG:cachedb_mongodb:mongo_db_query_trans: (unknown type 17)
DBG:cachedb_mongodb:mongo_db_query_trans: Fetched key connectionId
DBG:cachedb_mongodb:mongo_db_query_trans: (int) 516
DBG:cachedb_mongodb:mongo_db_query_trans: Fetched key n
DBG:cachedb_mongodb:mongo_db_query_trans: (int) 0
DBG:cachedb_mongodb:mongo_db_query_trans: Fetched key syncMillis
DBG:cachedb_mongodb:mongo_db_query_trans: (int) 0
DBG:core:wait_status_code: read code 0 ? rc = 0, errno=Success
INFO:core:daemonize: pre-daemon process exiting with -1
In my mongodb instance there is some error printing also .
[initandlisten] connection accepted from 104.131.6.7:35857 #565 (7 connections
now open)
2016-06-01T10:09:14.802+ I QUERY[conn565] assertion 13 not authorized
for query on db.my_version_table ns:db.my_version_table query:{ $query: {
table_name: "acc" } }
By login through a remote machine with same username and password . I am
able to execute all query manually . But why my opensips cant do this . If
anybody have tried this can you suggest me any solution . What else I should do
in my replica set or in opensips to make it happen ?
Please help me . As I am quite new to mongodb replica set so not able to
understand the actual problem also .
Thanks & Regards
Sasmita Panda
Network Testing and Software Engineer
3CLogic , ph:07827611765
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Re: [OpenSIPS-Users] Un recognized sip header from asterisk to OpenSIPS
Hi, do a normal sip trace using the sip trace module in opensips and post the output here. From what that can be understood from this log though, is that there are headers “x-asterisk-hangupcause:” “x-asterisk-hangupcausecode:” that are being rejected due to validity or case sensitivity, possibly the response from the user agent. Where are these headers being added? From: Travis Manson-Drake Sent: Thursday, March 31, 2016 1:47 PM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] Un recognized sip header from asterisk to OpenSIPS Hello everyone. Hope your all doing well! I seem to be having an issue in which when a call is sent through OpenSIPS to my Asterisk PBX asterisk with eventually send a BYE with a hang up Cause of 111/unrecognized sip header. I looked at the headers of all my packets but can’t find anything out of the norm. has anyone experienced this before and ideas on what it might be or what I might check? I found a few article on asterisk forums mention NAT issues, but I’ve implemented a NAT helper into my routing logic so that shouldn’t be the case. Thank you all for your time Travis Manson-Drake Voice Systems Analyst Simply Bits, LLC T: 520.545.0311 F: 520.545.7252 E: trav...@simplybits.com 5225 N. Sabino Canyon Road Tucson, AZ 85750 Support Hotline: 520.545.0333 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] RTPEngine - garbage collector deletes active branch
Hi, don’t send the rtpengine_delete command and let rtpengine handle it
automatically. The logic is that, in a multi-branch call only one can
answer, therefore there is only one established session even though multiple
'offers' . You can really only delete an established session, sending the
delete will delete that session. Only send the delete when missed call to
free the proxy quickly. This is how I handle it anyway, the unanswered
offers are just discarded.
-Original Message-
From: Julian Santer
Sent: Tuesday, March 08, 2016 5:47 AM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] RTPEngine - garbage collector deletes active
branch
Hi guys,
we are implementing OpenSips 2.1.2 with rptengine version 4.3.0.0-git
git-master-926d9d3.
Now we got a problem on calls with multiple branches.
We call rtpengine_manage("replace-origin replace-session-connection
ICE=remove via-branch=extra");
The extra-id is containing the call ID and the branch ID
Here a example with 2 branches:
INVITE 1. branch
(ZzJFVwVlR0BDd2RHMF8DFUZZLAQUQjBcBWReR1ZzEUFHWgQQS0dAAGFCWlwcYUE-0):
Mar 8 11:14:25 Received command 'offer' from 1.2.3.4:59038
Mar 8 11:14:25 Dump for 'offer' from 1.2.3.4:59038: { "sdp":
"v=0#015#012o=CiscoSystemsSIP-GW-UserAgent 4454 7726 IN IP4
2.3.4.5#015#012s=SIP
Call#015#012c=IN IP4 2.3.4.6#015#012t=0 0#015#012m=audio 59482 RTP/AVP 18 8
0 4 125 101#015#012c=IN IP4 2.3.4.6#015#012a=rtpmap:18
G729/8000#015#012a=fmtp:18 annexb=yes#015#012a=fmtp:4
bitrate=5.3;annexa=no#015#012a=rtpmap:125 X-CCD/8000#015#012a=rtpmap:101
telephone-event/8000#015#012a=fmtp:101 0-16#015#012", "ICE": "remove",
"replace": [ "origin", "session-connection" ], "call-id":
"ZzJFVwVlR0BDd2RHMF8DFUZZLAQUQjBc ...
Mar 8 11:14:25 ... BWReR1ZzEUFHWgQQS0dAAGFCWlwcYUE-", "via-branch":
"ZzJFVwVlR0BDd2RHMF8DFUZZLAQUQjBcBWReR1ZzEUFHWgQQS0dAAGFCWlwcYUE-0",
"received-from": [ "IP4", "1.2.3.5" ], "from-tag": "D40356B0-22E3",
"command": "offer" }
Mar 8 11:14:25 Creating new call
Mar 8 11:14:25 set FILLED flag for stream 2.3.4.6:59482
Mar 8 11:14:25 set FILLED flag for stream 2.3.4.6:59483
Mar 8 11:14:25 Replying to 'offer' from 1.2.3.4:59038
Mar 8 11:14:25 Response dump for 'offer' to 1.2.3.4:59038: { "sdp":
"v=0#015#012o=CiscoSystemsSIP-GW-UserAgent 4454 7726 IN IP4
1.2.3.6#015#012s=SIP
Call#015#012c=IN IP4 1.2.3.6#015#012t=0 0#015#012m=audio 5 RTP/AVP 18 8
0 4 125 101#015#012c=IN IP4 1.2.3.6#015#012a=rtpmap:18
G729/8000#015#012a=fmtp:18 annexb=yes#015#012a=fmtp:4
bitrate=5.3;annexa=no#015#012a=rtpmap:125 X-CCD/8000#015#012a=rtpmap:101
telephone-event/8000#015#012a=fmtp:101
0-16#015#012a=sendrecv#015#012a=rtcp:50001#015#012", "result": "ok" }
INVITE 1. branch
(ZzJFVwVlR0BDd2RHMF8DFUZZLAQUQjBcBWReR1ZzEUFHWgQQS0dAAGFCWlwcYUE-1):
Mar 8 11:14:25 Received command 'offer' from 1.2.3.4:59038
Mar 8 11:14:25 Dump for 'offer' from 1.2.3.4:59038: { "sdp":
"v=0#015#012o=CiscoSystemsSIP-GW-UserAgent 4454 7726 IN IP4
2.3.4.5#015#012s=SIP
Call#015#012c=IN IP4 2.3.4.6#015#012t=0 0#015#012m=audio 59482 RTP/AVP 18 8
0 4 125 101#015#012c=IN IP4 2.3.4.6#015#012a=rtpmap:18
G729/8000#015#012a=fmtp:18 annexb=yes#015#012a=fmtp:4
bitrate=5.3;annexa=no#015#012a=rtpmap:125 X-CCD/8000#015#012a=rtpmap:101
telephone-event/8000#015#012a=fmtp:101 0-16#015#012", "ICE": "remove",
"replace": [ "origin", "session-connection" ], "call-id":
"ZzJFVwVlR0BDd2RHMF8DFUZZLAQUQjBc ...
Mar 8 11:14:25 ... BWReR1ZzEUFHWgQQS0dAAGFCWlwcYUE-", "via-branch":
"ZzJFVwVlR0BDd2RHMF8DFUZZLAQUQjBcBWReR1ZzEUFHWgQQS0dAAGFCWlwcYUE-1",
"received-from": [ "IP4", "1.2.3.5" ], "from-tag": "D40356B0-22E3",
"command": "offer" }
Mar 8 11:14:25 Replying to 'offer' from 1.2.3.4:59038
Mar 8 11:14:25 Response dump for 'offer' to 1.2.3.4:59038: { "sdp":
"v=0#015#012o=CiscoSystemsSIP-GW-UserAgent 4454 7726 IN IP4
1.2.3.6#015#012s=SIP
Call#015#012c=IN IP4 1.2.3.6#015#012t=0 0#015#012m=audio 50040 RTP/AVP 18 8
0 4 125 101#015#012c=IN IP4 1.2.3.6#015#012a=rtpmap:18
G729/8000#015#012a=fmtp:18 annexb=yes#015#012a=fmtp:4
bitrate=5.3;annexa=no#015#012a=rtpmap:125 X-CCD/8000#015#012a=rtpmap:101
telephone-event/8000#015#012a=fmtp:101
0-16#015#012a=sendrecv#015#012a=rtcp:50041#015#012", "result": "ok" }
OK 1. branch
(ZzJFVwVlR0BDd2RHMF8DFUZZLAQUQjBcBWReR1ZzEUFHWgQQS0dAAGFCWlwcYUE-0):
Mar 8 11:14:33 Received command 'offer' from 1.2.3.4:47625
Mar 8 11:14:33 Dump for 'offer' from 1.2.3.4:47625: { "sdp":
"v=0#015#012o=root 2042928516 2042928517 IN IP4
3.4.5.6#015#012s=call#015#012c=IN IP4
3.4.5.6#015#012t=0 0#015#012m=audio 51372 RTP/AVP 18 8 0
101#015#012a=rtpmap:18 G729/8000#015#012a=fmtp:18
annexb=yes#015#012a=rtpmap:8
PCMA/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101
telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=sendrecv#015#012",
"ICE": "remove",
"replace": [ "origin", "session-connection" ], "call-id":
"ZzJFVwVlR0BDd2RHMF8DFUZZLAQUQjBcBWReR1ZzEUFHWgQQS0dAAGFCWlwcYUE-",
"via-branch" ...
Mar 8 11:14:33 ... :
"Zz
Re: [OpenSIPS-Users] Bflag problem!
I noticed that too with 2.1 latest git version, branch flags are not being
saved in the location table. I reverted as a result. This is just to confirm.
From: Dragomir Haralambiev
Sent: Thursday, March 03, 2016 9:41 AM
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] Bflag problem!
Hello,
After latest update of Opensips 2.2 I have problem with "setbflag".
I can not set bflag.
Here is example:
modparam("nathelper", "remove_on_timeout_bflag", "RM_ONTO_FLAG")
.
setbflag(RM_ONTO_FLAG);
if (isbflagset(RM_ONTO_FLAG)) {
xlog("L_ERR", "RM_ONTO_FLAG is set");
}
Best regards,
Dragomir
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Re: [OpenSIPS-Users] New OpenXCAP release 2.4.0
Hi, there is a message "No Module named sipsimple.core" when tried to start the OpenXCAP server. How can this be resolved? I'm thinking it is referring to the "python-sipsimple" package but that is a beast to compile along with ffmpeg. -Original Message- From: Adrian Georgescu Sent: Thursday, January 07, 2016 1:06 PM To: OpenSIPS users mailling list ; OpenSIPS devel mailling list Subject: [OpenSIPS-Users] New OpenXCAP release 2.4.0 Hello, There is a new release of OpenXCAP with various bug fixes openxcap (2.4.0) unstable; urgency=medium * Removed runtime dependency check * Updated copyright years openxcap (2.3.0) unstable; urgency=medium * Refactor xcapdiff publishing for OpenSIPS backend * Fix dependency name * Drop dependency on twisted-web * Raise python-application version dependency * Add dependency on python-sipsimple http://openxcap.org Regards, Adrian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [BOOK] Building Telephony Systems with OpenSIPS - 2.1 version
Hi, set_dlg_flag() only accepts indices 0-31 , so should be set_dlg_flag(“5”);
From: Flavio Goncalves
Sent: Monday, February 15, 2016 6:05 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] [BOOK] Building Telephony Systems with OpenSIPS -
2.1 version
Hi Schneur,
That's correct, it is NAT_FLAG. These typos were identified and fixed in the
reviews. I don't know how they appeared in the final version. Whenever you find
5 is (NAT_FLAG) and 6 (NAT_BFLAG)
Best regards,
Flavio E. Goncalves
CEO - V.Office Redes e Telecomunicações Ltda.
Phone: +55 48-3332-8590
Skype:flaviogoncalves1
Linkedin: www.linkedin.com/in/flavioegoncalves
Twitter: www.twitter.com/asteriskguide
2016-02-15 8:07 GMT-02:00 Schneur Rosenberg :
Hi Bogdan
On page 250 of the book the third line from the top it says
if (is_dlg_flag_set("5")) setflag(NAT_FLAG);
Is that correct, or is it supposed to be?
if (is_dlg_flag_set(NAT_FLAG)) setflag(NAT_FLAG);
The dialog flag on the bottom of page 249 is set
set_dlg_flag(NAT_FLAG); and not set_dlg_flag("5");
On Thu, Feb 11, 2016 at 11:35 AM, Bogdan-Andrei Iancu
wrote:
Hi Schneur,
OK, I will check with the editor if they can push fixes to the book. Thanks
for reporting it!
Thanks and Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.comOn 09.02.2016 13:20, Schneur Rosenberg wrote:
Chapter 1, page 11 (page 36 of the eBook) on the bottom of the page there
is a initial invite of userA calling userB, and the contact is of userB, it
should be userA
On Feb 8, 2016 3:13 PM, "Bogdan-Andrei Iancu" wrote:
Hi Schneur,
Could you point the name of the chapter and the context of that
mistake, so I can double check it ?
Thanks & Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.comOn 07.02.2016 14:53, Schneur Rosenberg wrote:
I bought the book and I think I found the first mistake, in the
initial invite packet on page 11, the contact header is of the destination and
not of the origination UAC
On Feb 4, 2016 3:37 PM, "Bogdan-Andrei Iancu"
wrote:
Hello all,
Flavio Goncalves and I are happy to announce the publishing of the
second edition of "Building Telephony Systems with OpenSIPS", covering OpenSIPS
version 2.1 .
Also many thanks to the Packt Publishing house for making it
happened and to all our reviewers who help us to make this book better.
https://www.packtpub.com/networking-and-servers/building-telephony-systems-opensips-second-edition
Enjoy !
--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
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Re: [OpenSIPS-Users] Linphone and OpenSIPS over TCP
Video and voice calls work fine with Linphone client over UDP, TCP, 3G/4G and wifi and when using a media proxy. Did not even try to install a TURN server to test your case, because Opensips does not have any modules to interact with a TURN server, you may be at a dead end with that. In the case of the SDP bodies being received by Opensips from the UA, these would need to be forwarded to the TURN server for modification and then returned to Opensips to be sent to the UA for the media to be redirected. Opensips interacts with the TURN in server MediaProxy only. From: Nabeel Sent: Tuesday, February 02, 2016 2:30 AM To: OpenSIPS users mailling list ; hamid2kv...@hotmail.com Subject: Re: [OpenSIPS-Users] Linphone and OpenSIPS over TCP Hi Hamid, I am not using the location table, I am using memory only to store the contacts. If you use TCP within the same network, where a TURN server is not required (TURN server not set in Linphone settings), I found that the calls then work depending on what the network allows. Within the same Wi-Fi network, in my case audio calls work but not video calls. But using 3G/4G only, with or without a TURN server set in Linphone settings, calls do not work at all. This is why I think the behaviour of TURN server has something to do with this. To those people stating that TCP works with Linphone, please test over 3G/4G where a TURN server might be required. On 2 Feb 2016 7:07 am, "Hamid Hashmi" wrote: Nabeel I have been using Linphone 3.6.1 with opensips for a long time. And its working fine on both UDP and TCP. I have gone through your logs, there is log line DBG:tm:matching_3261: RFC3261 transaction matching failed Please check contact of your "to number" in location table. Hamid R. Hashmi Software Engineer - VoIP Vopium A/S -- Date: Mon, 1 Feb 2016 19:35:53 + From: nabeelshik...@gmail.com To: bog...@opensips.org; users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Linphone and OpenSIPS over TCP Hi Bogdan, Below is the requested log for TCP call attempt. User is trying to call user via OpenSIPS server 162.248.6.120 : http://pastebin.com/UQ9mEemd On 1 February 2016 at 09:41, Bogdan-Andrei Iancu wrote: Hi, I strongly suggest to look into the opensips logs and see what opensips try to do with the call. Based on your saying (that you see a timeout), I suspect your OpenSIPS tries to deliver the call over TCP to a destination which does not listen on TCP. If you do not know hoe to interpret the logs, run opensips in debug=4 mode, upload the logs corresponding the INVITE execution and provide the link. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.comOn 31.01.2016 16:28, Nabeel wrote: Without using alias=domain.com, TCP still does not work. My initial request for someone to test this using Linphone remains. Please test and let me know if you can call using TCP with OpenSIPS listening on an IP address. On 31 January 2016 at 09:28, Nabeel wrote: On further testing, using the IP address instead of the domain name in the URI setting of Linphone works with TCP, so I think this might be to do with SRV/NAPTR records associated with the domain. On 31 January 2016 at 08:29, Nabeel wrote: Hello, There seems to be a problem with calls over TCP using Linphone, and since Linphone is a popular open source application, I would like someone to please verify this problem. Calls work fine with Linphone over UDP, but after registering with TCP using the same credentials, calls do not connect at all and lead to a request timeout. A request timeout does not say much about the cause, but in this case I suspect there is something wrong with TCP on the server side. I would like someone to please install Linphone on your phone and connect to your OpenSIPS server using UDP and TCP. Please report here if the calls work over both transports. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Linphone and OpenSIPS over TCP
It’s just a configuration issue you are having, start by: 1. In windows, do an “nslookup your.domain.net” if the returned IP’s are not what you expect, then correct this. You may also want to do a reverse lookup “nslookup xxx.xxx.xxx.xxx” to return its domain name. 2. Configure your Opensips to listen on these IP’s 3. Add “your.domain.net” in your domain table (these are the domains your sips is responsible for, IP’s can go in this table but not recommended if you are strictly using domain names in the clients for authentication) By the way, TCP works with Linphone very well. From: Nabeel Sent: Sunday, January 31, 2016 9:28 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Linphone and OpenSIPS over TCP Without using alias=domain.com, TCP still does not work. My initial request for someone to test this using Linphone remains. Please test and let me know if you can call using TCP with OpenSIPS listening on an IP address. On 31 January 2016 at 09:28, Nabeel wrote: On further testing, using the IP address instead of the domain name in the URI setting of Linphone works with TCP, so I think this might be to do with SRV/NAPTR records associated with the domain. On 31 January 2016 at 08:29, Nabeel wrote: Hello, There seems to be a problem with calls over TCP using Linphone, and since Linphone is a popular open source application, I would like someone to please verify this problem. Calls work fine with Linphone over UDP, but after registering with TCP using the same credentials, calls do not connect at all and lead to a request timeout. A request timeout does not say much about the cause, but in this case I suspect there is something wrong with TCP on the server side. I would like someone to please install Linphone on your phone and connect to your OpenSIPS server using UDP and TCP. Please report here if the calls work over both transports. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [OpenSIPS Public Meeting] Follow-up: Scripting variables in OpenSIPS 3.x
I want to suggest allowing some key functions to also accept variables as
input. To name a few: save(), lookup(), remove(), force_send_socket(). For
example, when using mongo, multiple Opensips servers and multiple location
tables it’s inflexible in the script to use some of these functions when you
have to hardcode something like, lookup(“location”,””) .
From: Liviu Chircu
Sent: Thursday, January 28, 2016 1:09 PM
To: OpenSIPS devel mailling list ; OpenSIPS users mailling list
Subject: [OpenSIPS-Users] [OpenSIPS Public Meeting] Follow-up: Scripting
variables in OpenSIPS 3.x
Hello!
The conclusions of yesterday's Public Meeting can be found here [1]
Here is a quick summary of the decided upon changes for variable restructuring
in OpenSIPS 3.x:
* variable referencing will now mandate an explicit scope (local, msg
(read/write into the SIP msg), tm, dlg, global)
* variables will still be dynamically typed
* improvements to explicit casting (from a couple of transformations to int(),
str(), list() and json())
* new data types: list and json, along with contructors and specific operators
("+" for concat, [] for indexing)
* AVPs dropped
* NULL special value to be kept
* find the best way of adapting the multi-index pseudo-vars, e.g.
$(hdr(Contact)[1])
This is still an ongoing effort, any feature is subject to change as we go
along with the development of OpenSIPS 3.0, so this is the ideal time to reply
to this thread and suggest improvements!
[1]: http://www.opensips.org/Community/IRCmeeting20160127
Best regards,
--
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
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Re: [OpenSIPS-Users] User location replication
When you do the mongo query of the location table, the socket field corresponds to the transport being used by the UAS. That can be used along with the $branch(q) to construct the branch. From: Tito Cumpen Sent: Friday, January 22, 2016 6:46 PM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] User location replication Group, At the previous opensips summit I heard mention of a new module that could aid with user replication to other servers. I am currently using a dns srv aware client and I'd like to know how this would work with active-active scenarios? Say a user exist in several servers. I am currently using a rabbit&mongo solution to keep track of where a user is registered and a mongo query add the branch and fork calls to other servers. Is there another solution to this? I am hitting a roadblock due to the necessity to change priority of a branch based on the callee transport type. This requirement is one that is enforced by rtpengine since it cannot deal with parallel branches that require different media attributes. My issue is if the user exist in another server how do I prioritize based on transport without knowing? Thanks, Tito ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] STUN Server Question
Hi, the documentation states “The sockets come from existing SIP sockets or are created”. Based the example in the documentation, “two sockets are defined and two are created”. The stun server will use 4 sockets: a.. socket1 = ip1 : port1 (assuming 5060 inbound/outbound) b.. socket2 = ip1 : port2 (inbound/outbound?) c.. socket3 = ip2 : port1 (assuming 3478 inbound/outbound) d.. socket4 = ip2 : port2 ((inbound/outbound?)) How to determine the port of the two sockets that are created (port2) to allow in a restrictive firewall? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Module EXEC exec command stdout stderr values
Hi list, there seems to be a bit of inconsistency when using a statement such as this: ... exec(“/usr/local/sbin/script.sh”,,”$var(out)”,”$var(err)”); ... if $var(err) had a value from another process it could hold contents from memory. Not a big issue, but when performing a xlog() with those variables, sometimes values show up from unrelated requests. This can be misleading. I think if the function is successful, then $var(err) and $var(out) should be set to empty.___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Message Compression Feedback
Hi, in testing the compression module, these below messages appear in the log when using mc_compress(“1”,bhs,). However, it works (compress/decompress) when the second param is “bhe”. ERROR:compression:check_zlib_rc: not enough room in output buffer ERROR:compression:mc_compress_cb: Compression failed ERROR:compression:wrap_tm_func: compression failed ERROR:compression:check_zlib_rc: not enough room in output buffer ERROR:compression:mc_compress_cb: Compression failed ERROR:compression:wrap_msg_func: compression failed. Probably not requested message ERROR:core:run_raw_processing_cb: failed to run callback When using mc_compress(“0”,bhs,) there is no error message, but on the receiving proxy this error is displayed. ERROR:compression:check_zlib_rc: input data incomplete or corrupted ERROR:compression:mc_decompress: decompression failed ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Realtime monitoring of registered end-points
Is this now compatible with the location table existing in mongodb? Secondly, some mobile clients do not respond to "opensipsctrl ping uri" but can still receive calls, does the module discern this? -Original Message- From: Ionut Ionita Sent: Thursday, October 01, 2015 4:33 AM To: n...@lists.opensips.org ; de...@lists.opensips.org ; OpenSIPS users mailling list Subject: [OpenSIPS-Users] Realtime monitoring of registered end-points *Hello all, I’m glad to announce a new feature that allows OpenSIPS to monitor (via SIP OPTIONS probing) and disable/delete in realtime the registrations which are not responding. The previous approach had two issues when came to so called “zombie” registrations (registrations which are not valid anymore): * resources - such zombie registrations may waste resources in your OpenSIPS server (memory, processing time, DB space, useless NAT pinging, TCP connect attempts); * user experience - using the zombie registration to reach un-existing users translates into useless calls (calls that will simply timeout), giving delays in the call setup (instead of going straight to VM, you may burn 5-10 seconds in trying to reach the user); The main idea behind these features is to delete contacts that do not respond to a certain number of SIP pings. The SIP pinging is provided by the nathelper module which was enhanced to keep the state of each pinging requests (basically, the module is waiting and checking the reply of each SIP request sent to the registered users). For registrations detected as “dead” (not responding), the nathelper module interacts directly with the usrloc module in order to remove the zombie contact. For usage perspective, the nathelper module now has two new parameters: * ping_threshold - timeout to consider a ping as unanswered; * max_pings_lost - the number of unresponded pings after which the contact is removed from usrloc; In order to activate this feature, every contact must have the sipping_bflag (to be pinged) and remove_on_timeout_bflag (to be deleted on no-answer) activated. This means these flags must also be configured in the nathelper module. [0] For full documentation of the nathelper module including the newly added feature see [1]. Any feedback is highly appreciated. Regards, Ionut Ionita [0]http://www.opensips.org/html/docs/modules/2.2.x/nathelper.html#id248011 [1]http://www.opensips.org/html/docs/modules/2.2.x/nathelper.html * ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Restart=always and After=mysql.service
Does it start in the usual way with “opensipsctl start”? If not, try to fix that method of starting, check /etc/opensips/opensipsctrl. Some things you might want to check. With systemd , opensips cannot normally create files in /tmp or /var/run. for your pid file directory, create /var/run/opensips from /etc/rc.d/rc.local on each boot as /var/run is cleaned on shutdown. From: Nabeel Sent: Sunday, September 20, 2015 1:11 AM To: Podrigal, Aron ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Restart=always and After=mysql.service No, I just compiled and installed OpenSIPS in the normal manner while logged in as root. On 20 September 2015 at 06:00, Podrigal, Aron wrote: Are you running this on a container / chrooted env? On Sun, Sep 20, 2015 at 12:39 AM, Nabeel wrote: This is what it says: Sep 20 05:21:45 server1.sipdomain.com systemd[1]: Starting OpenSIPS is a very fast and flexible SIP (RFC3261) server... -- Subject: Unit opensips.service has begun with start-up -- Defined-By: systemd -- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel -- -- Unit opensips.service has begun starting up. Sep 20 05:21:45 server1.sipdomain.com systemd[4022]: Failed at step EXEC spawning /usr/sbin/opensips: Permission denied -- Subject: Process /usr/sbin/opensips could not be executed -- Defined-By: systemd -- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel -- -- The process /usr/sbin/opensips could not be executed and failed. -- -- The error number returned while executing this process is 13. Sep 20 05:21:45 server1.sipdomain.com systemd[1]: opensips.service: control process exited, code=exited status=203 Sep 20 05:21:45 server1.sipdomain.com systemd[1]: Failed to start OpenSIPS is a very fast and flexible SIP (RFC3261) server. -- Subject: Unit opensips.service has failed -- Defined-By: systemd -- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel -- -- Unit opensips.service has failed. -- -- The result is failed. Sep 20 05:21:45 server1.sipdomain.com systemd[1]: Unit opensips.service entered failed state. On 20 September 2015 at 05:12, Podrigal, Aron wrote: Do you see anything informative in your journal after trying to start opensips? sudo journalctl -xn 350 On Sat, Sep 19, 2015 at 11:46 PM, Nabeel wrote: Please see the strace output at these links: http://pastebin.com/G1Uv5s1E http://pastebin.com/gnesw4tW On 20 September 2015 at 04:30, Podrigal, Aron wrote: Can you post the strace output by running sudo strace -ff -o /tmp/opensips_strace_output systemctl start opensips.service cat /tmp/opensips_strace_output* That will help see where the error occurs. On Sat, Sep 19, 2015 at 11:21 PM, Nabeel wrote: Why can't the service file be automatically generated at the time of installation/compilation on the relevant OS? It seems to be quite a basic feature missing. On 20 September 2015 at 04:16, Nabeel wrote: Unfortunately, changing the user/group to 'root' did not work either. The same error occurs and OpenSIPS fails to start: Failed at step EXEC spawning /usr/sbin/opensips: Permission denied Hasn't anyone else seen this on CentOS? On 18 September 2015 at 22:21, sevpal wrote: If you change User= and Group= in the service file to root, you might not see such a message. To run as user opensips, then user opensips must exist on the system; in addition, ownership on files/folders that will be accessed by Opensips needs to be changed. From: Nabeel Sent: Tuesday, September 15, 2015 9:47 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Restart=always and After=mysql.service I tried that service file on CentOS 7 but got the following error when trying to start the service: Failed at step EXEC spawning /usr/sbin/opensips: Permission denied Changing permissions of that directory did not solve the problem: chmod +x /usr/sbin/opensips chmod -R 777 /usr/sbin/opensips Does the service file need modification for CentOS? On 7 September 2015 at 19:14, Nabeel wrote: Hi, Most of my services have a .service file located at /usr/lib/systemd/system where I can set 'Restart=always' or 'After=mysql.service' / 'After=mariadb.service' to make sure the server starts at the appropriate times. However, there is no such
Re: [OpenSIPS-Users] Restart=always and After=mysql.service
If you change User= and Group= in the service file to root, you might not see such a message. To run as user opensips, then user opensips must exist on the system; in addition, ownership on files/folders that will be accessed by Opensips needs to be changed. From: Nabeel Sent: Tuesday, September 15, 2015 9:47 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Restart=always and After=mysql.service I tried that service file on CentOS 7 but got the following error when trying to start the service: Failed at step EXEC spawning /usr/sbin/opensips: Permission denied Changing permissions of that directory did not solve the problem: chmod +x /usr/sbin/opensips chmod -R 777 /usr/sbin/opensips Does the service file need modification for CentOS? On 7 September 2015 at 19:14, Nabeel wrote: Hi, Most of my services have a .service file located at /usr/lib/systemd/system where I can set 'Restart=always' or 'After=mysql.service' / 'After=mariadb.service' to make sure the server starts at the appropriate times. However, there is no such .service file for OpenSIPS. Please advise how to create this .service file for OpenSIPS. I heard about 'respawn' but don't know how exactly to use this and would prefer to use a .service file like other services. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] perl_exec parameter variable issue
Could try a round-about way if you don't need the API and function is
already written in perl:
$var(duration) = $DLG_lifetime;
exec("/bin/perl/update_user_balance
$var(duration)",,"$var(out)","$var(err)");
-Original Message-
From: Bogdan-Andrei Iancu
Sent: Friday, August 07, 2015 1:02 PM
To: OpenSIPS users mailling list ; smont...@twc.com
Subject: Re: [OpenSIPS-Users] perl_exec parameter variable issue
Hi,
the perl_exec() function does not support variables for its arguments.
It is not hard to add, but please open a feature request on the GITHUB
tracker :
https://github.com/OpenSIPS/opensips/issues?page=1&q=is%3Aopen+is%3Aissue
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 07.08.2015 06:04, Admin wrote:
Hi,
When passing a variable to perl_exec as a parameter, i received a
parsing error in the opensips log and won't start. Here is what I am
doing inside the opensips config. file
$var(duration) = $DLG_lifetime;
perl_exec("update_user_balance", $var(duration));
It seems that it wants the parameter in quotes ("$var(duration)") but i
don't get the variable value when i pass it in quotes.
There was a similar problem posted in the archives but no solution was
presented.
Any suggestion on how to work this around is very appreciated. May be
something similar to dialogue lifetime from the OPENSIPS Perl API.
I am running OpenSIPS (2.1.0 (x86_64/linux)).
Thanks.
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Re: [OpenSIPS-Users] Again BLF and Presence with Snom 7xx phones and OpenSips
You need to handle the in-dialog SUBSCRIBE requests. eg:
if has_totag() {
...
if (loose_route()) {
...
} else {
...
if (is_method("SUBSCRIBE")) {
route(2);
exit;
}
...
}
...
}
From: Bogdan-Andrei Iancu
Sent: Thursday, February 26, 2015 7:56 AM
To: OpenSIPS users mailling list ; mailto:michele.pina...@unisi.it
Subject: Re: [OpenSIPS-Users] Again BLF and Presence with Snom 7xx phones and
OpenSips
Hi Michele,
The problem in your script is that you do not handle the sequential (in-dialog)
SUBSCRIBE requests (as you have the second one in your trace, ending with 404
and terminating the subscription).
In the " if ( has_totag() ) " block, you have:
} else {
if (is_method("SUBSCRIBE") && $rd == "127.0.0.1:5060") { # CUSTOMIZE ME
The $rd detection does not cover all your cases, as you configure the presence
module to advertise as SIP contact "sip:prese...@voip.unisi.it:5060". So, the
test fails.
You can adapt the test like:
if (is_method("SUBSCRIBE") && $rd == "voip.unisi.it") { # CUSTOMIZE ME
Or set the contact in presence with the real IP:
modparam("presence", "server_address",
mailto:sip:presence@127.0.0.1:5060)
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.comOn 24.02.2015 12:04, Michele Pinassi wrote:
Hi all,
I'm still stuck on this issue: BLF not working. For example, on my SNOM 760
(ext 5002) i activated BLF for some ext, like 5020. Using SIPGREP i saw:
SUBSCRIBE sip:5...@voip.unisi.it;user=phone SIP/2.0.
Via: SIP/2.0/UDP 172.20.1.10:57286;branch=z9hG4bK-nprg3gvnk4q1;rport.
From: mailto:sip:5...@voip.unisi.it:5060;tag=nyux2omhly.
To: mailto:sip:5...@voip.unisi.it;user=phone.
Call-ID: 3944ec54dc20-pfzjpjhrpm6p.
CSeq: 2 SUBSCRIBE.
Max-Forwards: 70.
Contact: mailto:sip:5002@172.20.1.10:57286;reg-id=1.
Event: dialog.
Accept: application/dialog-info+xml.
User-Agent: snom760/8.7.3.25.9.
Proxy-Authorization: Digest
Expires: 3600.
Content-Length: 0.
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
172.20.1.10:57286;received=172.20.1.10;branch=z9hG4bK-nprg3gvnk4q1;rport=57286.
From: mailto:sip:5...@voip.unisi.it:5060;tag=nyux2omhly.
To:
mailto:sip:5...@voip.unisi.it;user=phone;tag=f315b2d58ae8829149b784764c5a40e3-163d.
Call-ID: 3944ec54dc20-pfzjpjhrpm6p.
CSeq: 2 SUBSCRIBE.
Expires: 3600.
Contact: mailto:sip:prese...@voip.unisi.it:5060.
Server: OpenSIPS (1.11.3-tls (i386/linux)).
Content-Length: 0.
NOTIFY sip:5002@172.20.1.10:57286 SIP/2.0.
Via: SIP/2.0/UDP 172.20.1.2:5060;branch=z9hG4bKdb02.83d58916.0.
To: mailto:sip:5...@voip.unisi.it;tag=nyux2omhly.
From: mailto:sip:5...@voip.unisi.it;tag=f315b2d58ae8829149b784764c5a40e3-163d.
CSeq: 1 NOTIFY.
Call-ID: 3944ec54dc20-pfzjpjhrpm6p.
Max-Forwards: 70.
Content-Length: 147.
User-Agent: OpenSIPS (1.11.3-tls (i386/linux)).
Event: dialog.
Contact: mailto:sip:prese...@voip.unisi.it:5060.
Subscription-State: active;expires=3600.
Content-Type: application/dialog-info+xml.
.
mailto:sip:5...@voip.unisi.it/>
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP 172.20.1.2:5060;branch=z9hG4bKdb02.83d58916.0.
From: mailto:sip:5...@voip.unisi.it;tag=f315b2d58ae8829149b784764c5a40e3-163d.
To: mailto:sip:5...@voip.unisi.it;tag=nyux2omhly.
Call-ID: 3944ec54dc20-pfzjpjhrpm6p.
CSeq: 1 NOTIFY.
Content-Length: 0.
SUBSCRIBE sip:prese...@voip.unisi.it:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.20.1.25:32768;branch=z9hG4bK-lbgnea3kuorq;rport.
From: mailto:sip:5...@voip.unisi.it:5060;tag=w8vp9q5iyn.
To:
mailto:sip:5...@voip.unisi.it;user=phone;tag=f315b2d58ae8829149b784764c5a40e3-29cc.
Call-ID: 54ec3a578c9e-klgn0s3i32zo.
CSeq: 75 SUBSCRIBE.
Max-Forwards: 70.
Contact: mailto:sip:5007@172.20.1.25:32768;reg-id=1.
Event: dialog.
Accept: application/dialog-info+xml.
User-Agent: snom710/8.7.3.25.9.
Expires: 3600.
Content-Length: 0.
SIP/2.0 404 Not here.
Via: SIP/2.0/UDP
172.20.1.25:32768;received=172.20.1.25;branch=z9hG4bK-lbgnea3kuorq;rport=32768.
From: mailto:sip:5...@voip.unisi.it:5060;tag=w8vp9q5iyn.
To:
mailto:sip:5...@voip.unisi.it;user=phone;tag=f315b2d58ae8829149b784764c5a40e3-29cc.
Call-ID: 54ec3a578c9e-klgn0s3i32zo.
CSeq: 75 SUBSCRIBE.
Server: OpenSIPS (1.11.3-tls (i386/linux)).
Content-Length: 0.
NOTIFY sip:5002@172.20.1.10:57286 SIP/2.0.
Via: SIP/2.0/UDP 172.20.1.2:5060;branch=z9hG4bKdbe9.7966c706.0.
To: mailto:sip:5...@voip.unisi.it;tag=iklb1qjh1v.
From: mailto:sip:5...@voip.unisi.it;tag=f315b2d58ae8829149b784764c5a40e3-b571.
CSeq: 2 NOTIFY.
Call-ID: ee35ec54a72b-draf1nwo4qn7.
Max-Forwards: 70.
Content-Length: 0.
User-Agent: OpenSIPS (1.11.3-tls (i386/linux)).
Event: dialog.
Contact: mailto:sip:prese...@voip.unisi.it:5060.
Subscription-State: terminated;reason=timeout.
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP 172.20.1.2:5060;branch=z9hG4bKdbe9.7966c706.0.
From: mailto:sip:5...@voip.unisi.i
Re: [OpenSIPS-Users] Last Modified AOR for User
Thank you, I have it working. To complete the process though, I also needed to
create a bubble sort in the script to sort the Json keys on the last_modified
field.
I’d like to make a feature request for a sort facility in the raw query.
From: Vlad Paiu
Sent: Tuesday, February 10, 2015 4:54 AM
To: sevpal ; OpenSIPS users mailling list ; Bogdan-Andrei Iancu
Subject: Re: [OpenSIPS-Users] Last Modified AOR for User
Hello,
Indeed, date fields where not properly supported in the cachedb_mongodb module.
I've just committed a fix for this - so please update your GIT sources to the
latest ones.
I've just tested this with a small script like :
if (!save("location"))
sl_reply_error();
sleep("1");
cache_raw_query("mongodb:instance1","{ \"op\" : \"find\",
\"ns\" : \"location.location\", \"query\": {\"username\" : \"vlad\"}
}","$avp(mongo_result)");
$json(json_res) := $avp(mongo_result);
$avp(expiration) = $json(json_res/expires)-$Ts;
$avp(last_edited) = $Ts-$json(json_res/last_modified);
xlog("Username vlad will expired in $avp(expiration) seconds
and was edited $avp(last_edited) seconds ago \n");
Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com On 10.02.2015 04:26, sevpal wrote:
The data in the DB are stored correctly, there aren’t any problems there, I
can view all the fields properly set with “db.location.find()” . On a raw Query
though, all the other fields return with Json data except the date/time fields.
It seems mongo needs a different kind of directive than what is currently in
the module to retrieve the data.
From: Bogdan-Andrei Iancu
Sent: Monday, February 09, 2015 4:24 AM
To: sevpal ; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Last Modified AOR for User
Hi Jalung,
If you list the records in mongoDB collection, do you see the "last_modified"
field properly set ? (I'm trying to understand if you have a problem with the
data in DB or with the query itself)
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.comOn 07.02.2015 01:47, sevpal wrote:
If I’m storing the location table in a mongodb collection, how to query the
most current AOR for a user? I can query all the fields except the date/time
fields eg; “last_modified”, they return empty. I’m doing this using the mongo
raw query in Opensips.
Jalung
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Re: [OpenSIPS-Users] Last Modified AOR for User
The data in the DB are stored correctly, there aren’t any problems there, I can view all the fields properly set with “db.location.find()” . On a raw Query though, all the other fields return with Json data except the date/time fields. It seems mongo needs a different kind of directive than what is currently in the module to retrieve the data. From: Bogdan-Andrei Iancu Sent: Monday, February 09, 2015 4:24 AM To: sevpal ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Last Modified AOR for User Hi Jalung, If you list the records in mongoDB collection, do you see the "last_modified" field properly set ? (I'm trying to understand if you have a problem with the data in DB or with the query itself) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.comOn 07.02.2015 01:47, sevpal wrote: If I’m storing the location table in a mongodb collection, how to query the most current AOR for a user? I can query all the fields except the date/time fields eg; “last_modified”, they return empty. I’m doing this using the mongo raw query in Opensips. Jalung ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Last Modified AOR for User
If I’m storing the location table in a mongodb collection, how to query the most current AOR for a user? I can query all the fields except the date/time fields eg; “last_modified”, they return empty. I’m doing this using the mongo raw query in Opensips. Jalung ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
