Re: [OpenSIPS-Devel] One session per account

2016-03-15 Thread Eric Tamme

yes

http://www.opensips.org/html/docs/modules/devel/registrar.html#id293748


On 03/15/2016 05:51 AM, Anderson Catao wrote:

Hello !

I would like to know if I can limit the number of register per account. I
want that one account can just logging in one device.


Thanks !



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Re: [OpenSIPS-Devel] SIP 302 REDIRECT SERVER

2016-03-09 Thread Eric Tamme

http://www.opensips.org/Documentation/Tutorials-Redirect

On 03/09/2016 11:15 AM, acatao wrote:

Hello guys !!! I'm trying to configure my opensisps as a redirect server. I
have my opensips installed and working well. Now what I need to do is
configure my opensips.conf to works as a redirect server. What I want is
something similar to this:

-> INVITE : sip:1234@myserver:5060
<- 100 TRYING
<- 302 Redirect (URI sip:551234@myserver:5060)

..

Somebody can tell me some way to do this ?


Thanks !!!



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Re: [OpenSIPS-Devel] [opensips] segfault on startup with latest master eaef57e (#772)

2016-01-26 Thread Eric Tamme
Hey Raz,

Im actually not 100% sure what happened.  I did a git clean -fxd and rebuilt 
from master and things seem to be working fine, so I can only assume it was not 
a clean build.  Since I can no longer reproduce, I will close the ticket.  

I was using F_MALLOC.

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Re: [OpenSIPS-Devel] [opensips] segfault on startup with latest master eaef57e (#772)

2016-01-26 Thread Eric Tamme
Closed #772.

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[OpenSIPS-Devel] [opensips] segfault on startup with latest master eaef57e (#772)

2016-01-25 Thread Eric Tamme
on commit  eaef57e  

When i try to start opensips it segfaults immediately.

here are logs: https://gist.github.com/etamme/224380afca240940c525

here is a core dump, and the executable which generated the core:

https://owncloud.uphreak.com/index.php/s/wLFMXfRaRCdE7RG

https://owncloud.uphreak.com/index.php/s/aBaZJb2mlzRcEUx

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Re: [OpenSIPS-Devel] [opensips] Memory leak/out of memory when using proto_wss (#743)

2016-01-14 Thread Eric Tamme
Closed #743.

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Re: [OpenSIPS-Devel] [opensips] Memory leak/out of memory when using proto_wss (#743)

2016-01-14 Thread Eric Tamme
yes - memory is completely stable after several days of usage, i will close the 
issue.

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[OpenSIPS-Devel] [opensips] abandoned turn causes processes to sleep forever when handling retransmitted publishes (#754)

2016-01-13 Thread Eric Tamme
This exists on latest master 055f4b1 

Scenario:  2 PUBLISHES with the same E-Tag enter 
https://github.com/OpenSIPS/opensips/blob/master/modules/presence/presentity.c#L546
 as a result of a retransmission.

The first publish finds the presentity, increments last turn to say 5 for the 
sake of argument, then it enters the while loop, finds the presentity and 
continues processing updating it with a new e-tag and saving it the db releases 
locks and finishes.

The second publish enters because its turn matches 5, it finds the presentity 
and increments last turn to 6, it then enters the while loop, but it fails to 
find the presentity because the E-Tag has been changed by the previous publish. 
 It continues processing, but it never increments turn before exiting.  

This causes turn to stay at 6, which will cause the process to lockup forever, 
waiting for its turn, on the next publish received.

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Re: [OpenSIPS-Devel] [opensips] Memory leak/out of memory when using proto_wss (#743)

2016-01-12 Thread Eric Tamme
Awesome Raz!  I'll do a pull/rebuild and let you know.

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Re: [OpenSIPS-Devel] [opensips] Memory leak/out of memory when using proto_wss (#743)

2016-01-12 Thread Eric Tamme
This definitely seems to help, after 145 connections, all pids are using 
~50 bytes where as they were over 150 before - also sending sigusr1 to 
get memory status shows no ws handshake memory usage 
https://gist.github.com/etamme/7470c3e677bd2d947089

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Re: [OpenSIPS-Devel] [opensips] Memory leak/out of memory when using proto_wss (#743)

2016-01-11 Thread Eric Tamme
Hey Raz,

I repeated my previous test: open few tabs, close them, repeat. but I dumped 
the statistics, and did a list_tcp_conns between each test.  All the tcp conns 
seem to tear down properly, but pkg memory use does grow over time.  As you can 
see by the connection number in my gist, I had to open 144 connections to get 
it to fail this time, but i could see the pkg mem increasing across several 
processes each time i did a fifo getstatists.

Here is the gist: https://gist.github.com/etamme/f2a1c3da0706e1d4ee8c

I had a single tab open with a single wss registration, I dumped the stats and 
did a list_tcp_cons, then I opened a new tab which failed to handshake because 
the pid ran out of memory.



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Re: [OpenSIPS-Devel] [opensips] Memory leak/out of memory when using proto_wss (#743)

2016-01-11 Thread Eric Tamme
I am only using 2 children.  

I will try to get it to run out of memory again and see how many active 
connections there are.

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Re: [OpenSIPS-Devel] [opensips] Memory leak/out of memory when using proto_wss (#743)

2016-01-11 Thread Eric Tamme
I recompiled with -DBG_MALLOC as suggested by liviu.

I started the proxy, opened a few tabs to register webrtc clients over wss, 
closed the tabs and repeated the process till the proxy ran out of memory.  I 
then sent sigusr1 signals to all the pids and got the following log output.

https://gist.github.com/etamme/d1d96c08ea68f9873e40

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Re: [OpenSIPS-Devel] [opensips] Memory leak/out of memory when using proto_wss (#743)

2016-01-08 Thread Eric Tamme
Hey Raz,

I cought opensips when it ran out of memory again and sent SIGUSR1 to all the 
PID's to get them to dump memory.

Here is the log before the memory dump, showing the out of memory

```
Jan  8 13:52:57 voipphreaker /usr/local/opensips/sbin/opensips[12073]: 
INFO:core:probe_max_sock_buff: using snd buffer of 509 kb
Jan  8 13:52:57 voipphreaker /usr/local/opensips/sbin/opensips[12073]: 
INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 41
Jan  8 13:52:59 voipphreaker /usr/local/opensips/sbin/opensips[12065]: 
WARNING:core:fm_malloc: Not enough free memory (106904), will attempt 
defragmentation
Jan  8 13:52:59 voipphreaker /usr/local/opensips/sbin/opensips[12065]: 
ERROR:core:fm_malloc: Defragmentation unsuccessful
Jan  8 13:52:59 voipphreaker /usr/local/opensips/sbin/opensips[12065]: 
ERROR:proto_wss:ws_server_handshake: No more mem for dynamic con request buffer
Jan  8 13:52:59 voipphreaker /usr/local/opensips/sbin/opensips[12065]: 
ERROR:proto_wss:wss_read_req: cannot complete WebSocket handshake
```

and here is the memory dump 

https://gist.github.com/etamme/050e335725a19c0d9e60

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Re: [OpenSIPS-Devel] [opensips] Memory leak/out of memory when using proto_wss (#743)

2016-01-08 Thread Eric Tamme
A little more info - the proxy seems to be handling other protocol (UDP) 
messages without issue.  The specific PID is the only one that appears to be 
having some issues.  All UDP traffic is working without issue.

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Re: [OpenSIPS-Devel] [opensips] Memory leak/out of memory when using proto_wss (#743)

2016-01-06 Thread Eric Tamme
Hey Raz,

I did not add any log output for debugging - this is vanilla master runing at 
debug 4.

I have not tried running with memory debugging - i will compile with that 
turned on.


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[OpenSIPS-Devel] [opensips] repeated proto_wss error messages in logs (#741)

2016-01-05 Thread Eric Tamme
on master commit 968a4e004ebf85687f864b885a1ad5dc6632f41a

I've enabled proto_wss and have a client registered.  I see the following 
proto_wss errors in the logs on a repeating basis - this is at debug 4.

https://gist.github.com/etamme/ef1a049ffd50fd67ea6e#file-proto_wss-errors-L25

No immediate issue is apparent, just the error in the logs.

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[OpenSIPS-Devel] [opensips] failing to loose route sequential requests that are proto wss (#742)

2016-01-05 Thread Eric Tamme
On master commit 968a4e004ebf85687f864b885a1ad5dc6632f41a

If I place a call from a WSS client to a UDP client, then the UDP client hangs 
up, opensips will show the following log indicating that it cant identify the 
protocol.
```
Jan  5 12:35:23 voipphreaker /usr/local/opensips/sbin/opensips[9885]: 
DBG:rr:after_loose: Topmost route URI: 
'sip:159.203.234.107;r2=on;lr;ftag=s6992f0ms9;did=e6a.94fb3f15' is me
Jan  5 12:35:23 voipphreaker /usr/local/opensips/sbin/opensips[9885]: 
WARNING:core:grep_sock_info: unknown proto 9
Jan  5 12:35:23 voipphreaker /usr/local/opensips/sbin/opensips[9885]: 
WARNING:rr:after_loose: no socket found to match 2nd RR [9][159.203.234.107:443]
```

I dug around in sock_info.c and net/trans.c to find how the protos list was 
populated, but I'm not able to immediately see whats wrong.  get_sock_info_list 
appears to be return 0, but it looks like the wss proto is in the list as proto 
9 as the log indicates, so I'm not clear on whats going on.

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Re: [OpenSIPS-Devel] [opensips] failing to loose route sequential requests that are proto wss (#742)

2016-01-05 Thread Eric Tamme
This has been tracked down to parse_uri.c not parsing wss - looking into how to 
fix it now

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Re: [OpenSIPS-Devel] [opensips] failing to loose route sequential requests that are proto wss (#742)

2016-01-05 Thread Eric Tamme
im close on a patch that fixes this, but im having trouble because WS and WSS 
are the same letters up till the last S.  Im not sure how to avoid the 
transport_fin call for WS as i basically need to look ahead.

here is my current diff https://gist.github.com/etamme/953cb286cb73b6a62216
It you uncomment 

```
//transport_fin(VW_S_FIN, PROTO_WS);
```

It will fail to compile, but i tested it as is with WSS and got clean loose 
routing on a sequential request towards a WSS client.

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[OpenSIPS-Devel] [opensips] avpops checks for async functions on 1.11 (#738)

2015-12-29 Thread Eric Tamme
Looks like commit 398b029b6766f9d54b6d518f92892a124bfeaceb added a check in avp 
init that checks for async function calls.

https://github.com/OpenSIPS/opensips/commit/398b029b6766f9d54b6d518f92892a124bfeaceb#diff-3aae073954d799e2ed6f034235e1b5deR222

This breaks compilation as async doesnt exist in 1.11

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[OpenSIPS-Devel] Chrome no longer allows insecure GetUserMedia requests

2015-12-04 Thread Eric Tamme
"getUserMedia() no longer works on insecure origins. To use this 
feature, you should consider switching your application to a secure 
origin, such as HTTPS. See https://goo.gl/rStTGz for more details."


This means that you can no longer do any WebRTC work with chrome and 
OpenSIPS because neither Chrome nor firefox will allow a WS connection 
outbound from an HTTPS page.


I know WSS is in the works, but I just wanted to pass this along.

-Eric

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Re: [OpenSIPS-Devel] [opensips] running out of pkg_mem (#651)

2015-10-19 Thread Eric Tamme
hey @bogdan-iancu I actually have an entirelog from startup to shutdown of this 
happening - its not very verbose though.

basically i set up a cronjob to search the log fork pkg mem warnings - as soon 
as it detects one it kills opensips and makes a back up of the log, then starts 
opensips again, this is kind of a cheap way to keep things running as well as 
catch it in the act.

The full log is available at the following link:

https://owncloud.uphreak.com/index.php/s/ql8qPgJD2vEqtTQ

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[OpenSIPS-Devel] [opensips] running out of pkg_mem (#651)

2015-09-25 Thread Eric Tamme
running on master commit 72926c21780712fabd00a86dafeb292fde656b28


Every few days my https://github.com/etamme/federated-sip server runs out of 
pkg_mem.


It is a very low traffic box so I definitely think something is leaking some 
where.  I will post logs the next time it happens.

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[OpenSIPS-Devel] [opensips] proto_ws fails to build on master (#639)

2015-09-16 Thread Eric Tamme
gist of the issue https://gist.github.com/etamme/f5feeddc0cbdda98fefb

I am on master commit 72926c21780712fabd00a86dafeb292fde656b28

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Re: [OpenSIPS-Devel] [opensips] proto_ws fails to build on master (#639)

2015-09-16 Thread Eric Tamme
local issue - fixed with git clean -fxd

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Re: [OpenSIPS-Devel] [opensips] proto_tls fails build on master (#638)

2015-09-16 Thread Eric Tamme
Closed #638.

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Re: [OpenSIPS-Devel] [opensips] proto_ws fails to build on master (#639)

2015-09-16 Thread Eric Tamme
Closed #639.

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[OpenSIPS-Devel] [opensips] proto_tls fails build on master (#638)

2015-09-16 Thread Eric Tamme
gist of error: https://gist.github.com/etamme/e6044cdbd3925c3c8ed6

Im on master commit 72926c21780712fabd00a86dafeb292fde656b28

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Re: [OpenSIPS-Devel] [opensips] proto_tls fails build on master (#638)

2015-09-16 Thread Eric Tamme
local issue - fixed with git clean -fxd

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[OpenSIPS-Devel] [opensips] tls parameter errors on master (#640)

2015-09-16 Thread Eric Tamme
gist of log errors https://gist.github.com/etamme/6a5b5b86b6f839a00fc8

on master commit 72926c21780712fabd00a86dafeb292fde656b28

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[OpenSIPS-Devel] [opensips] sqlite avpdb queries are broken (#625)

2015-08-28 Thread Eric Tamme
master
commit: 079de59850d9b9e199c5ff5285ba11600a8d2faf

avp_dbquery using sqlite is broken.

see the following gist:
https://gist.githubusercontent.com/etamme/262ff6c7aa2c4889b2af/raw/7a445ee329784e467623eaa7401cc9537671309b/gistfile1.txt

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Re: [OpenSIPS-Devel] [opensips] RTPEngine SDP Re-Invite Issue (#542)

2015-06-04 Thread Eric Tamme
Do not delete the sesssion.  Simply answer/offer again.  RTPengine will 
update the monologues.  The session is still the same dialog so don't 
delete it.  I am using RTPengine and have no issues with re-invites.


see my my federated SIP server config 
https://github.com/etamme/federated-sip/blob/master/core/opensips.cfg.erb 
for example usage.


-Eric

On 06/04/2015 03:54 AM, apsaras wrote:


Hello

I am not sure if that is a feature that has to be implemented or a 
feature that is not working.


I am using RTPEngine for RTP handling on a mhomed environment. 
Initialy I was tried to use the rtpengine_manage which didn't work for 
me so now I am using offer, delete and answer.


The problem I have now is when the carrier or customer sends re-invite 
to renegotiate codecs in fax transmission. How should I handle such 
case? If not handled, carrier's Public RTP IP is passed to the 
internal interface back to customers on the internal network and the 
OK coming from customer to OpenSIPs with the Internal SDP of the 
customer, is forwarded as it is to the carrier.


Thank you for your help

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[OpenSIPS-Devel] [opensips] do not always retry write to pipe if we get blocking errors (#519)

2015-05-21 Thread Eric Tamme
If you have an rabbitmq event subscription the event_rabbitmq module will 
shm_alloc rmq events and write pointers of the structs to the pipe.

In the event that the node you have connected to goes down, the pipe will start 
to fill and when it reaches its max capacity (65535 bytes) or approximately 
2700 events based on the 8 byte pointer, the write call with return EAGAIN 
causing the while loop to become an infinite loop until pointers start getting 
pulled off the pipe.  This causes massive CPU consumption, as well as blocking 
any process that generates an event to event_rabbitmq.

Attempts to publish to rabbit MQ timeout after 3 minutes based on a default 
system tcp timeout, so only one event every 3 minutes will be pulled from the 
pipe while the amqp node is down.

You can recreate this issue by setting up a proxy with an event_rabbitmq 
subscription, adding an iptables rule to block access to the amqp node and send 
traffic to the proxy that would generate an event till you hit the max pipe 
size ~2703 events pointers.

This commit changes the logic to simply retry the write 3 times, then abort.


You can view, comment on, or merge this pull request online at:

  https://github.com/OpenSIPS/opensips/pull/519

-- Commit Summary --

  * do not always retry write to pipe if we get blocking errors as this causes 
the proxy to lock up and consume cpu

-- File Changes --

M modules/event_rabbitmq/rabbitmq_send.c (3)

-- Patch Links --

https://github.com/OpenSIPS/opensips/pull/519.patch
https://github.com/OpenSIPS/opensips/pull/519.diff

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Re: [OpenSIPS-Devel] [opensips] (discussion only) uri_resolve method build dest uri with SRV support (#483)

2015-05-01 Thread Eric Tamme
I agree with points 1-3, as well as the follow on that it would allow for 
multiple attempts against weighted SRV records.

I am not sure I follow that we need to know the destination protocol/address 
family before the main request route - I would say that would not be useful as 
the request could end up forking via append_branch mechanisms.  Knowing in 
branch_route IS crucial, and any processing that is happening based on a 
destination should happen in the branch route father than the request route 
block.

In the case of failure route, you need to know what AF/protocol you had 
attempted, but this can be achieved via transactional storage mechanisms: avp, 
flags etc.  If you t_relay from failure_route it will hit the branch route 
again for further branch processing.

I assume in your example that in the failure route you have modified plist to 
remove the previously attempted protocol from the list before passing it to 
uri_resolve?

It seems like a reasonable approach to me in general.

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Re: [OpenSIPS-Devel] [opensips] resolve inet6 records for hosts and prefer them if dns_try_ipv6 is set (#474)

2015-04-22 Thread Eric Tamme
very much looking forward to discussion/merging of this commit :)

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Re: [OpenSIPS-Devel] [opensips] resolve inet6 records for hosts and prefer them if dns_try_ipv6 is set (#474)

2015-04-22 Thread Eric Tamme
It looks like the commit actually attempts both - ipv4 first, then if ipv6 is 
enabled do ipv6.  I think @bogdan-iancu is right and the logic should be serial 
as follows:

 if (ipv6_preferred==true){
   do IPV6 lookup
   if (no IPV6 result){
 do IPV4 lookup
   }
} else {
   do IPV4 lookup
   if (no IPV4 result  IPV6 enabled) {
 do IPV6 lookup
   }
}

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Re: [OpenSIPS-Devel] [opensips] ERROR:proto_tls:_tls_read: SYSCALL error - (0) Success (#470)

2015-04-20 Thread Eric Tamme
Closed #470.

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[OpenSIPS-Devel] [opensips] ERROR:proto_tls:_tls_read: SYSCALL error - (0) Success (#470)

2015-04-17 Thread Eric Tamme
Using master from commit 24341a06cd18ed32f4d818729122e154e7db520f I am getting 
odd error logs from the TLS module.


Apr 17 11:04:08 alpha /usr/local/sbin/opensips[18512]: 
ERROR:proto_tls:_tls_read: SYSCALL error - (0) Success
Apr 17 11:04:08 alpha /usr/local/sbin/opensips[18512]: 
ERROR:proto_tls:_tls_read: TLS connection to 50.198.204.238:42846 read failed
Apr 17 11:04:08 alpha /usr/local/sbin/opensips[18512]: 
ERROR:proto_tls:_tls_read: TLS read error: 5


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[OpenSIPS-Devel] [opensips] ERROR:db_sqlite:db_sqlite_insert: insert query failed unable to open database file (#466)

2015-04-16 Thread Eric Tamme
db_sqlite is not able to insert, but it can make select queries.

# ls -alh /var/db/opensips
-rw-r--r-- 1 opensips opensips 19K Apr 16 13:12 /var/db/opensips

full debug trace of usrloc failing to insert contact after successful 
registration 

https://gist.githubusercontent.com/etamme/bd7365f77f0ea0919e61/raw/393af1d236488706b6bc8096ce38423585abfc0c/gistfile1.txt

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Re: [OpenSIPS-Devel] [opensips] db_sqlite errors, and segfaults (#458)

2015-04-16 Thread Eric Tamme
the new refactor has resolved this

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Re: [OpenSIPS-Devel] [opensips] db_sqlite errors, and segfaults (#458)

2015-04-16 Thread Eric Tamme
Closed #458.

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Re: [OpenSIPS-Devel] [opensips] ERROR:db_sqlite:db_sqlite_insert: insert query failed unable to open database file (#466)

2015-04-16 Thread Eric Tamme
Closed #466.

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Re: [OpenSIPS-Devel] [opensips] ERROR:db_sqlite:db_sqlite_insert: insert query failed unable to open database file (#466)

2015-04-16 Thread Eric Tamme
I found out that sqlite actually needs the directory to be writeable by the 
process owner as well as the database file it self. 

changing permissions on the directory that the db lives in fixes this issue

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Re: [OpenSIPS-Devel] [opensips] Incoming TLS request gets outbound proto set to TLS (#452)

2015-04-09 Thread Eric Tamme
This issue is only occuring when there is a route header, an example route 
header from on of the UA experiencing the issue is:

Route: sip:alpha.uphreak.com:5061;lr


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Re: [OpenSIPS-Devel] [opensips] Added pike mi command for unblocking IP addresses (#456)

2015-04-09 Thread Eric Tamme
This came up when I was doing some rollout to production with pike module.  
Some actual customers were getting pike blocked, and I wanted to unblock them.  
Jarrod decided to code it up after we talked about it.

We ended up having to put an iptables rule in to drop the customer entirely, 
then remove it, then drop them, then remove it until we could get over the pike 
threshold.

I think pike_rm should be that literal - this is not a soft unblock, it is when 
you want to remove the ip in the pike tree.

If there was a soft unblock command, then I agree it would be better to reset 
counters etc.  But for the situation where traffic has been blocked that you do 
not want blocked - I want to remove the ip from the tree.

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Re: [OpenSIPS-Devel] [opensips] Added pike mi command for unblocking IP addresses (#456)

2015-04-09 Thread Eric Tamme
Awesome!

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[OpenSIPS-Devel] [opensips] db_sqlite segfaults on avb_dbquery 100% of the time (#455)

2015-04-08 Thread Eric Tamme
back trace from core file: 
https://gist.githubusercontent.com/etamme/f19270e0757a07519af8/raw/022530cc5b1628f668b3e3a6c9f8f67c88b41460/db_sqlite

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Re: [OpenSIPS-Devel] [opensips] 2.x sqlite module errors out: ERROR:db_sqlite:str_strstr: bad parameters (#451)

2015-04-08 Thread Eric Tamme
Closed #451.

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Re: [OpenSIPS-Devel] [opensips] 2.x sqlite module errors out: ERROR:db_sqlite:str_strstr: bad parameters (#451)

2015-04-07 Thread Eric Tamme
I looked into this further last night and found that 
db_sqlite_submit_dummy_query() is never actually being called, and as a result 
query_holder is never set.

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Re: [OpenSIPS-Devel] [opensips] Incoming TLS request gets outbound proto set to TLS (#452)

2015-04-07 Thread Eric Tamme
I believe this is related to the initial request coming in with a port 5061 on 
it, even though I strip that by setting $rp=;

Here is a trace from a UA that does not include the port on the initial request 
line:

https://gist.githubusercontent.com/etamme/13ec00e78d585083100f/raw/ad5120e10ce86e6930785b4a6dfe2db8c8532bc3/gistfile1.txt

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[OpenSIPS-Devel] [opensips] Incoming TLS request gets outbound proto set to TLS (#452)

2015-04-07 Thread Eric Tamme
I am making an initial request from a TLS UA to a off network domain that 
should simply t_relay per RFC3263.

The request has no port, and is not a sips URI so I do not know why it is being 
assumed that it has a proto set.

See full debug 4 trace at:

https://gist.githubusercontent.com/etamme/f776aa8c9045358c2b17/raw/b3e9211a3f16a429f9125f04cd2540d0624334fb/proto

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[OpenSIPS-Devel] [opensips] 2.x sqlite module errors out: ERROR:db_sqlite:str_strstr: bad parameters (#451)

2015-04-06 Thread Eric Tamme

Full debug log is available at:
https://gist.githubusercontent.com/etamme/b8148792f97746956f12/raw/e7eb57af6b1e6f943c63eddbbc5fbf8113cef783/sqlite%20errors

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Re: [OpenSIPS-Devel] [opensips] 2.x sqlite module errors out: ERROR:db_sqlite:str_strstr: bad parameters (#451)

2015-04-06 Thread Eric Tamme
I added some logs for query_holder and searched_str, it looks like query holder 
is empty 

DBG:avpops:ops_dbquery_avps: query [select tran_strip, tran_prefix, tran_user, 
tran_domain, tran_port, tran_add_header,tran_strip_port from translations where 
(from_username='eric' or from_username IS NULL) and 
from_domain='voipphreaker.com' and (match_user_agent like 'Grandstream GXP1610 
1.0.0.38%' or match_user_agent IS NULL )  and ('7000' regexp match_regex or 
match_regex IS NULL) order by tran_priority limit 1]
 DBG:db_sqlite:db_copy_rest_of_count: XXX query_holder: 
 DBG:db_sqlite:db_copy_rest_of_count: XXX searched_str: from
 ERROR:db_sqlite:str_strstr: bad parameters


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Re: [OpenSIPS-Devel] [opensips] SQLite3 database support - Feature Request (#409)

2015-04-01 Thread Eric Tamme
Is it possible to include the sqlite-pcre stuff with some build flag, that way 
the REGEX command is exposed.

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[OpenSIPS-Devel] [OpenSIPS/opensips] f1ffe7: fixed flex sub version check for isatty definition...

2015-03-31 Thread Eric Tamme
  Branch: refs/heads/2.1
  Home:   https://github.com/OpenSIPS/opensips
  Commit: f1ffe775cb49f9c8de2027b925bdbadfdef30a01
  
https://github.com/OpenSIPS/opensips/commit/f1ffe775cb49f9c8de2027b925bdbadfdef30a01
  Author: Eric Tamme e...@uphreak.com
  Date:   2015-03-31 (Tue, 31 Mar 2015)

  Changed paths:
M cfg.lex

  Log Message:
  ---
  fixed flex sub version check for isatty definition - isatty was removed in .36

(cherry picked from commit 4015747f3d0771e2f71cc7897fbd28dd3e49de9f)


  Commit: e696f6f776159030cf17a42bdb7ac6ca5ca28287
  
https://github.com/OpenSIPS/opensips/commit/e696f6f776159030cf17a42bdb7ac6ca5ca28287
  Author: Eric Tamme e...@uphreak.com
  Date:   2015-03-31 (Tue, 31 Mar 2015)

  Changed paths:
M cfg.lex

  Log Message:
  ---
  fixed flex sub version check to include unistd.h for anything subversion 36 
or higher

(cherry picked from commit 9393492307b85a91031a5f4f35338eac44af4f5d)


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[OpenSIPS-Devel] [opensips] lex.yy.c missing unistd.h for isatty definition with flex 2.5.37 (#443)

2015-03-30 Thread Eric Tamme
lex.yy.c is generated without a include for unistd.h, which defines isatty.

Compiling lex.yy.c
lex.yy.c: In function ‘yy_init_buffer’:
lex.yy.c:4154:9: warning: implicit declaration of function ‘isatty’ 
[-Wimplicit-function-declaration]
 b-yy_is_interactive = file ? (isatty( fileno(file) )  0) : 0;
 ^


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[OpenSIPS-Devel] [opensips] fixed flex sub version check for isatty definition (#444)

2015-03-30 Thread Eric Tamme
It looks like the isatty definition was removed from flex in subversions =36. 
This fixes #443 
You can view, comment on, or merge this pull request online at:

  https://github.com/OpenSIPS/opensips/pull/444

-- Commit Summary --

  * fixed flex sub version check for isatty definition - isatty was removed in 
.36
  * fixed flex sub version check to include unistd.h for anything subversion 36 
or higher

-- File Changes --

M cfg.lex (2)

-- Patch Links --

https://github.com/OpenSIPS/opensips/pull/444.patch
https://github.com/OpenSIPS/opensips/pull/444.diff

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Re: [OpenSIPS-Devel] [opensips] lex.yy.c missing unistd.h for isatty definition with flex 2.5.37 (#443)

2015-03-30 Thread Eric Tamme
I am on flex version 2.5.37

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[OpenSIPS-Devel] [opensips] 2.1 TLS read_fd: no fd read bug (#438)

2015-03-19 Thread Eric Tamme
I set up TLS and registered a single UA via tls.  I got the following error.

`Mar 19 14:31:53 alpha /usr/local/sbin/opensips[28491]: 
CRITICAL:core:handle_io: 
 read_fd:no fd read

It seems you have hit a programming bug.
Please help us make OpenSIPS better by reporting it at 
https://github.com/OpenSIPS/opensips/issues
`

see the following 
[gist](https://gist.githubusercontent.com/etamme/5c0fc8f0358a62e082e3/raw/e31140977ffa092da4fd2e141ca6cb07231b/gistfile1.txt)
 for more complete log output.

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Re: [OpenSIPS-Devel] [opensips] $rtpstat pvar in rtpengine is empty (#434)

2015-03-18 Thread Eric Tamme
The function calls that populate the rtpstat pvar are not properly set 
up at the moment.  I will work on getting a pull request together with a 
fix.

-Eric

On 03/18/2015 11:05 AM, Rob Gagnon wrote:


After successful call with rtpengine proxying the RTP, an attempt to 
log, or insert the value of $rtpstat into a header does not work. The 
content is blank.


|...
if (has_totag()) {
if (loose_route() || topology_hiding_match()) {
   if (is_method(INVITE)) {
  record_route();
   }
   if (is_method(CANCEL|BYE)) {
  append_hf(x-rtp-statistics: $rtpstat\r\n);
  rtpengine_delete();
   }
}
route(relay);
} else {
...
}
...
|

The header appears in the BYE, but there is nothing following the colon.

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[OpenSIPS-Devel] [OpenSIPS/opensips] 25d47d: fix rtpstat for rtpengine - thanks to jarrod for t...

2015-03-18 Thread Eric Tamme
  Branch: refs/heads/2.1
  Home:   https://github.com/OpenSIPS/opensips
  Commit: 25d47d85e557030fb7391908c35987da74f2447c
  
https://github.com/OpenSIPS/opensips/commit/25d47d85e557030fb7391908c35987da74f2447c
  Author: Eric Tamme e...@uphreak.com
  Date:   2015-03-18 (Wed, 18 Mar 2015)

  Changed paths:
M modules/rtpengine/rtpengine.c

  Log Message:
  ---
  fix rtpstat for rtpengine - thanks to jarrod for tracking down the fix

(cherry picked from commit d10e6796aa312374592892e58af98a017f6925d5)


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[OpenSIPS-Devel] [opensips] fix rtpstat for rtpengine - thanks to jarrod for tracking down the fix (#435)

2015-03-18 Thread Eric Tamme
This resolves #434
You can view, comment on, or merge this pull request online at:

  https://github.com/OpenSIPS/opensips/pull/435

-- Commit Summary --

  * fix rtpstat for rtpengine - thanks to jarrod for tracking down the fix

-- File Changes --

M modules/rtpengine/rtpengine.c (28)

-- Patch Links --

https://github.com/OpenSIPS/opensips/pull/435.patch
https://github.com/OpenSIPS/opensips/pull/435.diff

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Re: [OpenSIPS-Devel] [OpenSIPS/opensips] 522dc7: update opensips.cfg to 2.1

2015-03-13 Thread Eric Tamme

its not disabled 

On 03/13/2015 07:49 AM, Satish Patel wrote:

Just wonder why disable TCP and TLS in 2.1? Or it will be available in future?

--
Sent from my iPhone


On Mar 13, 2015, at 9:28 AM, Razvan Crainea raz...@opensips.org wrote:

  Branch: refs/heads/master
  Home:   https://github.com/OpenSIPS/opensips
  Commit: 522dc7d654ca35c3a2cad1521c4e9151be9d7522
  
https://github.com/OpenSIPS/opensips/commit/522dc7d654ca35c3a2cad1521c4e9151be9d7522
  Author: Razvan Crainea raz...@opensips.org
  Date:   2015-03-13 (Fri, 13 Mar 2015)

  Changed paths:
M etc/opensips.cfg

  Log Message:
  ---
  update opensips.cfg to 2.1


  Commit: cb427c9019021384d1f4f7bf98848ebd294445ab
  
https://github.com/OpenSIPS/opensips/commit/cb427c9019021384d1f4f7bf98848ebd294445ab
  Author: Razvan Crainea raz...@opensips.org
  Date:   2015-03-13 (Fri, 13 Mar 2015)

  Changed paths:
M cfg.y

  Log Message:
  ---
  relax alias declaration syntax


Compare: 
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[OpenSIPS-Devel] [opensips] expose the DTLS passive flag to allow for passive relay through rtpengine (#429)

2015-03-12 Thread Eric Tamme
not much else to say really.
You can view, comment on, or merge this pull request online at:

  https://github.com/OpenSIPS/opensips/pull/429

-- Commit Summary --

  * expose the DTLS passive flag to allow for passive relay through rtpengine

-- File Changes --

M modules/rtpengine/rtpengine.c (16)

-- Patch Links --

https://github.com/OpenSIPS/opensips/pull/429.patch
https://github.com/OpenSIPS/opensips/pull/429.diff

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Re: [OpenSIPS-Devel] [opensips] invite from WS client to non-websockets client - no invite sent (#426)

2015-03-12 Thread Eric Tamme
I think this was resolved with some other patch.  I can no longer reproduce the 
problem.

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Re: [OpenSIPS-Devel] [opensips] invite from WS client to non-websockets client - no invite sent (#426)

2015-03-12 Thread Eric Tamme
Closed #426.

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Re: [OpenSIPS-Devel] Blox SBC

2015-03-10 Thread Eric Tamme

... where is the source for this?

On 03/10/2015 06:11 AM, Varadhan Work wrote:

Hello,

I'm Varadhan,

We have come up with an idea to build opensource SBC with opensips as 
core SIP session route, thanks to opensips and its community.


Now we have launched beta version of Blox (www.blox.org 
http://www.blox.org)


Blox is a Session Border Controller(SBC) used to control VoIP 
signaling and media streams. SBC is responsible for setting up, 
conducting, and tearing down calls. SBC allows owners to control the 
types of call that can be placed through the networks and also 
overcome some of the problems caused by firewalls and NAT for VoIP 
calls. A common location for a stand-alone SBC is a connection point, 
called a border, between a private local area network (LAN) and the 
Internet. SBC polices real-time voice traffic between IP network 
borders ensuring your private network is robustly secure and fully 
manageable.


You can download and install Blox via www.blox.org 
http://www.blox.org, please post your valuable feedback.


Thanks  Regards,
Varadhan M


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Re: [OpenSIPS-Devel] [opensips] usrloc websockets errors on restart (#427)

2015-03-10 Thread Eric Tamme
Closed #427.

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[OpenSIPS-Devel] [opensips] invite from WS client to non-websockets client - no invite sent (#426)

2015-03-09 Thread Eric Tamme
I am sending an initial invite to opensips from a webrtc user agent, sip.js, to 
a tcp registered user through opensips.

opensips gets the invite from sip.js, sends 100 trying ... then does nothing 
and eventually the call gets timed out in sip.js.

I am pretty sure i got a invite from opensips like .. 5 minutes later some how, 
but this i can not confirm.  the gist is just of opensips getting the call and 
... kinda doing nothing with it.

https://gist.github.com/etamme/7593263139be8e09cad2

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[OpenSIPS-Devel] [opensips] usrloc websockets errors on restart (#427)

2015-03-09 Thread Eric Tamme
on restarting opensips I got the following errors in my log.

```
Mar  9 16:54:18 alpha /usr/local/sbin/opensips[24254]: 
ERROR:usrloc:parse_phostport: bad protocol in ws:104.236.248.128:8080
Mar  9 16:54:18 alpha /usr/local/sbin/opensips[24254]: ERROR:usrloc:dbrow2info: 
bad socket ws:104.236.248.128:8080
Mar  9 16:54:18 alpha /usr/local/sbin/opensips[24254]: 
ERROR:usrloc:preload_udomain: sipping record for test in table location
Mar  9 16:54:18 alpha /usr/local/sbin/opensips[24254]: 
ERROR:usrloc:parse_phostport: bad protocol in ws:104.236.248.128:8080
Mar  9 16:54:18 alpha /usr/local/sbin/opensips[24254]: ERROR:usrloc:dbrow2info: 
bad socket ws:104.236.248.128:8080
Mar  9 16:54:18 alpha /usr/local/sbin/opensips[24254]: 
ERROR:usrloc:preload_udomain: sipping record for test2 in table location
```

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Re: [OpenSIPS-Devel] [opensips] core timer delay warnings on master (#421)

2015-03-05 Thread Eric Tamme
@bogdan-iancu the warnings happen all the time, but I can not see any 
particular interval to them.

I did 'grep timer /var/log/opensips' and made a gist of the last 24 hours 
https://gist.github.com/etamme/a75198a5590c22bfa574

This box is litereally sitting idle - I have one UA that I am registering via 
TCP, but the warnings were occurring before I even enabled tcp.


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Re: [OpenSIPS-Devel] [opensips] Allow TCP without transport=tcp header (#420)

2015-03-05 Thread Eric Tamme
If there is a preference for UDP, and an SRV lookup is done with no result, I 
would still want to fall through and attempt a TCP SRV lookup.  It seems like 
this SHOULD be the way things work.

If there is a parameter to place preference on one protocol vs. the other I 
would still want the fall through behaviour should the preferred protocol not 
get any SRV result.  

If this fall through behaviour would be undesirable for speed purposes to some, 
then perhaps two parameters:  
proto_preference, [UDP|TCP|TLS]
proto_fallthrough, [1|0]

In this case all the existing mechanisms to previously determine the protocol 
would still be in place, and these parameters would only take effect in the 
case where an SRV lookup with an assumed protocol happens.

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[OpenSIPS-Devel] [opensips] core timer delay warnings on master (#421)

2015-03-03 Thread Eric Tamme
running on master commit f3ed4db9903d85cba7d7d491c931b550e29932d1

I am getting repeated logs with the following warning:

 WARNING:core:handle_timer_job: utimer job tm-utimer has a 10 us delay in 
execution

This is a centos7 kvm using kvm clock.  The machine is currently processing no 
traffic at all.

I am running the following config 
https://github.com/etamme/federated-sip/tree/2.1-config/core


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Re: [OpenSIPS-Devel] [opensips] Allow TCP without transport=tcp header (#420)

2015-03-03 Thread Eric Tamme
I looked into resolve.c sip_resolvehost() 
https://github.com/OpenSIPS/opensips/blob/master/resolve.c#L1742

Throughout the function the code is consistent about using tls or udp depending 
on the uri scheme.  If there is no naptr record, then the code makes this 
assumption again, and calls do_srv_lookup() to get the host entity.

The way the code works, it determines the protocol before doing the SRV lookup 
entirely - which is exactly why you are getting 476 - b/c its trying to lookup 
the UDP SRV, which ... doesn’t exist.

I agree that this is frustration, but im not exactly certain how it could be 
fixed.  One idea would be to have some modparam cascade_srv that would 
attempt to determine the proto, as the code does now, but on failure of a udp 
srv lookup, attempt a tcp srv lookup, and potentially lastly a tls srv lookup.  
The tls is a little less ... belonging in this type of feature since it has a 
mechanism of proto detection with a sips uri. 

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Re: [OpenSIPS-Devel] [opensips] RTP Engine transport-protocol bug (#415)

2015-02-18 Thread Eric Tamme
Oops - looks like an oversight on my part.  Thanks for catching it, created a 
merge request with the fix.

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[OpenSIPS-Devel] [opensips] fixed bitmask to cover all 3 bits vs. 1st and 3rd. Thanks to eldonogith... (#416)

2015-02-18 Thread Eric Tamme
fixes issue #415 
You can view, comment on, or merge this pull request online at:

  https://github.com/OpenSIPS/opensips/pull/416

-- Commit Summary --

  * fixed bitmask to cover all 3 bits vs. 1st and 3rd.  Thanks to eldonogithub.

-- File Changes --

M modules/rtpengine/rtpengine.c (2)

-- Patch Links --

https://github.com/OpenSIPS/opensips/pull/416.patch
https://github.com/OpenSIPS/opensips/pull/416.diff

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Re: [OpenSIPS-Devel] [opensips] SQLite3 database support - Feature Request (#409)

2015-02-04 Thread Eric Tamme
OpenSips is the best sip proxy that exists, your arbitrary requirement 
of sqlite is not relevant.  Explain why sqlite is important to you, or 
go away.



On 02/04/2015 08:25 PM, anto wrote:


Sorry. I don't think you can help in this case, because what I want to 
is to have a good SIP Proxy with SQLite database. So far I have been 
using Asterisk but that is not pure SIP Proxy. I hope this is clearly 
explained what I want.


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Re: [OpenSIPS-Devel] [opensips] SQLite3 database support - Feature Request (#409)

2015-02-04 Thread Eric Tamme
A lot of times people think they need to do some thing a certain way.  I 
suggest you say what you want to do, and people will tell you how you 
can do it, instead of you trying to tell people how you want to do some 
thing in a system you seem to know little about.


I saw your post from several years ago - there are already many router 
distros that have embedded opensips instances... so like i said, tell us 
what you are trying to do... not how you want to do it.


On 02/04/2015 08:12 PM, anto wrote:


Thanks. I know there are several database modules supported in 
OpenSIPS. I thought that I was clearly expressing myself that I want 
to use SQLite3 database.


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Re: [OpenSIPS-Devel] [opensips] SQLite3 database support - Feature Request (#409)

2015-02-04 Thread Eric Tamme
You have not answered the question... what are you trying to do, why is 
sqlite important to you.


I answered your initial question - there is no sqlite support, and no 
plans for it.  That said, I am certain opensips can do what you want as 
it is extremely flexible and capable with its existing modules.



On 02/04/2015 08:29 PM, anto wrote:


If you are not a developer or somebody who is competence enough to 
answer my question, I suggest to shut your mouth.


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Re: [OpenSIPS-Devel] [opensips] SQLite3 database support - Feature Request (#409)

2015-02-04 Thread Eric Tamme

ok ... good luck

On 02/04/2015 08:39 PM, anto wrote:


I don't think you are representing the developers of OpenSIPS project. 
So I will just wait for the answer from them.


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[OpenSIPS-Devel] [opensips] added missing force-relay flag option for ICE (#397)

2014-12-18 Thread Eric Tamme
The parse_flags method was missing the option to specify force-relay which is 
a valid ICE value in the rtpengine ng protocol.  Without the force-relay, ice 
candidates are all written in as host candidates.  This change has been tested 
against rtpengine and it does cause ICE relay candidates to be inserted into 
SDP.
You can merge this Pull Request by running:

  git pull https://github.com/etamme/opensips master

Or you can view, comment on it, or merge it online at:

  https://github.com/OpenSIPS/opensips/pull/397

-- Commit Summary --

  * added missing force-relay flag option for ICE

-- File Changes --

M modules/rtpengine/rtpengine.c (2)

-- Patch Links --

https://github.com/OpenSIPS/opensips/pull/397.patch
https://github.com/OpenSIPS/opensips/pull/397.diff

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Re: [OpenSIPS-Devel] Supported RFCs

2014-07-23 Thread Eric Tamme
Many of the RFC's you mention are implemented within user agents, and 
not by proxies.  3262, and 3581 are implemented by proxy servers, and 
opensips implements both.


All the others are support for specific methods, headers, or timers and 
would be implemented by the user agents.


You can certainly use all the other RFC's you mentioned with opensips, 
but again - they would be implemented by user agents.


Might I ask what it is you are interested in more generally?

-Eric


On 07/23/2014 09:02 AM, Conners, James wrote:

RFCs: 3262, 3311, 3515, 3581, 3891, 3966, 4028

-Original Message-
From: devel-boun...@lists.opensips.org 
[mailto:devel-boun...@lists.opensips.org] On Behalf Of Saúl Ibarra Corretgé
Sent: Friday, July 18, 2014 7:59 AM
To: OpenSIPS devel mailling list
Subject: Re: [OpenSIPS-Devel] Supported RFCs

Too many to keep a list, I guess :-)

Which one are you particularly interested in?

On 17 Jul 2014, at 16:59, Conners, James james.conn...@aspect.com wrote:


What RFCs are supported by OpenSIPS?
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[OpenSIPS-Devel] [opensips] fix for RFC5245 15.1 compliant candidate injection (#264)

2014-07-02 Thread Eric Tamme
 lt;rel-addrgt; and lt;rel-portgt; MUST be present for server reflexive, 
peer, and relay candidates
You can merge this Pull Request by running:

  git pull https://github.com/etamme/opensips mediaproxy-rfc5245fix

Or you can view, comment on it, or merge it online at:

  https://github.com/OpenSIPS/opensips/pull/264

-- Commit Summary --

  * fix for RFC5245 15.1 lt;rel-addrgt; and lt;rel-portgt; MUST be present 
for server reflexive, peer, and relay candidates

-- File Changes --

M modules/mediaproxy/mediaproxy.c (8)

-- Patch Links --

https://github.com/OpenSIPS/opensips/pull/264.patch
https://github.com/OpenSIPS/opensips/pull/264.diff

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