>Hi,
>
>I have some questions regarding SerMyAdmin.
>
>1) Is this the only web interface tool to provision users?
I'm not sure, there should be others. CounterPath has one for their softphones.
>
>2) Is a user able to submit a registration and reset his password
>via this tool?
Yes, th
>Message: 3
>Date: Fri, 14 Aug 2009 11:27:06 +1000
>From: "Leon Li"
>Subject: [OpenSIPS-Users] Can't run Opensips after install SerMyAdmin
>To:
>Message-ID:
> <964c943783cdce4bba0ce327aa8e7ba50508d...@vm-a-ex1.ms.aarnet.edu.au>
>Content-Type: text/plain; charset="us-ascii"
>
>Hi,
>
>
>
>After
>From: "Leon Li"
>Subject: Re: [OpenSIPS-Users] Can't run Opensips after install
> SerMyAdmin
>To: "OpenSIPS users mailling list"
>Message-ID:
> <964c943783cdce4bba0ce327aa8e7ba50508d...@vm-a-ex1.ms.aarnet.edu.au>
>Content-Type: text/plain; charset="iso-8859-1"
>
>This is a fresh instal
I'm glad to announce the release of serMyAdmin 1.2.8
Changes since the last version:
- New installation tool
- It uses now the opensips database instead of their own.
- Does not delete and recreate any opensips tables
- Five new modules
serMyAdmin 1.2.8 is available for download at
https://s
>Message: 1
>Date: Mon, 24 Aug 2009 14:02:31 +0530
>From: ram
>Subject: Re: [OpenSIPS-Users] serMyAdmin 1.2.8 released
>To: OpenSIPS users mailling list
>Message-ID:
>
>Content-Type: text/plain; charset="iso-8859-1"
>
>On Fri, Aug 21, 2009 at 4:32 PM,
Hi Yannick
There is an open source software http://oreka.sourceforge.net/ capable not only
to listen, but to record and manage the calls. It requires a separate machine
connected to a SPAN/MONITOR port of the network switch. I have tested both
OREKA and the RTP Proxy feature, it works, you will
Hi,
I got the following error when trying to insert a new user.
Fatal error: Call to undefined method MDB2_Error::execute() in
/var/www/opensips-cp/web/tools/users/user_management/user_management.php
on line 223
Any help would be welcome.
Thanks,
Flavio E. Goncalves
__
New.
>
> Can you please check that you have installed php-pear and MDB2 and
> MDB2_Driver_mysql ?
> Let me know if this solved your problem.
>
> Regards,
> Alex
>
> On 3/19/2010 00:39, Flavio Goncalves wrote:
>> Fatal error: Call to undefined method MDB2
gt; Can you give me some details about the user you are trying to insert ?
> Any special characters involved ?
> I will continue digging ... maybe I find something relevant regarding
> your problem.
>
> Regards,
> Alex
>
> On 3/19/2010 12:12, Flavio Goncalves wrote:
>>
Hi,
I got the following error when trying to insert a new user.
Fatal error: Call to undefined method MDB2_Error::execute() in
/var/www/opensips-cp/web/tools/users/user_management/user_management.php
on line 223
Any help would be welcome.
Thanks,
Flavio E. Goncalves
__
Hi David,
In the last version opensips-cp 3.0 it worked, I have this setup in my
machine. In the new one (4.0) I didn´t test.
Regards,
Flavio E. Goncalves
2010/3/22 David J. :
> Just curious, Whats the sense of this?
>
> Christian Vo wrote:
>> Trying to manually install SerMyAdmin (step by step
New.
>
> Can you please check that you have installed php-pear and MDB2 and
> MDB2_Driver_mysql ?
> Let me know if this solved your problem.
>
> Regards,
> Alex
>
> On 3/19/2010 00:39, Flavio Goncalves wrote:
>> Fatal error: Call to undefined method MDB2
Hi Leon,
It should work, but I haven't tried. It is built using Java6. I was checking
in the Netbeans IDE and I can generate it using Java5 (never tested) if it
matters. In the development environment it actually runs using Jetty, Grails
1.05 and Netbeans 1.6.8. Let me know if you have any issues
Hi Leon,
It should work, but I haven't tried. It is built using Java6. I was checking
in the Netbeans IDE and I can generate it using Java5 (never tested) if it
matters. In the development environment it actually runs using Jetty, Grails
1.05 and Netbeans 1.6.8. Let me know if you have any issues
Hi,
The problem occurs because setflag(22) only traces the requests after being
processed by the proxy while sip_trace traces the requests before.
I’m using the sip trace combined with the dialog module. Try with
trace_dialog() instead of setflag or sip_trace, the dialog module has to be
loaded.
stopped. To prevent a memory leak, the JDBC Driver has been forcibly
> unregistered.
>
>
>
> Any ideas?
>
>
>
> Regards,
>
> Leon
>
>
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Flavio Gonc
Hi,
Please try to insert the data in this format:
my...@maurice.co.uk
Maybe it works. I will check the application to see if something is
hardcoded.
Best regards,
Flavio E. Goncalves
2010/5/7 rocky881
>
> I have installed sermyadmin on server A in the same network and i have my
> SMTP email
Hi Indiver,
There is another option but depends on the phone. Some phones support the
SIP "Join" header as defined in the rfc3911. There are some phones that
support it such as Aastra and Atcom. I haven't checked Polycom and Linksys
but I think they support too. Basically you will need to get th
Hi Twanda,
serMyAdmin is using a customized version of the opensips database with
additional fields. It points to serMyAdmin database. Please change the modparam
of the modules to point the serMyAdmin database.
regards,
Flavio
Date: Thu, 15 Jan 2009 16:48:25 +0200
From: TCB
Subject: [
There is a mistake in the regular expression for long distance calls.
if (uri=~"^sip:1[2-9][1-9]{9}@") should be if (uri=~"^sip:1[2-9][0-9]{9}@")
Flavio E. Goncalves
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/m
Hello, There is working example on
http://www.sermyadmin.org/w/index.php/Asterisk_Integration. I have created my
own database, sermyadmin, but you can easily adapt the example for the opensips
database. It is tested in the version 1.4.x
Flavio___
User
Hi,
Instructions on how to dowload the source code and prepare a development
environment for serMyadmin were posted yesterday in our website,
http://www.sermyadmin.org/w/index.php/Development. You can now easily download,
change the code and build serMyAdmin. We will freeze the features in the
Hi Franz,
Check your database parameters. The sermyadmin will try to connect a local
database using JDBC. The error (Communication Link) is telling that
sermyadmin is unable to connect the database. Did you have any errors
running the installation utility?
Regards,
Flavio E. Goncalves
2010/5/29
Hi Nezdad,
Your database was not updated by the install utility. Please delete your
database, recreate-it using opensipsdbctl create and reinstall the database
modifications. The message below says that you do not have the areacode
field in your database and this field is created by the installati
Hi Mike,
It is possible to implement all PBX functions using OpenSIPS. It is not
easy, it depends on your phones and gateways. They have to support several
RFCs such as RFC3515, RFC3891, RFC3892. Check RFC5359 (
http://tools.ietf.org/html/rfc5359) for more details about call flows. I
have successf
Hi,
Register attacks are now an epidemy. In most cases they are using the
friendly-scanner (svcrack.py) from sipvicious.org. One easy way to
block is to check the user agent for the words "friendly-scanner"and
drop the packets (an attacker could easily change the user agent, but
most of them are j
V.Office
Fone: +554830258590/+554884085000
OpenSIPS Bootcamp (Frankfurt Sep 20-24)
2010/11/3 Kennard White :
> Hi Flavio,
>
> How did you originally detect these register attacks? Are you using the pike
> module or notice them some other way?
>
> Thanks,
> Kennard
>
>
Hi Osiris,
You can use IP authentication on the OpenSIPS using
check_source_address() to bypass the SIP authentication for calls
coming from the Asterisk server.
Best regards,
--
Flavio E. Goncalves
CEO - V.Office
OpenSIPS Bootcamp (Frankfurt Sep 2
Hi David,
It seems that you have inserted the wrong fields in the database. If
you are using from_uri and to_uri, these are the fields to be inserted
in the database instead of caller_id and callee_id.
Regards,
Flavio E. Goncalves
2010/11/5 David Santiago :
> Hi,
>
> I'm trying to generate
Hi Saul,
I did like your solution. My only concern about Pike was to block
legitimate traffic. A SIP dialer can easily get to the pike threshold,
but doing pike_check_req() just for register, options and bye requests
seems to avoid this.
The only "but" is, the attack can also be done using INVIT
Hi,
Is there anyone with experience using rtpproxy_stream2uac command. I'm
trying to use with OpenSIPS 1.6.4 with rtpproxy-1.2.1. I'm getting the
following error:
ERROR:nathelper:rtpproxy_stream: required functionality is not supported by
the version of the RTPproxy running on the selected node.
uac/uas. Otherwise it checks further for the 'dmcaller.8' file, and so on
> until it finds a suitable media file. If it doesn't, it should log an error.
> I guess your problem is that RTPProxy is unable to find a suitable file to
> open.
>
> Regards,
> Razvan
>
&g
lay it to the
> uac/uas. Otherwise it checks further for the 'dmcaller.8' file, and so on
> until it finds a suitable media file. If it doesn't, it should log an error.
> I guess your problem is that RTPProxy is unable to find a suitable file to
> open.
>
>
ably this is a little
> bug. I will dig into this and let you know as soon as I solve the problem.
>
> Regards,
> Razvan
>
>
> On 03/11/2011 04:35 PM, Flavio Goncalves wrote:
>
> Hi Razvan,
>
> I got to make it work using a version downloaded from GIT. There was a
>
wikipedia.org/wiki/GSM#Voice_codecs>I think, patent issues
> here too.
>
> *
> Thanks
> -- Kamen*
>
>
>
> On 11 March 2011 16:41, Flavio Goncalves wrote:
>
>> Razvan,
>>
>> I need to implement this feature with GSM and G729, but I see that this
>
Hi Chris,
I have tried in the last week. Use a normal wav file as the input (8khz, I
have used one generated by the Asterisk record function). The output will be
a file with .0, .8 and if you have the libgsm installed before compiling .3
for GSM. Specify the name of the file in the rtpproxy_stream
Hi Bogdan,
I would have two suggestions for the next releases.
1. Support for attended transfers on B2BUA
2. Support for the RFC6140 Registration for Multiple Phone Numbers in
the Session Initiation Protocol (SIP) (Part of the SIpConnect 1.1
specification).
Flavio E. Goncalves
CEO - SipPulse
ds,
>
> Vlad Paiu
> OpenSIPS Developer
> http://www.opensips-solutions.com
>
>
>
> On 11/06/2012 01:17 PM, Flavio Goncalves wrote:
>>
>> Hi Bogdan,
>>
>> I would have two suggestions for the next releases.
>>
>> 1. Support for attended transfers on
I'm having an issue with timeout notification of rtpproxy.
Usually sessions are disconnected properly with rtpproxy timeout
notification:
INFO:do_timeout_notification: notification timeout sent 1715.2112525045 :
Success (The dialog hash is presented correctly with period .)
However, In some case
Hi
AVPs are tied to transactions, after the processing the avp is cleaned, you
can't test for duplicate messages in this way. I suggest you check the core
functions cache_store() and cache_fetch(). You can save in memory the
call-id of each message ($ci) and if it is duplicated, drop.
Flavio E. G
Hi Miha,
There are two options:
single domain
1. Use the core statement, alias=domain and then use if(uri!=myself) exit;
at some point in the script.
multidomain
2. Load the domain module, insert the domains in the domain table and use
the function is_uri_host_local() to drop foreign messages
F
Hi Mark,
There is no simple way to traverse NAT. Unless all your routers use a non
symmetric NAT, you will have to use rtpproxy or mediaproxy (you can check
this with a stun client). OpenSIPS behind NAT make things even more
complicated. So I suggest that you follow an example with rtpproxy or med
eloperhttp://www.opensips-solutions.com
>
> On 12/13/2012 07:01 PM, Flavio Goncalves wrote:
>
> I'm having an issue with timeout notification of rtpproxy.
>
> Usually sessions are disconnected properly with rtpproxy timeout
> notification:
>
> INFO:do_timeout_not
get the fix_nat_sip() and fix_nat_contact() to
> work. Any pointers with that?
>
> ** **
>
> Regards
>
> Mark
>
> ** **
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Flavio Goncalves
> *Sent
Hi Chen,
You can use dialog profiles. Create a dialog for all calls, set a profile
named rtpproxy each time you call the rtpproxy and then use
get_profile_size to discover how many calls you already have. Drop after a
certain limit.
Flavio E. Goncalves
2013/1/4 microx
> Hi all,
>
> I in
Hi,
I think you should investigate the following rtpproxy commands.
rtpproxy_stream2uac(prompt_name, count),rtpproxy_stream2uas(prompt_name,
count)
It should do the intent.
Flavio E. Goncalves
SipPulse, www.sippulse.com
2013/1/4 Dragomir Haralambiev
> Hello,
>
> I use RTPproxy and latest Ope
Sounds silly,but remove the space before the lines. I'm not sure if this is
the problem, but I remember once having these problems because of the extra
space in front of the statements. DBHOST is certainly a valid command.
Flavio E. Goncalves
SipPulse, www.sippulse.com
Hi Nick,
This setup seems to be wrong. If you are not going to register on Asterisk,
it does not make sense to use host=dynamic. Use host=opensips_ip or
opensips_domain.
Please check the tutorial http://www.opensips.org/Resources/DocsTutAsterisk.
Create the view exactly in the way presented.
I'm
tbound trunking always worked perfectly.
>
>
> Thanks in Advnace,
>
> Nick.
>
>
>
> On Sat, Jan 5, 2013 at 11:49 AM, Flavio Goncalves
> wrote:
> > Hi Nick,
> >
> > This setup seems to be wrong. If you are not going to register on
> Asterisk,
> >
; OpenSIPS Core Developerhttp://www.opensips-solutions.com
>
> On 01/03/2013 12:31 PM, Flavio Goncalves wrote:
>
> Hi Razvan,
>
> I'm checking if the problem is not related to re-invites and calling
> rtpproxy set with an invalid value. Anyway, the
; Can you tell me what versions of OpenSIPS and RTPProxy are you using?
>
> Best regards,
>
> Razvan Crainea
> OpenSIPS Core Developerhttp://www.opensips-solutions.com
>
> On 01/03/2013 12:31 PM, Flavio Goncalves wrote:
>
> Hi Razvan,
>
> I'm checking if the p
Hi Mariana,
There is the t_flush_flags to push changes in the flags after t_newtran().
For CANCEL, if you call t_check_trans, you can relay the CANCEL
automagically based on the transaction. It is like a shortcut. It is in the
default script.
if (is_method("CANCEL")) {
if (t_check_trans
Hi Samuel,
I suggest you investigate the new feature called distributed dialog
profiles http://lists.opensips.org/pipermail/users/2012-February/020657.html
Flavio E. Goncalves
www.sippulse.com
2013/1/7 samuel
> Hi folks,
>
> I'm started reading about dialog module and how to use it in a dis
I'm using what is below and it works.
server_header="Server: XX"
user_agent_header="User-Agent: XX"
In your configuration Server: is missing.
Flavio E. Goncalves
2013/5/29 M.Khaled W Chehab
> HI,
>
> ** **
>
> I set user_agent_header="User-Agent: Opensips-Switch"
>
> server_header
Hi Miha,
I would suggest two ways. The first is to use the RFC3263 and implement DNS
redundancy using DNS SRV and NAPTR. The only other way that I know is using
BGP and migrating the IP address to another server. You will need to have
an AS to do this. It can be implemented by a good (if he is wil
not be a problem with zones
> as need time to refresh (i just did I quick look to RFC
> that you posted in previous post.).
>
> tnx!
>
> Miha
>
> On Wed, 29 May 2013 11:19:59 -0300
> Flavio Goncalves wrote:
> > Hi Miha,
> >
> > I would suggest two wa
Hi Roman,
I had the same problem and in all the cases I've used a new IP address for
Asterisk. The problem is related to the routing of sequential messages, if
the address is the same (not considering the port) it considers the
destination as itself and loops the sequential requests. I'm not sure
Hi Nandini,
The addresses specified at the module rtpproxy.so are not matching the
addresses specified in the -s parameter of your rtpproxy daemons. Do you
have two instances of the rtpproxy running in the addresses 127.0.0.1:5000and
192.168.2.40:5.
Flavio E. Goncalves
Dear All,
>
>
> i have
Hi Samuel,
Try using add_path_received().
Flavio E. Goncalves
2013/5/28 Samuel
> Hi Iñaki,
>
> I'm struggling with this NAT traversal module.
>
> I have the following setup:
>
> UA-NAT box P1 - P2/REGISTRAR---WORLD!
>
> - I just own P1 configuration (P2 is a voip service provide
he rtpproxy is loaded, and
> on which ip address is it running
>
> Thanks in advance
> Nandini
>
>
> On Mon, Jun 3, 2013 at 4:05 PM, Flavio Goncalves wrote:
>
>> Hi Nandini,
>>
>> The addresses specified at the module rtpproxy.so are not matching the
>
bin/opensips[4372]:
> ERROR:mediaproxy:mediaproxy_connect: failed to connect to
> /var/run/mediaproxy/dispatcher.sock: No such file or directory
>
>
> please help me.
>
>
>
> On Mon, Jun 3, 2013 at 4:27 PM, Flavio Goncalves wrote:
>
>> You can use ps -ef |grep rtpp
Hi Khaled,
You can discover the memory actually used by the system using
opensipsctl fifo get_statistics shmem
opensipsctl fifo get_statistics pkmem
I guess you will be a little disappointed with so much ram wasted, OpenSIPS
do not consume much ;-).
Flavio.
2013/6/16 M.Khaled W Chehab
> I
**
>
> FIFO command was:
>
> :get_statistics:opensips_receiver_23466
>
> Pkmem
>
> ** **
>
> ** **
>
> ** **
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Flavio Goncalves
> *Sent:* Monday, June
= 655016
>
> pkmem:58-max_used_size = 655152
>
> pkmem:58-free_size = 32899416
>
> pkmem:58-fragments = 2
>
> pkmem:59-total_size = 0
>
> pkmem:59-used_size = 0
>
> pkmem:59-real_used_size = 655016
>
> pkmem:59-max_used_size = 655152
Hi Khaleb,
Set uac_replace_to and uac_replace_from in a branch_route.
Flavio E. Goncalves
2013/6/19 M.Khaled W Chehab
> Hi,
>
> ** **
>
> I am running opensips 1.8.3 with do_routing module
>
> A dial_rule prefix has 3 trunk gateways ( gw1,gw2,gw3)
>
> ** **
>
> After do_routing
Hi,
Please, check your DB parameters in db.inc.php.
Flavio E. Goncalves
2013/6/19 Nandini madhu
> Dear All,
> Greetings,
>
> i have got the mi_xmlrpc.so file. but in control panel the error is:-
>
> *Failed to issue query, error message : MDB2 Error: no such table*
>
> thanks in advance
>
>
Hi Khaled,
I can try to show you the way, but to walk is up to you ;-).
Flavio E. Goncalves
2013/6/19 M.Khaled W Chehab
> Yes,
>
> ** **
>
> I am in need to change it every time I send the call to different trunk ,*
> ***
>
> Is there a way to restore the original header before changing i
Hi Jens,
Stun does not work over symmetrical nat. Please use an utility called
winstun and test the firewall/nat traversal of your clients. If they are
symmetric, the only way is rtpproxy. Rtpproxy does not work behind a
firewall without a patch. The scenario you are trying to create is not an
ea
Hi Ouvidiu,
Thanks for pointing a better way to use rtpproxy behind NAT without a
patch. I was actually patching rtpproxy with the mentioned patch. Your
solution is much better. I will try next time :-).
Best regards,
Flavio E. Goncalves
2013/6/23 Ovidiu Sas
>
> > Rtpproxy does not work behi
Hi Nandini,
There is a mysql stored procedure call to calculate the duration of the
calls. There are some files at the tool subdirectory.
/var/www/opensips=cp/config/tools/system/cdr (if I'm remembering
correctly). There is a sql file with a procedure to import to mysql (or
postgresql) and a shell
|
> ||
> | sip_from_tag| varchar(128) | NO | |
> ||
> | sip_to_tag | varchar(128) | NO | |
> ||
> | created | datetime | NO | | -00-00 00:00:00
> ||
>
> +-+--+--+-+
Hi Alexander,
I had exactly the same problem. I'm sending to your email the pcap file.
Actually I don't think the problem is the request. I have inspected the
pcap file and seems fine according to HEP v2. I've changed the source code
to force UDP and it worked fine after that. It is is not a patc
param("siptrace","db_url","mysql://OPENSIPS_USER:OPENSIPS_PASS@DB_IP
/DB_NAME"
Flavio E. Goncalves
2013/7/10 Alexandr Dubovikov
> Hi Flavio,
>
> how was the original SIP messages generated ?
>
> Wbr,
> Alexandr
>
>
> 7/10/2013 5:05 PM,
exandr
>
>
> 7/10/2013 10:41 PM, Flavio Goncalves wrote:
>
> Hi Alexander,
>
> They were generated by the siptrace module in another OpenSIPS Server 1.8
>
> # Sip Trace Params
> modparam("siptrace","duplicate_uri","sip:ix.y.z.w:9060")
Hi Chen-Che
If you are using CDR_FLAG, you must create the dialog. BYEs are accounted
in the loose_route (sequential request) section, toplogy_hiding also
depends on the dialog.
Failed calls are astore in the ACC table, but if your call was forwarded to
a voicemail or to other gateway, the ACC re
Hi Rik,
Try to use pedantic=no (sip.conf) on Asterisk. it stops some SIP checkings
for Asterisk. Usually this is the default setting, but it is worth
checking.
Best regards,
Flavio E. Goncalves
2013/11/6 Rik Broers
> Hmm I can see that increasing Cseq on proxy would create some out of
> s
Hi,
I'm trying to use the b2b_logic module. I want to grab the b2b_key just
after calling b2b_init_request, I'm using OpenSIPS 1.8. In some older
versions we had the b2bl_key_avp. Now this parameter has disappeared (
https://sourceforge.net/p/opensips/bugs/502/)
When trying to use i get "Paramete
opensipsctl fifo cache_remove
Flavio E. Goncalves
2014/1/21 Martin Stock
> Hi guys,
>
> is there a chance to clear local cache in opensips without a restart. E.g.
> via opensipsctl command?
>
> For example I use cache_store() in opensips.cfg:
> --- snip ---
> cache_store( "local", "username_$a
Hello,
Your script seems to be ok. I would insert an exit after xlog("L_INFO", "No
conditional forward found"); Check the Asterisk server, did you issued an
answer() before starting the IVR? Use ngrep to troubleshoot the SIP flow and
check if the messages are being relayed to the right place.
Re
Hi Jain
In the first place check if the acc is being populated. If it is not check
the acc module parameter db_url. If the events INVITE and BYE are in the
acc, check the stored procedure used to calculate the cdrs,
opensips_cdrs_sql (If i remember correctly). you need to have a call to
this sto
Hi Luis,
Which version of SIPp are you using? I had several issues with version 2.x.
Please try the latest version from svn. Check the error runing sipp using
-trace_err(or something similar, I don't remember the exact parameter now).
Identify which unexpected messages are ocurring (e.g. 408, 404
Hi Jerry,
Check if the daemon is up using:
ps -ef |grep opensips.
If were not up, check the syslog file
tail /var/log/messages
Sometimes one start opensips, the screen shows the ip/names, but the daemon
is not starting for some reason.
Flavio E. Goncalves
CEO - SipPulse
On Tue, Feb 14, 20
Hi,
Usually, too many hops means that the domain is not in the domain table.
The server identifies the destination as an external server and relay using
dns to the same server.
Flavio E. Goncalves
SipPulse Sip Solutions
On Fri, Feb 17, 2012 at 11:28 AM, Faisal Rehman
wrote:
> Hi Everyone,
>
Hello all,
I'm noticing some issues with validate dialogs in one of our servers.
ERROR:dialog:dlg_validate_dialog: failed to validate remote contact: dlg=[
sip:140012556185441445@216.59.16.137] , req=[
sip:140012556185441445@216.59.16.137:5060].
What I see different on dlg and req is the port nu
Hi Jan,
I had the same issue recently, I have captured some packets at the same
exact time of the messages. In some cases, if your system becomes
unresponsive for a few seconds, CANCELs were not relayed by the proxy
(actually I saw several requests without any reply for some time). Then,
the other
Hi,
I'm receiving some push_reply errors. I've checked the logs and traces and
I can't spot the problem. Any help would be welcome, below the debug of the
exact moment of the error. .
Apr 4 18:43:32 sip /sbin/opensips[13240]: DBG:tm:t_check:
start=0x
Apr 4 18:43:32 sip /sbin/ope
et
> Toll-Free: 888.929.VOIP ( 8647 )
>
>
>
> On Wed, Apr 4, 2012 at 6:27 PM, Flavio Goncalves
> wrote:
> >
> > Hi,
> >
> > I'm receiving some push_reply errors. I've checked the logs and traces
> and I
> > can't spot the problem. Any help
ixed after the release of
> 1.7.2, make sure you are building from SVN
> https://opensips.svn.sourceforge.net/svnroot/opensips/branches/1.7 to
> ensure you got the fix(es) .
>
> Regards,
> --Rudy
> Dynamic Packet
> Toll-Free: 888.929.VOIP ( 8647 )
>
>
>
> On Wed,
gards,
>
> Vlad Paiu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
>
> On 04/05/2012 02:29 PM, Flavio Goncalves wrote:
>
> Hi,
>
> I'm using the latest SVN and validate_dialog, but the problem persists.
> When I check the CANCEL coming from the originat
I don't know if anyone has faced this issue.
Rtpproxy timeout notification does not work after opensips restart. When
you start opensips and rtpproxy just after, timeout notifications work. If
you restart opensips, the session is timed out, but the notification is not
sent.
## Session with notif
see that debug message. Have you tried to
> take a trace on port 7891 and see if RTPProxy is sending any message?
>
> Best regards,
>
> Razvan Crainea
> OpenSIPS Core Developerhttp://www.opensips-solutions.com
>
> On 09/14/2012 08:05 PM, Flavio Goncalves wrote:
>
>
Hi
You can use failure_route to generate events for gateway timeouts (Internal
408), save in cache the number of timeouts and after a certain thershold
raise an event. Seems simpler and more effective than checking all the time
using fifo.
Flavio E. Goncalves
CEO - SipPulse, www.sippulse.com
Remco,
Use hangup with a code.
Example: hangup(17) generates a 486 message.
A complete list can be found at
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
Flavio E. Goncalves
2012/10/14 Remco .
> Thanks Max. That does the trick for the Asterisk part. However, calls are
>
Remco
Use hangup(3) and Asterisk will send a 404
Excerpt from
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause.
ISUP Cause value SIP response
1 unallocated number 404 Not Found
2 no route to network 404 Not found
3 no route to destination 404 Not found
16 normal call clearing --
Hi Daniel,
I don't know exactly why, but it doesn't work as far as I know.
loose_route() does not work in the same server with different ports. I have
had the same problem and the solution I used was to add an alias with
another ip address in the same machine for the media server.
Flavio E. Gonca
Hi,
One of the best ways to solve it is to include the DID in an extra header.
We usually include something such as X-DID: did_number. Then in the
Asterisk/Freeswitch PBX server you can recover this DID reading this header
and routing internally. Many ITSPs implement in this way. Thus, you can
hav
Is it possible to remove completely the server header on OpenSIPS?
Regards,
Flavio E. Goncalves
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Hi,
Yes you can extract audio from rtpproxy. The extractaudio utility is very
handy and you can compile with G.729 from the linphone project bcg729. It
is very easy to use, simply use the utility followed by the name of the
recording without any extension. Check the source code for the other
opti
Hi,
I'm getting some errors like below (2 simultaneous calls). No stress test.
pkmem is ok, shmem is ok.
CRITICAL:core:timer_ticker: timer handler lasted (503 us)
for more than timer tick (100 us) -> potential timer shifting.
Anyone with the same problem?
Flavio E. Goncalves
__
Hi,
I'm getting some errors on drouting and I got some routes sent to wrong
gateways (some gateways being skipped). It seems a fail to insert in the
blacklist, but never saw this message before, opensips 1.11. I don't know
if the problems are correlated.
Att,
Flavio E. Goncalves
V.Office Redes e
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