Check
https://blog.opensips.org/2016/12/13/how-to-proxy-sip-registrations/
https://blog.opensips.org/2016/12/20/mid-registrar-scalable-registration-and-call-forking/
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
https://www.siphub.com
On
Still looking for possibly a template/example code on this.
I am setting a bounty of $150 for anyone willing to help.
You can reach out to me via E-Mail or phone at 407-999-
Thanks!
On Mon, Mar 13, 2023 at 8:26 PM Dylan Cruz wrote:
> I'd love a sample OpenSIPS Config that would let me
I'd love a sample OpenSIPS Config that would let me accomplish using it as
a transparent proxy to Asterisk running on the same system. I found a few
tutorials but found a lot of conflicting information and outdated sources,
Once I have that I will have enough to work on to do what I want...
Here is a list of changes I found:
1) Must build asterisk with ODBC storage enabled for voicemail because
using file storage will not store messages in the database.
2) Uncomment the lines *'odbcstorage=asterisk'* and
*'odbctable=voicemessages'* in voicemail.conf to enable database storage
for
Nice !
As a way of helping us (project) back, could synthesize a list with
things that did changed since the tutorial was written ? And I will
re-generate the tutorial, so other people will benefit from it.
Thanks and regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
I also found the correct way to deal with the LIMIT problem. Asterisk has a
built-in way to deal with this. In file* /etc/asterisk/res_odbc.conf*, the
following should be added under [asterisk] :
limit => 5
share_connections => no
Now everything is working well without problems.
Nabeel
Perfect ! is there any left to be solved, or everything works fine ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 14.07.2016 13:33, Nabeel wrote:
Hi Bogdan,
I have been able to solve that problem.
The issue was that I had asterisk compiled
Hi Bogdan,
I have been able to solve that problem.
The issue was that I had asterisk compiled with file storage enabled
instead of ODBC storage. I recompiled asterisk with ODBC storage enabled
and now database storage is working.
Thanks.
Nabeel
On 14 Jul 2016 11:15 a.m., "Bogdan-Andrei Iancu"
Hi Nabeel,
1) that limit should not be necessary, as you should have in DB a single
record for each subscriber. If multiple records are returned, it means
your data is not correct.
2) in those lines, the "asterisk" and "asteriskcfg" are the names of the
odbc connection - I pasted an example
Hi Nabeel,
That means the vmusers and vmaliases do work ok, still the VM storage
engine does not. Do you have in voicemail.conf the following:
odbcstorage=asteriskrt
odbctable=voicemessages
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
I have been able to solve the issue of loading numbers without using the
voicemail.conf file.
After adding the line *'voicemail => odbc,asterisk,vmaliases'* to
extconfig.cfg, I removed the suffix " |u " from extensions.conf:
exten => _VMR_.,n,Voicemail(${EXTEN:4}*|u*)
Now all phone numbers in
0. The line block was in the default OpenSIPS config, but I agree that it
is not in the tutorial so should be removed (for voicemail).
1. I think there is a misunderstanding here. 'limit' is not a column; I am
referring to the mysql LIMIT clause:
Thank you Nabeel,
The number you added in voicemail file - does it exist in the
sipuser/subscriber table ??
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 04.07.2016 11:38, Nabeel wrote:
Hi Bogdan,
I just added the column to the view by
Hi,
0. The cfg block you mentioned as removed does not exists in the cfg as
per tutorial.
1. the "limit" column does not exist in the sipusers as per tutorial, so
it might have been added in newer asterisk versions; not sure what is
its meaning, but if setting it to 1 makes asterisk happy,
Hi Bogdan,
I just added the column to the view by adding "NULL AS `callbackextension`"
to the SQL view definition. I haven't linked the column to the subscriber
column, so this may not be the correct definition. However, it got rid of
the error.
About the voicemail.conf file, when I attempted to
Hi,
the voicemail.conf file exists in almost all asterisk versions. But if
you use the odbc storage for voicemail, you do not need this file at all.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02.07.2016 15:41, Nabeel wrote:
In the
Hi,
What is the definition you used for this new column ?
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02.07.2016 05:29, Nabeel wrote:
In the last error message,/'//callbackextension = ?' /suggested that
this column is missing from the
Hi,
This kind of ordering is valid in older versions of Asterisk. Maybe not
anymore in the newer versions.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02.07.2016 04:23, Nabeel wrote:
The tutorial contains a mistake where the priority
Hi Samy,
Point 1 I cant imagine how those lines possibly relate to no media error in
> asterisk, I guess it depends on your config setup.
In point 1 I was referring to this error:
WARNING[17112] res_odbc.c: SetConnectAttr (Txn isolation) returned an
> error: HY000: [MySQL][ODBC 5.2(w)
Hi Nabeel,
Point 1 I cant imagine how those lines possibly relate to no media error in
asterisk, I guess it depends on your config setup.
The logical answer to your point 2 would be Asterisk realtime. However this
is not going to be as staraight forward as making asterisk use subscriber
table
The last error message has been solved by removing the following lines from
opensips.cfg:
if (!db_does_uri_exist()) {
>send_reply("420","Bad Extension");
>exit;
>}
>
>t_newtran();
>
In the latest version of Asterisk, there is a new file voicemail.conf which
must be configured correctly for voicemail, but the tutorial does not
mention this file at all. Please let me know how to configure this file for
integration with OpenSIPS.
Nabeel
In the last error message,* '**callbackextension = ?' *suggested that this
column is missing from the sipusers mysql view. So I added this column to
the view and now that error has been resolved. Only the following error
remains now:
[Jul 2 03:25:48] WARNING[19330][C-0005]: app.c:1633
>
The issue in my last Email has solved the error about missing extension.
Now the following errors remain:
[Jul 2 02:29:18] WARNING[18226][C-0001]: res_config_odbc.c:117
> custom_prepare: SQL Prepare failed![SELECT * FROM sipusers WHERE host = ?
> AND callbackextension = ? AND port = ?]
>
The tutorial contains a mistake where the priority ordering in
extensions.conf should start with 1, not n:
; Voicemail
> exten => _VMR_.,1,Ringing
> exten => _VMR_.,n,Wait(1)
> exten => _VMR_.,n,Answer
> exten => _VMR_.,n,Wait(1)
> exten => _VMR_.,n,Voicemail(${EXTEN:4}|u)
> exten =>
Hi,
Adding 'limit 1' or 'limit 5' to the supusers mysql view resolves part of
the error, but I don't understand why that is and whether this is correct
for the setup. Maybe something to do with connection pooling?
Now the following errors remain:
[Jun 30 01:07:53] NOTICE[17067][C-]
Hi Nabeel,
The "sipusers" mysql view (as per
http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8#toc7
) has both the name and host fields - not sure why that query may fail.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
Hi Bogdan,
I was able to install the latest versions of Asterisk (13.1) and Opensips
(2.3) according to the tutorial, but when attempting to leave a voicemail I
get the following errors:
> [Jun 30 01:07:53] NOTICE[17067][C-] chan_sip.c: Call from
> '+447867958678' (162.249.6.206:12221)
Hi Nabeel,
We will update the tutorial for 2.2, but it should still match. Give it
a try and if you hit issues, just let me know.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12.06.2016 10:18, Nabeel wrote:
Hi,
I will be following this
Hi,
I will be following this tutorial to integrate OpenSIPS and Asterisk:
http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8
The tutorial mentions the use of OpenSIPS version 1.8 and Asterisk version
1.8. I would like to know if I can use the latest versions of
One disadvantage is performance for example if you have too many
registrations on Asterisk the system will just crash.
On Jul 26, 2015 1:34 PM, Nabeel nabeelshik...@gmail.com wrote:
Perhaps a better way to word my question: what are advantages of using
OpenSIPS over Asterisk for a basic SIP
To: OpenSIPS users mailling listusers@lists.opensips.org
Reply-To: OpenSIPS users mailling list users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] OpenSIPS vs Asterisk as SIP server
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi
Perhaps a better way to word my question: what are advantages of using
OpenSIPS over Asterisk for a basic SIP service?
On 25 Jul 2015 21:33, Nabeel nabeelshik...@gmail.com wrote:
I understand that Asterisk is a PBX but it also has core SIP
functionality. What are the disadvantages of using
I understand that Asterisk is a PBX but it also has core SIP
functionality. What are the disadvantages of using Asterisk over OpenSIPS
for a basic SIP service?
___
Users mailing list
Users@lists.opensips.org
13 apr 2013 kl. 21:43 skrev Nick Khamis sym...@gmail.com:
Make sure that you have host=dynamic on both the general level (i.e.,
sip.conf) and at the
peer level (i.e., extensions, sip_peers in the database etc...)
host=dynamic has no effect whatsoever in the general section of sip.conf.
You
13 apr 2013 kl. 22:08 skrev sjs205 sjs205.li...@gmail.com:
Hello N,
Thanks for getting back to me on this. This is one of the issues with this
tutorial, one can not set the asterisk sip_peers to dynamic since the
tutorial creates a view from the 'subscriber' table and uses the 'domain'
Hello Olle,
I had previously tried this, and it seemed to help somewhat, although I
then have to create the following fields too otherwise Asterisk
complains about not being able to update them:
alter table subscriber add column `fullcontact` int(35) DEFAULT NULL;
alter table subscriber add
On 4/14/13, Olle E. Johansson o...@edvina.net wrote:
13 apr 2013 kl. 21:43 skrev Nick Khamis sym...@gmail.com:
Make sure that you have host=dynamic on both the general level (i.e.,
sip.conf) and at the
peer level (i.e., extensions, sip_peers in the database etc...)
host=dynamic has no
But should the registration be carried out by asterisk or opensips in
the tutorial environment?
On 04/14/2013 08:31 PM, Nick Khamis wrote:
On 4/14/13, Olle E. Johansson o...@edvina.net wrote:
13 apr 2013 kl. 21:43 skrev Nick Khamis sym...@gmail.com:
Make sure that you have host=dynamic on
Apologies, I just re-read the start of the tutorial that confirms that
registration should be carried out by openSIPS rather than asterisk,
which is there to support media services...
I've clearly spent too long looking at this! ;)
On 04/14/2013 08:50 PM, sjs205 wrote:
But should the
Hello all,
I'm going round and round in circles trying to integrate openSIPS and
asterisk using the tutorial found at:
http://www.opensips.org/Resources/DocsTutAsterisk18
I have managed to get openSIPS installed and starting without errors
using the configuration scripts included in the
Make sure that you have host=dynamic on both the general level (i.e.,
sip.conf) and at the
peer level (i.e., extensions, sip_peers in the database etc...)
N.
On 4/13/13, sjs205 sjs205.li...@gmail.com wrote:
Hello all,
I'm going round and round in circles trying to integrate openSIPS and
If you are using Asterisk then you dont need media proxy as asterisk can
handle NAT and Media issues. You just need to forward SIP messages to
asterisk. Use OpenSIPs in LoadBalancer/Dispatcher scenerio.
Regards,
Qasim
On Thu, Mar 28, 2013 at 12:07 AM, Jagadish Thoutam
HI Qasim,
Thanks for the Replay
Can you post one sample config file so that i can go head with that
Thanks
Jagadish
On 29 March 2013 06:04, qasimak...@gmail.com qasimak...@gmail.com wrote:
If you are using Asterisk then you dont need media proxy as asterisk can
handle NAT and Media issues.
HI All,
I am New Here, i am getting Confusion while i am useing Openisps with My
asterisk Cluster My Implimentation Plan is Like this
(NAT)Opensips
1--- |
Asterisk1
Hi Bogdan,
connecting Opensips with Asterisk I can see that if a client registered on
Opensips server tries to make a call to a client in Asterisk domain, after
the INVITE, it receives a forbidden message from asterisk. I have set the
forwarding functionality in Opensips (rewriteuri function)
You need to add a route in your extensions.conf in the context where you
send all un-authenticated calls.
Maybe its your default context?
[default]
exten = 1001,1,Dial(SIP/1001,10,tTr);
On 5/4/10 9:02 AM, wüber wrote:
Hi Bogdan,
connecting Opensips with Asterisk I can see that if a
The problem seems to be not only in the extensions.conf file, but also in the
sip.conf file.
I still get this forbidden message!
--
View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/opensips-and-asterisk-tp4962200p5003971.html
Sent from the OpenSIPS - Users
Check in the SIP.conf where you send all unauthenticated calls.
On 5/4/10 11:45 AM, wüber wrote:
The problem seems to be not only in the extensions.conf file, but also in the
sip.conf file.
I still get this forbidden message!
___
Users
Carmelo,
If you have an SIP peer that matches the host and port of the opensips
server.. ie:
[opensips]
type=friend
host=ip of opensips.
port=port of opensips (can be omitted if port 5060)
Then it'll match that.. typically if it's coming from opensips you'll want
to add:
insecure=invite
so that
4 maj 2010 kl. 18.30 skrev Brett Nemeroff:
Carmelo,
If you have an SIP peer that matches the host and port of the opensips
server.. ie:
[opensips]
type=friend
host=ip of opensips.
port=port of opensips (can be omitted if port 5060)
Then it'll match that.. typically if it's coming from
Sorry, The way I recommend doing this was assuming the user on the
Asterisk box needed to be publicly reachable from anywhere.
I think that approach makes sense when using DID's and inbound routing
that does need authentication.
On 5/4/10 12:55 PM, Olle E. Johansson wrote:
4 maj 2010 kl.
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] opensips and asterisk
Sorry, The way I recommend doing this was assuming the user on the
Asterisk box needed to be publicly reachable from anywhere.
I think that approach makes sense when using DID's and inbound routing
that does need
is that is there a way to authenticate with
the real ip off the client
Thanks
-Original Message-
From: users-boun...@lists.opensips.org [mailto:
users-boun...@lists.opensips.org] On Behalf Of David J.
Sent: 04 May 2010 18:00
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users
Hi Carmelo,
routing between different SIP domains in typically done via DNS
(resolving the DNS part of the RURI).
OpenSIPS supports this by default - if you check the default config file
that comes with OpenSIPS, you will find the section when calls targeting
other domains are routed - also
Hi Robert,
The opensips dialog module mainly does dialog monitoring and has limited
capability when comes to checking dialog health (like it the call is not
zombie and it is really ongoing). The dialog module can just expire too
long calls (using a timeout for call duration).
First of all,
Hi,
sorry for cross-posting on both mailing lists, but I think a setup of Asterisk with OpenSIPS as frontend isn't unusual. So maybe both parties would be interested in this.
I'm using Asterisk (v1.4.21) to connect my OpenSIPS (v1.5.1) server to the PSTN (Asterisk connects to a local SIP
On Sat, Mar 13, 2010 at 11:28 AM, bhrugu mehta mehtabhr...@gmail.comwrote:
Hi, all
i have used opensips as registrar.
my scenario,
opensips-asterisk(routing logic)-opensips
i have done with opensips to asterisk call .
asterisk deside where to call go , and if local call then go to
Hi, all
i have used opensips as registrar.
my scenario,
opensips-asterisk(routing logic)-opensips
i have done with opensips to asterisk call .
asterisk deside where to call go , and if local call then go to opensips.
asterisk to opensips call not done.
any suggetion?
--
Bhrugu Mehta
Sr. S/W
general context of sip.conf is as following:
[general]
port = 5566
bindaddr = 0.0.0.0
dbname = mysql db name
dbhost = mysql db host ip
dbuser = mysql db username
dbpass = mysql db password
context = others
as here context is others so i add other context in extensions.ael as
following:
Hi,
On 30/12/09 10:35 AM, Saeed Akhtar wrote:
general context of sip.conf is as following:
[general]
port = 5566
bindaddr = 0.0.0.0
dbname = mysql db name
dbhost = mysql db host ip
dbuser = mysql db username
dbpass = mysql db password
context =
Look at this document for better idea
http://www.opensips.org/Resources/DocsTutAsterisk
ram
On Wed, Dec 30, 2009 at 3:05 PM, Saeed Akhtar saeedakhtar@gmail.comwrote:
general context of sip.conf is as following:
[general]
port = 5566
bindaddr = 0.0.0.0
dbname = mysql db name
hi all,
I'm configuring OpenSIPS with Asterisk. I used
seturi(sip:2...@asterisk_ip:ASTERISK_PORT); to forward my call to
Asterisk. Now Asterisk receives the call but shows a message that it can't
transfer call to my extension because it says call from ' ' to '2001' cannot
transfer because
create a user in sip.conf
Ram
On Tue, Dec 29, 2009 at 5:47 PM, Saeed Akhtar saeedakhtar@gmail.comwrote:
hi all,
I'm configuring OpenSIPS with Asterisk. I used
seturi(sip:2...@asterisk_ip:ASTERISK_PORT); to forward my call to
Asterisk. Now Asterisk receives the call but shows a message
Hi,
could you please draw a small flow of the call, just to understand it...
Like
UA(104) (INVITE)--- Asterisk - proxy1 -etc
For INVITE, 3xx , etc
Regards,
Bogdan
Jennifer-4 wrote:
Hi!
I´m using two Opensips as proxys, and they also take decisions about
redirections of
Hello Bogdan,
Thank you for the example. In that case the asterisk have to accept the
registration without starting a auth itself?
BR
Uwe
Bogdan-Andrei Iancu schrieb:
Hi Uwe,
If you look at the default opensips script, you have a section (by
default commented out) where the REGISTER
In that example, Asterisk has nothing to do with AUTH.
1) opensips get the REGISTER and sends back a challange
2) opensips get the REGISTER with credentials
3) opensips does the auth and forwards the REGISTER to *
Regards,
Bogdan
Uwe Kastens wrote:
Hello Bogdan,
Thank you for the example. In
Hello,
Has anybody a starting point for me to achieve the following:
UAC should register with asterisk put should be pre-authorized with
opensips. I saw an EMail from Bogdan, that this should be possible but
ATM I could only use opensips as a registrar or route all sip messages
through opensips.
Hi, There:
I am using opensips/kamailio in front of asterisk pool. my user register
on the opensips, and pstn call are routed out via asterisk. what I find out
is that when the caller calls callee, some of the UA doesn't generate ring
back. for example, if I use xlite, the ring back works fine.
Hello,
Yes, That was my mistake. now my asterisk fail over mechanism is working
fine.
Thanks a lot for helping me.
-urmi
On Tue, Aug 4, 2009 at 5:28 PM, Saúl Ibarra sag...@gmail.com wrote:
It's OPTIONS, not OPTION :)
--
Saúl -- Nunca subestimes el ancho de banda de un camión lleno de
Hello,
I am using Opensips and Asterisk for my call flow.
I am using 3 Asterisks for call forwarding. Opensips's dispatcher module is
doing the task of load balancing among all 3 Asterisk servers in a round
robin fashion. i. e.
1st call to 1st Asterisk
2nd call to 2nd Asterisk
3rd call to 3rd
urmi lakkad wrote:
modparam(dispatcher, ds_ping_method, INFO)
Asterisk does not respond to these. Try using the OPTIONS method instead.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678)
Hello Alex,
First of all Thanks for ur Attention and quick response.
If I use the OPTION then on Asterisk I m getting following:
[Aug 4 17:15:32] NOTICE[7765]: chan_sip.c:14958 handle_request: Unknown SIP
command 'OPTION' from '192.168.1.30'
Even when that Asterisk Comes up, opensips is not
It's OPTIONS, not OPTION :)
--
Saúl -- Nunca subestimes el ancho de banda de un camión lleno de disketes.
http://www.saghul.net/
___
Users mailing list
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Hi Ram,
By default, if failover is enabled, Dispatcher module will try all
destinations from the set, until it finds one working. You need to catch
the failures in failure_route and to use ds_next_domain|dst() functions
to try the next available destinations.
Also, by using the pringing
Hi Ruud,
Thanks for your help. I found the problem and now it is working.
regards,
Dimitris
On Thu, Jul 2, 2009 at 7:40 PM, Ruud Klaverr...@ag-projects.com wrote:
Hi Dimitrios,
On 02 Jul 2009, at 13:17, Dimitrios Giannakopoulos wrote:
Hi,
So, according to our scenario the asterisk has
Do you have nat=yes in your Asterisk configuration? In that case
Asterisk behaves just as Adrian said...
On Wed, Jul 1, 2009 at 9:33 PM, Adrian Georgescua...@ag-projects.com wrote:
Dimitrios,
Asterisk could be configured to wait for RTP from the end-point before
sending. If Mediaproxy is
Hi all,
Thanks for the help.
I have set nat=no but problem persists.
On Thu, Jul 2, 2009 at 9:24 AM, Saúl Ibarrasag...@gmail.com wrote:
Do you have nat=yes in your Asterisk configuration? In that case
Asterisk behaves just as Adrian said...
On Wed, Jul 1, 2009 at 9:33 PM, Adrian
Hi,
On 02 Jul 2009, at 08:58, Dimitrios Giannakopoulos wrote:
Hi all,
Thanks for the help.
I have set nat=no but problem persists.
So is your Asterisk on a public IP? Could you at least confirm with a
network trace that mediaproxy is forwarding RTP packets from the
gatways (from both
Hi Dimitrios,
On 02 Jul 2009, at 13:17, Dimitrios Giannakopoulos wrote:
Hi,
So, according to our scenario the asterisk has private ip. Any traffic
from/to the asterisk can be routed to/from our network.
The Network trace (between asterisk and opensips) shows that
mediaproxy does not forward
Hi,
Fist of all, I would like to express my apologies for sending
multiple mails to the list.
Firstly I would strongly suggest that you only send your question once,
sending it more than once will not get it answered sooner.
Secondly, I do not know what a Asterisk packet2packet bridge or a
On 01 Jul 2009, at 13:33, Dimitrios Giannakopoulos wrote:
Scenario B (failed RTP connection- Unknown IP of Asterisk)
The IP of Asterisk is displayed as Unknown simply because it never
sent any RTP packet to the port on the relay. both on the session
coming from the PSTN and on the session
Dimitrios,
Asterisk could be configured to wait for RTP from the end-point before
sending. If Mediaproxy is used in between this behavior should be
disabled otherwise they will wait one for another.
Adrian
On Jul 1, 2009, at 3:14 PM, Ruud Klaver wrote:
On 01 Jul 2009, at 13:33, Dimitrios
Hi all
After a long iam back to forum
back to my own topic and several readings done on this forum
how people doing same kind of setup what iam trying to achive
so here i have done some good developements
for testing iam doing all in one Server
Step1 :
Installed in Fresh BOX with Debian
Hi Ram,
I found your email on the Asterisk mailing list also ;)
So, to answer here also: do you get any reply back from Asterisk ?
Regards,
Bogdan
ram wrote:
Hi all
After a long iam back to forum
back to my own topic and several readings done on this forum
how people doing same kind of
On Tue, Jun 30, 2009 at 5:20 PM, Bogdan-Andrei Iancu bog...@voice-system.ro
wrote:
Hi Ram,
I found your email on the Asterisk mailing list also ;)
So, to answer here also: do you get any reply back from Asterisk ?
Hi Bogdan
thanks for the reply
I have made a quick Fix, iam not sure how
Hi,
On 29 Jun 2009, at 08:14, Dimitrios Giannakopoulos wrote:
Hi,
I have implemented the following scenario:
[incoming pstn]---[opensips]--[asterisk] ---[sip phone]
|
[outgoing pstn]---[opensips]--|
Opensips acts as SBC with
Hi,
I have implemented the following scenario:
[incoming pstn]---[opensips]--[asterisk] ---[sip phone]
|
[outgoing pstn]---[opensips]--|
Opensips acts as SBC with mediaproxy functionality. Moreover, I use
the LCR module to route calls.
Hello,
I have implemented the following scenario:
[incoming pstn]---[opensips]--[asterisk] ---[sip phone]
|
[outgoing pstn]---[opensips]--|
Opensips acts as SBC with mediaproxy functionality. Moreover, I use the LCR
module to route
El Miércoles, 29 de Abril de 2009, Iñaki Baz Castillo escribió:
El Miércoles, 29 de Abril de 2009, troxlinux escribió:
Hi Bogdan , I don't have any alias en mi opensips.cfg , the only thing
that I have is that when they make calls to the pstn they leave to
that ip port
route[4] {
Hi Bogdan , something stranger happens when I put the debug in 6 I
don't see that it shows me the opensips log
tail -f /var/log/openser.log
twoxserver /sbin/opensips[3744]: INFO:core:sig_usr: signal 15 received
twoxserver /sbin/opensips[3733]: INFO:core:sig_usr: signal 15 received
twoxserver
Hi,
Get a ngrep capture of the SIP traffic between * and OSIPS . Typically a
retransmission is triggered by a lack of response from the other party,
but to see what response is lacking, you need to see the ngrep capture
of the SIP traffic.
Regards,
Bogdan
troxlinux wrote:
Hi list , I have
excuseme , I didn't remember that there was a list
2009/4/27 Alex Balashov abalas...@evaristesys.com:
You may wish to consider posting this to the SER-Asterisk-Interwork list.
regardss
--
rickygm
http://gnuforever.homelinux.com
___
Users mailing
You may wish to consider posting this to the SER-Asterisk-Interwork list.
troxlinux wrote:
Hi list , I have some days fighting with asterisk and opensips to
solve this problem, when I use asterisk to listen my voicemail and to
call to the pstn, asterisk shows me this error message:
Hi,
There is a new functionality available for dispatcher module on 1.5 -
you can define (as module parameter) your custom set of SIP reply codes
to be considered as OK for probing:
http://www.opensips.org/html/docs/modules/devel/dispatcher.html#id271069
This is a new enhancement Anca did
Hello,
I am hoping someone can point me in the right direction. I have
configured my OpenSIPs server to load balance 10+ asterisk servers
using the dispatcher module. To date I have not been able to
implement the probe functionality because the OPTIONS and INFO
methods both cause asterisk to
Hi Geoff,
It's very strange that Asterisk answers OPTIONS pings with a 4xx error,
because OPTIONS is the method Asterisk uses to do its own availability
pings -- that's what the qualify= setting for peers in sip.conf enables.
What exactly is the 4xx error? Is it 403 Forbidden? Might it have
Iñaki Baz Castillo wrote:
El Domingo, 1 de Febrero de 2009, Alex Balashov escribió:
It's very strange that Asterisk answers OPTIONS pings with a 4xx error,
because OPTIONS is the method Asterisk uses to do its own availability
pings -- that's what the qualify= setting for peers in sip.conf
El Lunes, 2 de Febrero de 2009, Geoffrey Mina escribió:
Thanks for pointing me in the right direction. I didn't have an s
extension defined in my default context, so asterisk was returning a
404 error because OpenSIPs doesn't specify an extension in the OPTIONS
packet. The s is apparently
How would I configure the ruri in opensips to provide an extension
similar to sip:p...@asterisk.mydomain.com?
I couldn't get anything other than sip:asterisk.mydomain.com
Thanks.
Geoff
On 2/1/09, Iñaki Baz Castillo i...@aliax.net wrote:
El Lunes, 2 de Febrero de 2009, Geoffrey Mina escribió:
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