Re: [OpenSIPS-Users] How to implement a SIP Trunk in between two SIP servers
Hi Bogdan, Leon
Thank you very much for your information
I have some questions about failover configuration.
I have two IP addresses of the Cisco redundancy SIP Trunking servers. The
OpenSIPS can ping and detect which server is available before perform the
routing and making call to do the fail-over situation, can you provide me
some script examples for this situation. Thanks
My Sip Trunk configuration is :
The sip trunk configuration would be like below.
# all numbers starting with 55 are to be sent to CUCM
if ($rU =~ ^55[0-9]+$) {
# replace the domain part of RURI to point to CUCM
rewritehostport(CUCM_IP:CUCM_PORT);
# route the call out based on RURI
route(1);
}
Thanks
Kind Regards,
Steven
On 13 April 2015 at 14:40, steven chew steven.chew.jacq...@gmail.com
wrote:
Hi Bogdan, Leon
Thank you very much for your information
I have some questions about failover configuration.
I have two IP addresses of the Cisco redundancy SIP Trunking servers. The
OpenSIPS can ping and detect which server is available before perform the
routing and making call to do the fail-over situation, can you provide me
some script examples for this situation. Thanks
My Sip Trunk configuration is :
The sip trunk configuration would be like below.
# all numbers starting with 55 are to be sent to CUCM
if ($rU =~ ^55[0-9]+$) {
# replace the domain part of RURI to point to CUCM
rewritehostport(CUCM_IP:CUCM_PORT);
# route the call out based on RURI
route(1);
}
Thanks
Kind Regards,
Steven
On 12 January 2011 at 13:18, Leon Li leon...@aarnet.edu.au wrote:
Hi Steven,
To configure the trunk in CUCM, go to Device Trunk, add a new “SIP
trunk”.
The configuration fields are pretty straight forward. Important ones are
· Destination Address, i.e. opensips IP
· Port, if not 5060
· CSS for inbound and outbound calls. (this decide what number
you can send calls to and receive calls from opensips)
· Any number transformation if you have
This is the basic. If you have questions about particular fields, please
mail in details.
Regards,
Leon
*From:* users-boun...@lists.opensips.org [mailto:
users-boun...@lists.opensips.org] *On Behalf Of *steven chew
*Sent:* Tuesday, 11 January 2011 11:50 AM
*To:* OpenSIPS users mailling list
*Subject:* Re: [OpenSIPS-Users] How to implement a SIP Trunk in between
twoSIP servers.
Hi Bogdan,
Thanks for your reply.
Your script is very useful for calling between two opensips servers which
I have tested.
However, I don't know how to configure on CUCM 7.0 which I am using.
At the moment, CUCM 7.0 is using Web Config via the Web Browser.
Can you let me know how to configure on CUCM 7.0?
I will appreciate very much if you give some instructions
for configuring SIP Trunk on CUCM7.0
Thanks
Kind regards,
Steven,
On 10 January 2011 19:33, Bogdan-Andrei Iancu bog...@voice-system.ro
wrote:
Hi Steven,
To do that, you need to add in opensips some routing to 1) recognize the
numbers that needs to be sent to CUCM and 2)route that calls to CUCM.
For script logic it sounds like : if you receive a new call (initial
INVITE) for your local domain, check the URI and divert. If you look at the
default config file, there is comment # requests for my domain - from
that point further you have only initial INVITEs for your local domain, so
you can add after:
# all numbers starting with 55 are to be sent to CUCM
if ($rU =~ ^55[0-9]+$) {
# replace the domain part of RURI to point to CUCM
rewritehostport(CUCM_IP:CUCM_PORT);
# route the call out based on RURI
route(1);
}
For the other way around, you have to put a similar logic in CUCM, like
to divert all calls starting with 12 to opensips - and replace the domain
on RURI with the IP/domain of opensips.
Regards,
Bogdan
steven chew wrote:
Hi Bogdan,
Thank you very much for your reply.
I have an Opensips Server and a Cisco Unified Communication Manager
(CUCM).
If I want to make calls from Opensips Server to CUCM and CUCM to Opensips
Server.
For example:
1) If I dial an extension number 5566 from a SIP Phone 12345 under
Opensips Server, it will try to call to a Cisco IP Phone 5566 from CUCM
through a SIP Trunk.
2) If I dial an extension number 12345 from a Cisco IP Phone 5566
under CUCM, it will try to call to a SIP Phone 12345 under Opensips
Server through a SIP Trunk.
Can you give some instructions how to configure the above scenario for
dialing extension numbers?
Thanks
Steven,
On 6 January 2011 21:31, Bogdan-Andrei Iancu bog...@voice-system.ro
mailto:bog...@voice-system.ro wrote:
Hi Steven,
If you use the opensips default script, your opensips will accept
calls from any other external SIP entities (call targeting a local
opensips subscriber).
If you want to
Re: [OpenSIPS-Users] How to implement a SIP Trunk in between two SIP servers for failover situation
Hi Bogdan, Leon
Thank you very much for your information
I have some questions about failover configuration.
I have two IP addresses of the Cisco redundancy SIP Trunking servers. The
OpenSIPS can ping and detect which server is available before perform the
routing and making call to do the fail-over situation, can you provide me
some script examples for this situation. Thanks
My Sip Trunk configuration is :
The sip trunk configuration would be like below.
# all numbers starting with 55 are to be sent to CUCM
if ($rU =~ ^55[0-9]+$) {
# replace the domain part of RURI to point to CUCM
rewritehostport(CUCM_IP:CUCM_PORT);
# route the call out based on RURI
route(1);
}
Thanks
Kind Regards,
Steven
On 12 January 2011 at 13:18, Leon Li leon...@aarnet.edu.au wrote:
Hi Steven,
To configure the trunk in CUCM, go to Device Trunk, add a new “SIP
trunk”.
The configuration fields are pretty straight forward. Important ones are
· Destination Address, i.e. opensips IP
· Port, if not 5060
· CSS for inbound and outbound calls. (this decide what number
you can send calls to and receive calls from opensips)
· Any number transformation if you have
This is the basic. If you have questions about particular fields, please
mail in details.
Regards,
Leon
*From:* users-boun...@lists.opensips.org [mailto:
users-boun...@lists.opensips.org] *On Behalf Of *steven chew
*Sent:* Tuesday, 11 January 2011 11:50 AM
*To:* OpenSIPS users mailling list
*Subject:* Re: [OpenSIPS-Users] How to implement a SIP Trunk in between
twoSIP servers.
Hi Bogdan,
Thanks for your reply.
Your script is very useful for calling between two opensips servers which
I have tested.
However, I don't know how to configure on CUCM 7.0 which I am using.
At the moment, CUCM 7.0 is using Web Config via the Web Browser.
Can you let me know how to configure on CUCM 7.0?
I will appreciate very much if you give some instructions for configuring
SIP Trunk on CUCM7.0
Thanks
Kind regards,
Steven,
On 10 January 2011 19:33, Bogdan-Andrei Iancu bog...@voice-system.ro
wrote:
Hi Steven,
To do that, you need to add in opensips some routing to 1) recognize the
numbers that needs to be sent to CUCM and 2)route that calls to CUCM.
For script logic it sounds like : if you receive a new call (initial
INVITE) for your local domain, check the URI and divert. If you look at the
default config file, there is comment # requests for my domain - from
that point further you have only initial INVITEs for your local domain, so
you can add after:
# all numbers starting with 55 are to be sent to CUCM
if ($rU =~ ^55[0-9]+$) {
# replace the domain part of RURI to point to CUCM
rewritehostport(CUCM_IP:CUCM_PORT);
# route the call out based on RURI
route(1);
}
For the other way around, you have to put a similar logic in CUCM, like to
divert all calls starting with 12 to opensips - and replace the domain on
RURI with the IP/domain of opensips.
Regards,
Bogdan
steven chew wrote:
Hi Bogdan,
Thank you very much for your reply.
I have an Opensips Server and a Cisco Unified Communication Manager (CUCM).
If I want to make calls from Opensips Server to CUCM and CUCM to Opensips
Server.
For example:
1) If I dial an extension number 5566 from a SIP Phone 12345 under
Opensips Server, it will try to call to a Cisco IP Phone 5566 from CUCM
through a SIP Trunk.
2) If I dial an extension number 12345 from a Cisco IP Phone 5566
under CUCM, it will try to call to a SIP Phone 12345 under Opensips
Server through a SIP Trunk.
Can you give some instructions how to configure the above scenario for
dialing extension numbers?
Thanks
Steven,
On 6 January 2011 21:31, Bogdan-Andrei Iancu bog...@voice-system.ro
mailto:bog...@voice-system.ro wrote:
Hi Steven,
If you use the opensips default script, your opensips will accept
calls from any other external SIP entities (call targeting a local
opensips subscriber).
If you want to configure your opensips to accept foreign calls
only form a specific IP address, you can use the permission
module, with address table to implement IP-based authentication.
Best regards,
Bogdan
steven chew wrote:
Hi everyone,
I am a newbie with SIP-Trunk in OpenSips.
I have a Cisco Communication Unified Manager and a OpenSips
Server running in two different Virtual Machines.
I would like to have a SIP trunk in between them Cisco
Communication Unified Manager and OpenSips Server.
Therefore, I can make a call from OpenSips Server's SIP
Clients to Cisco IP Phone.
What should I need to add into opensips.cfg configuration file?
I hope you can give some simple examples how to do it.
I look forward to hearing from your advise asap.
Re: [OpenSIPS-Users] Pointers for configuring a simple SBC with Opensips
Hello Paul, Checkout Blox SBC www.blox.org it is built based on Opensips as a core session router. On Fri, Apr 10, 2015 at 7:41 PM, POTOCHNIAK, PAUL A pp8...@att.com wrote: Hello, Can anyone give me some advice on using Opensips as a simple SBC? I have downloaded and built a 1.11.4 Opensips instance. I have played around with the Trunking script and the B2B tutorial. At a minimum I would like to configure a B2B system utilizing two NICs. I’m somewhat at loss after playing around with the B2B tutorial on how to get the B2B receive on one interface and writing out to another. Is this the correct approach? Or is there a better way to approach this? Any advice would be appreciated. Thanks, Paul ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] g.729 license on wlan0
According the https://wiki.freeswitch.org/wiki/Mod_com_g729, it says ... Currently, we only support interfaces named with the ethX scheme. Please change the interface name accordingly. Can I run freeswitch g.729 on wlan0? I would like to test g.729 using my wireless. Is there a trial license for this? Thanks, Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensipsctl fifo dlg_list to_uri:: shutdown Opensips
Bump!! Please help -- Sent from my iPhone On Apr 10, 2015, at 7:56 AM, Satish Patel satish@gmail.com wrote: Any thought? Why that command killing my opensips? I think it's a BUG. On production it will be dangerous if it kill service with random command. -- Sent from my iPhone On Apr 9, 2015, at 11:25 PM, Satish Patel satish@gmail.com wrote: Is this a bug? or its normal behavior? It is very dangerous, I was just playing with command and i type following command which kill my opensips process, when i restarted service then it back. I am using opensips 2.1 [root@sip ~]# opensipsctl fifo dlg_list to_uri:: [root@sip ~]# opensipsctl fifo dlg_list ERROR: /tmp/opensips_fifo does not exist ERROR: Make sure you have the line 'modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)' in your config ERROR: and also have loaded the mi_fifo module. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
