Re: [OpenSIPS-Users] openSIPS with an AS that would handle the routing logic
Hello, Since OpenSIPS is a SIP server, I would try to leverage it's capabilities and communicate over SIP with the AS - that means scripting your way through. On 20.04.2024 11:50, julien.royann...@orange.com wrote: Hello everyone, I'm reaching out to get your opinion on using openSIPS with a business Application Server (AS) that would handle the routing logic. At first glance, the SEAS module seems designed for this purpose, but it doesn't appear to be a good fit as it seems highly coupled with a Sip Servlet implementation using a specific protocol only supported for WeSIP. It might be better to use REST interfaces, a 302 redirect-based mechanism, or possibly another module. Thank you for your insights & advice! JR Orange Restricted Orange Restricted Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. This message and its attachments may contain confidential or privileged information that may be protected by law; they should not be distributed, used or copied without authorisation. If you have received this email in error, please notify the sender and delete this message and its attachments. As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified. Thank you. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to access to a column of a location database
Hello,
You can use
https://opensips.org/html/docs/modules/3.3.x/registrar.html#param_attr_avp
in order to populate any custom info at save() time and the info will
automatically be populated for you at lookup() time.
Regards,
Vlad
On 29.03.2024 15:08, guillaume.desgeo...@orange.com wrote:
Hi Bogdan,
Thank you for your answer.
And is it possible to use that lookup("location") function to put the
“registered socket” field in a variable in order to use it for my script ?
Regards,
Guillaume
*De :*Bogdan-Andrei Iancu
*Envoyé :* jeudi 28 mars 2024 11:15
*À :* OpenSIPS users mailling list ;
DESGEORGE Guillaume INNOV/IT-S
*Objet :* Re: [OpenSIPS-Users] How to access to a column of a location
database
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Hi Guillaume,
The registered contact (and the additional info) is fetched via the
lookup("location") function. It also sets the registered socket for
the routing the current request.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
https://www.siphub.com
On 22.03.2024 10:57, guillaume.desgeo...@orange.com wrote:
Hi everybody,
In my routing logic of my “opensips.cfg” file, I’m trying to
access the “socket” field of the location table of registered
contacts.
I have the registering informations correctly written in MySQL
location table but can’t find a function to access it.
I’d like to have a function which I give the registered username
and can give me back the associated socket.
Is the lookup () function the good one ? I didn’t understand how
to use it that way.
Thanks for your help,
Guillaume
Orange Restricted
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ce message par erreur, veuillez le signaler
a l'expediteur et le detruire ainsi que les pieces jointes. Les messages
electroniques etant susceptibles d'alteration,
Orange decline toute responsabilite si ce message a ete altere, deforme ou
falsifie. Merci.
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If you have received this email in error, please notify the sender and
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Ce message et ses pieces jointes peuvent contenir des informations
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pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce
message par erreur, veuillez le signaler
a l'expediteur et le detruire ainsi que les pieces jointes. Les messages
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Re: [OpenSIPS-Users] Stir Shaken: Failed to load certificate
Hi Ryan,
You have to provide to the stir_shaken_auth() function the actual
content of the certificate file and not just the path. The same goes for
the private key.
Regards,
--
Vlad Patrascu
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 21.11.2022 14:02, ryan embgrets wrote:
Greetings
I am trying to generate an Identity header by using the stir_shaken
module of the opensips.
But I am encountering the below error each time upon call.
Nov 21 11:15:20 local /usr/sbin/opensips[5051]:
ERROR:stir_shaken:w_stir_verify: Failed to load certificate
Nov 21 11:15:20 local /usr/sbin/opensips[5051]:
ERROR:stir_shaken:load_cert: Failed to parse certificate
#Module section.
loadmodule "stir_shaken.so"
route{
$var(cert) = "/etc/opensips/certs.pem";
$var(privKey) = "/etc/opensips/key.pem";
stir_shaken_auth("A", "",$var(cert), $var(privKey),
"https://domain.org/cert.pem";);
}
Though the cert looks valid, has proper permission for the opensips to
access them and cross checked with openssl for the verification.
Any pointer on what might be causing this?
Ryan
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Re: [OpenSIPS-Users] Load a DB Table to hash with MI control
Hello, Check out the sql_cacher [1] module. [1] https://opensips.org/html/docs/modules/2.4.x/sql_cacher.html Regards, Vlad On 17.10.2022 10:36, Antonis Psaras wrote: Dear all I am looking to implement a functionality similar to htable module were I will have a db table with some key values which will be loaded during service start and will expire every x sec or forced to be reloaded by MI. Using OpenSIPs 2.4 Any suggestion? Best regards *Antonis Psaras* ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Check the User-Agent for registered user
Hello, At registration time you could save the User Agent in the attributes per registration [1] and at INVITE time you can check that in branch_route and make your decision there. [1] https://opensips.org/html/docs/modules/3.2.x/registrar.html#param_attr_avp Regards, Vlad On 14.10.2022 17:02, Callum Guy wrote: Hi All, I'm working on a project that requires me to evaluate the user agent of a registered contact before making a decision on a current registration attempt. Is there a method to do this natively? If not, what is the best approach? I haven't found anything yet, the current options i'm evaluating are to: 1. Use mi_script module to allow me to pull the contact record via ul_show_contact 2. Use rest_get to pull this data via a web service Neither is ideal for my requirements but both would do the job, I'm hoping someone in the community has a cleaner solution before I start! Best regards, Callum *^0333 332 | x-on.co.uk <https://www.x-on.co.uk> | _**_^<https://www.linkedin.com/company/x-on> <https://www.facebook.com/XonTel> <https://twitter.com/xonuk> **^ | Coronavirus <https://www.x-on.co.uk/service/surgery-connect/coronavirus.htm> **^ | Practice Index Reviews <https://practiceindex.co.uk/gp/x-on> * *Our new office address: 22 Riduna Park, Melton IP12 1QT.* /THE COMMS COUNCIL UK AWARDS 2022 AND Best Comms Software are trade marks of the Comms Council UK, used under licence - https://commscouncil.uk/awards/ X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. Registered Office : Glebe Farm, Down Street, Dummer, Basingstoke, Hampshire, England RG25 2AD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Connect to AWS RDS database with SSL enabled
Hi Jacky,
I cant think of any workaround unfortunately.
Regards,
--
Vlad Patrascu
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 17.09.2022 18:46, jacky z wrote:
Hi Vlad,
Is there any workaround to disable the client cert? Thanks!
On Wed, Sep 14, 2022 at 9:16 PM Vlad Patrascu wrote:
Hi Jacky,
OpenSIPS will always require you to configure a client certificate
for TLS client domains and will also present that certificate when
connecting. But normally, a TLS server can simply choose not to
verify the client certificate. I don't have any experience with
AWS RDS though but it seems odd to not accept a connection only
because the client did present a certificate.
Regards,
--
Vlad Patrascu
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 14.09.2022 05:42, jacky z wrote:
Hi Bogdan-Andrei,
I checked the mariadb documentation and found mariadb has two
options to set ssl connection: two-way TSL and one-way TSL. It
seems AWS RDS only supports one-way TSL, that is, TSL is used
without a client cert. Does OPENSIPS support such one-way TSL to
connect a database? Thanks!
On Wed, Sep 14, 2022 at 12:06 AM jacky z wrote:
Hi Bogdan-Andrei,
I have set the "certificate" and "private_key" in my script,
as I explained in method 1. However, AWS RDS doesn't support
a client cert. Please refer to
https://stackoverflow.com/questions/53760104/how-to-configure-x509-client-certificate-based-authentication-to-connect-to-aws
Is there any workaround to use the public cert list provided
by AWS? Anyone has successfully used RDS with SSL
connections? Thanks!
On Tue, Sep 13, 2022 at 9:54 PM Bogdan-Andrei Iancu
wrote:
Set the certificate and key you have in the tls_mgm
module, for the "certificate" and "private_key" parameters.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Summit 27-30 Sept 2022, Athens
https://www.opensips.org/events/Summit-2022Athens/
On 9/13/22 2:57 PM, jacky z wrote:
Hi Bogdan-Andrei,
I tried two methods.
Method 1:
#enabled TLS connection:
modparam("db_mysql", "use_tls", 1)
#setup a client domain:
modparam("tls_mgm", "client_domain", "dom1")
modparam("tls_mgm", "match_ip_address", "[dom1]*")
modparam("tls_mgm", "match_sip_domain", "[dom1]*")
modparam("tls_mgm","certificate",
"[dom1]/etc/ssl/certs/rootCACert.pem")
modparam("tls_mgm","private_key",
"[dom1]/etc/ssl/private/rootCAKey.pem")
modparam("tls_mgm","ca_list",
"[dom1]/etc/ssl/certs/rootCACert.pem")
modparam("tls_mgm","tls_method", "[dom1]SSLv23")
modparam("tls_mgm","verify_cert", "[dom1]0")
modparam("tls_mgm","require_cert", "[dom1]0")
# set db_url
modparam("usrloc", "db_url",
"mysql://root:1234@/opensips?tls_domain=dom1")
...
I couldn't figure out how to use global-bundle.pem AWS
provided with this method. No luck to get a connection
with RDS. If I don't use ssl, opensips can connect to
RDS without encryption.
Method 2:
I tried
modparam("usrloc", "db_url",
"mysql://root:1234@/opensips?ssl=true&ssl_ca_certs=/etc/ssl/certs/global-bundle.pem")
to include the AWS cert. Still no luck.
Thanks!
On Tue, Sep 13, 2022 at 4:52 PM Bogdan-Andrei Iancu
wrote:
Hi,
sorry for my silly question, but how do you connect
from the OpenSIPS side ??
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Summit 27-30 Sept 2022, Athens
https://www.opensips.org/events/Summit-2022Athens/
On 9/13/22 10:41 AM, jacky z wrote:
Hi Team,
We hope to connect to aws RDS database with ssl
encryption. We have setup a client domain according
to OPENSIPS documents. However, AWS RDS does not
Re: [OpenSIPS-Users] Connect to AWS RDS database with SSL enabled
Hi Jacky,
OpenSIPS will always require you to configure a client certificate for
TLS client domains and will also present that certificate when
connecting. But normally, a TLS server can simply choose not to verify
the client certificate. I don't have any experience with AWS RDS though
but it seems odd to not accept a connection only because the client did
present a certificate.
Regards,
--
Vlad Patrascu
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 14.09.2022 05:42, jacky z wrote:
Hi Bogdan-Andrei,
I checked the mariadb documentation and found mariadb has two options
to set ssl connection: two-way TSL and one-way TSL. It seems AWS RDS
only supports one-way TSL, that is, TSL is used without a client cert.
Does OPENSIPS support such one-way TSL to connect a database? Thanks!
On Wed, Sep 14, 2022 at 12:06 AM jacky z wrote:
Hi Bogdan-Andrei,
I have set the "certificate" and "private_key" in my script, as I
explained in method 1. However, AWS RDS doesn't support a client
cert. Please refer to
https://stackoverflow.com/questions/53760104/how-to-configure-x509-client-certificate-based-authentication-to-connect-to-aws
Is there any workaround to use the public cert list provided by
AWS? Anyone has successfully used RDS with SSL connections? Thanks!
On Tue, Sep 13, 2022 at 9:54 PM Bogdan-Andrei Iancu
wrote:
Set the certificate and key you have in the tls_mgm module,
for the "certificate" and "private_key" parameters.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Summit 27-30 Sept 2022, Athens
https://www.opensips.org/events/Summit-2022Athens/
On 9/13/22 2:57 PM, jacky z wrote:
Hi Bogdan-Andrei,
I tried two methods.
Method 1:
#enabled TLS connection:
modparam("db_mysql", "use_tls", 1)
#setup a client domain:
modparam("tls_mgm", "client_domain", "dom1")
modparam("tls_mgm", "match_ip_address", "[dom1]*")
modparam("tls_mgm", "match_sip_domain", "[dom1]*")
modparam("tls_mgm","certificate",
"[dom1]/etc/ssl/certs/rootCACert.pem")
modparam("tls_mgm","private_key",
"[dom1]/etc/ssl/private/rootCAKey.pem")
modparam("tls_mgm","ca_list",
"[dom1]/etc/ssl/certs/rootCACert.pem")
modparam("tls_mgm","tls_method", "[dom1]SSLv23")
modparam("tls_mgm","verify_cert", "[dom1]0")
modparam("tls_mgm","require_cert", "[dom1]0")
# set db_url
modparam("usrloc", "db_url",
"mysql://root:1234@/opensips?tls_domain=dom1")
...
I couldn't figure out how to use global-bundle.pem AWS
provided with this method. No luck to get a connection with
RDS. If I don't use ssl, opensips can connect to RDS without
encryption.
Method 2:
I tried
modparam("usrloc", "db_url",
"mysql://root:1234@/opensips?ssl=true&ssl_ca_certs=/etc/ssl/certs/global-bundle.pem")
to include the AWS cert. Still no luck.
Thanks!
On Tue, Sep 13, 2022 at 4:52 PM Bogdan-Andrei Iancu
wrote:
Hi,
sorry for my silly question, but how do you connect from
the OpenSIPS side ??
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Summit 27-30 Sept 2022, Athens
https://www.opensips.org/events/Summit-2022Athens/
On 9/13/22 10:41 AM, jacky z wrote:
Hi Team,
We hope to connect to aws RDS database with ssl
encryption. We have setup a client domain according to
OPENSIPS documents. However, AWS RDS does not support
client cert as someone has confirmed with AWS
https://stackoverflow.com/questions/53760104/how-to-configure-x509-client-certificate-based-authentication-to-connect-to-aws
Is there any way to use the cert provided by AWS to
connect? AWS provides a global-bundle.pem
(https://docs.aws.amazon.com/AmazonRDS/latest/UserGuide/UsingWithRDS.SSL.html)
for such a connection, but we don't know how to include
it in the config file.
Thanks
Jacky z
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Re: [OpenSIPS-Users] About obstacles to implement the matched Id in event
Hi Li, In 2.4 I think the only way is to use the same cachedb_local storage for everything you need to pass around. Regards, -- Vlad Patrascu OpenSIPS Core Developer http://www.opensips-solutions.com On 18.08.2022 16:59, Li Cai wrote: Hi Vlad, Thank you very much for the suggestion. Sorry that I didn't mention our OpenSIPs version, we're with 2.4 which doesn't provide exported pseudo variables. Can I ask if there is similar way to do so in V2.4? Thanks, Li -Original Message- From: Users On Behalf Of users-requ...@lists.opensips.org Sent: August 17, 2022 3:29 AM To: users@lists.opensips.org Subject: Users Digest, Vol 169, Issue 25 WARNING: External Email: Exercise Caution Send Users mailing list submissions to users@lists.opensips.org To subscribe or unsubscribe via the World Wide Web, visit https://gcc02.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.opensips.org%2Fcgi-bin%2Fmailman%2Flistinfo%2Fusers&data=05%7C01%7Cli.cai%40comtech.com%7C502ac1e5f328416b934908da80223535%7Ca9a26e696ae040c1bd801ca6cc677828%7C0%7C0%7C637963181666956869%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata=eN7ZRHTe3r03mOVKSiQdmxv8EEwIm7QS957iAccxudA%3D&reserved=0 or, via email, send a message with subject or body 'help' to users-requ...@lists.opensips.org You can reach the person managing the list at users-ow...@lists.opensips.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Users digest..." Today's Topics: 1. Re: About obstacles to implement the matched Id in event header of NOTIFY(REFER) (Vlad Patrascu) 2. Re: Bug on TLS Management Interface (Bogdan-Andrei Iancu) -- Message: 1 Date: Tue, 16 Aug 2022 22:19:10 +0300 From: Vlad Patrascu To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] About obstacles to implement the matched Id in event header of NOTIFY(REFER) Message-ID: <91e7202d-ac0f-9ab3-252e-49640130a...@opensips.org> Content-Type: text/plain; charset="utf-8"; Format="flowed" Hi Li, For passing around data between the b2b request route and local_route, in the context of an ongoing B2B session, you can use the $b2b_logic.ctx variable[1]. That would take care of transferring the Cseq from step 1 to step 2 and the id from step 3 to step 4. As for your cseq map, you can use the cachedb_local module, and maybe the $b2b_logic.key variable alongside Cseq2 for building a proper key. [1] https://gcc02.safelinks.protection.outlook.com/?url=https%3A%2F%2Fopensips.org%2Fdocs%2Fmodules%2F3.3.x%2Fb2b_logic.html%23b2b_logic.key&data=05%7C01%7Cli.cai%40comtech.com%7C502ac1e5f328416b934908da80223535%7Ca9a26e696ae040c1bd801ca6cc677828%7C0%7C0%7C637963181666956869%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata=6QxyCjy0NEZoKBkpyy2VYKakriwsEg6xrt%2FSV%2BisERk%3D&reserved=0 Regards, -- Vlad Patrascu OpenSIPS Core Developer https://gcc02.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.opensips-solutions.com%2F&data=05%7C01%7Cli.cai%40comtech.com%7C502ac1e5f328416b934908da80223535%7Ca9a26e696ae040c1bd801ca6cc677828%7C0%7C0%7C637963181666956869%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata=dA%2Fa%2BWYy3xqc9qAGN3koDHWwBn5rWC1gRxG6PM5OebM%3D&reserved=0 On 22.07.2022 23:02, Li Cai wrote: Hi experts, I'm working on the implementation for '2.4.6 Multiple REFER Requests in a Dialog' of RFC3515 in a B2B SIP Proxy. Right now, I got stuck with two problems in the solution. Please see the paragraph from RFC- A REFER creates an implicit subscription sharing the dialog identifiers in the REFER request. If more than one REFER is issued in the same dialog (a second attempt at transferring a call for example), the dialog identifiers do not provide enough information to associate the resulting NOTIFYs with the proper REFER. Thus, for the second and subsequent REFER requests a UA receives in a given dialog, it MUST include an id parameter[2] in the Event header field of each NOTIFY containing the sequence number (the number from the CSeq header field value) of the REFER this NOTIFY is associated with. This id parameter MAY be included in NOTIFYs to the first REFER a UA receives in a given dialog. A SUBSCRIBE sent to refresh or terminate this subscription MUST contain this id parameter. Different from the definition in RFC, the NOTIFY forwarded by the proxy contains the same Id in Event header as the other side. Please see the below chart - B2BProxy | <- REFER 2 (Cseq=2) |
Re: [OpenSIPS-Users] About obstacles to implement the matched Id in event header of NOTIFY(REFER)
Hi Li,
For passing around data between the b2b request route and local_route,
in the context of an ongoing B2B session, you can use the $b2b_logic.ctx
variable[1]. That would take care of transferring the Cseq from step 1
to step 2 and the id from step 3 to step 4.
As for your cseq map, you can use the cachedb_local module, and maybe
the $b2b_logic.key variable alongside Cseq2 for building a proper key.
[1] https://opensips.org/docs/modules/3.3.x/b2b_logic.html#b2b_logic.key
Regards,
--
Vlad Patrascu
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 22.07.2022 23:02, Li Cai wrote:
Hi experts,
I’m working on the implementation for ‘2.4.6 Multiple REFER Requests
in a Dialog’ of RFC3515 in a B2B SIP Proxy. Right now, I got stuck
with two problems in the solution. Please see the paragraph from RFC-
A REFER creates an implicit subscription sharing the dialog
identifiers in the REFER request. If more than one REFER is issued
in the same dialog (a second attempt at transferring a call for
example), the dialog identifiers do not provide enough information to
associate the resulting NOTIFYs with the proper REFER.
Thus, for the second and subsequent REFER requests a UA receives in a
given dialog, it MUST include an id parameter[2] in the Event header
field of each NOTIFY containing the sequence number (the number from
the CSeq header field value) of the REFER this NOTIFY is associated
with. This id parameter MAY be included in NOTIFYs to the first
REFER a UA receives in a given dialog. A SUBSCRIBE sent to refresh
or terminate this subscription MUST contain this id parameter.
Different from the definition in RFC, the NOTIFY forwarded by the
proxy contains the same Id in Event header as the other side. Please
see the below chart –
B2BProxy
|
<- REFER 2 (Cseq=2) | <-REFER 1
(Cseq=1003)
. . . | . . .
->NOTIFY 1 (event:*id=2*) | ->NOTIFY 2 (_event*:id=2*)_
The current problem is, the above NOTIFY on the right side should
_include ‘id=1003’ instead of ‘id=2’_.
The solution I’m trying is as in the following flow –
1. get CSeq 1 of REFER 1 in route[b2b_request]{} ->
2. get CSeq 2 of REFER 2 in local_route{}, then save
the pair(key=CSeq2, value=CSeq1) in a map set->
3. get Id from NOTIFY 1 event in route[b2b_request]{} ->
4. use Id as key then get matched CSeq1 from the map
set, use remove_hf() and append_hf() to modify the event header in
local_route{}
My two questions are:
1. To transfer Cseq 1 from step1 to step2, I tried a variable in the
AVP type but it didn’t work. The script variable can work but it
doesn’t fit because the processing is based on per request. Can I
ask if you have any suggest on how to transfer the value from
route[b2b_request]{} to local_route{}?
2. In the step2, I tried to create and operate a JSON map set to save
the CSeq pairs. But the JSON data didn’t work for me.
My source :
$json(csList) = ””; #
Initialize the data set, not sure if it’s correct
$json(csList/”$cs”) = “$avp(csNum)”;
The error I got:
ERROR:core:do_assign: setting PV failed
ERROR:core:do_assign: error at /usr/local//et/opensips/opensips.cfg:531
Thank you very much for the help! Any your suggests are very welcomed.
Thanks,
Li
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[OpenSIPS-Users] [Blog] MSRP Gateway
Hi all, OpenSIPS 3.3 introduces a new MSRP gateway module, which translate between Page Mode (SIP MESSAGE method) and Session Mode (MSRP) Messaging. Read more about this in the following blog post: https://blog.opensips.org/2022/06/09/msrp-gateway/ Enjoy, -- Vlad Patrascu OpenSIPS Core Developer http://www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Stir Shaken Verification issue
Hi Devang, The URL in the info param has nothing to do with the verification itself. I suspect you are somehow not using the proper certificate and/or CA, as the certificate generated by the script you mentioned should not be self signed, as the error indicates. Regards, -- Vlad Patrascu OpenSIPS Core Developer http://www.opensips-solutions.com On 25.04.2022 15:47, Devang Dhandhalya via Users wrote: Hello All I am testing STIR/SHAKEN calls using two servers. calls originating to the first server adding identity header and when sending calls to the second server for verification service at the time of verification service i am getting below error . error :437 , Unsupported Credential , Verification Fails with Return code :-8 INFO:stir_shaken:verify_callback: certificate validation failed: self signed certificate INFO:stir_shaken:w_stir_verify: Invalid certificate OpenSIPS Version : 3.2.2 I generate certificate using domain which mapped with those 2 server : https://github.com/OpenSIPIt/OpenSIPIt_00/blob/master/STIR_SHAKEN/Certgen/gencert.sh <https://github.com/OpenSIPIt/OpenSIPIt_00/blob/master/STIR_SHAKEN/Certgen/gencert.sh> When the same server generates an identity header and verifies it at that time not getting an issue call is working fine but when the identity header generated by server 1 and going to verify it by server 2 we get this above error. Is it related to the URL which is in the info param ? When I open that URL in the browser I am able to see the certificate. Please suggest a solution for this issue. Regards Devang Dhandhalya *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] is_from_gw() DNS Names
Hi Mark,
We are aware of this limitation with wolfssl, and do plan to address it
somehow but we have not found a straight-forward solution yet. Keep an
eye on the feature request Ovidiu mentioned.
Regards,
--
Vlad Patrascu
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 28.02.2022 10:50, Mark Farmer wrote:
Thanks Ovidiu, that is great information.
I am using wolfssl as that seems to be the way to go these days.
I wonder given the rising popularity of Direct Routing if it would be
possible/sensible to have wolfsssl populate the $tls_peer_subject_cn
variable in the future?
Mark.
On Fri, 25 Feb 2022 at 17:32, Ovidiu Sas wrote:
With MS, you can authenticate based on $tls_peer_subject_cn. This
works ok with openssl but not with wolfssl. When wolfssl is using
session tickets to establish new connections, the $tls_peer_subject_cn
is not populated.
Another alternative is to perform a lookup for each request received
over a tls connection using the ip.resolve transformation and enable
dbs_cache to help a little bit. It's messy but it works.
-ovidiu
On Fri, Feb 25, 2022 at 6:51 AM Mark Farmer wrote:
>
> Thanks Bogdan
>
> It's no secret really, I was just speaking generically.
> They are the MS Direct Routing domains, EG
sip.pstnhub.microsoft.com <http://sip.pstnhub.microsoft.com>
>
> Mark.
>
>
>
> On Tue, 22 Feb 2022 at 12:50, Bogdan-Andrei Iancu
wrote:
>>
>> Hi Mark,
>>
>> You say the DNS is publishing only one IP for the domain, but
one may change ? If you want, you can PM me the actual domain to
see how the DNS records looks like.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>> https://www.opensips-solutions.com
>> OpenSIPS eBootcamp
>> https://www.opensips.org/Training/Bootcamp
>>
>> On 2/22/22 12:31 PM, Mark Farmer wrote:
>>
>> Hi Bogdan
>>
>> The GW's have 2 CNAME records which I have no control over. DR
has entries like subdomain.example.com:5061
<http://subdomain.example.com:5061>
>> I suspect the issue arises when the CNAMES swap around
resulting in a mismatch.
>>
>> Currently I am using this to identify the source of the message
which is probably not the best in terms of security.
>>
>> $avp(fd) = "subdomain.example.com <http://subdomain.example.com>";
>> if($(ct.fields(uri){s.index, $avp(fd)}) != NULL)
>>
>> Perhaps there is a better way?
>>
>> Best regards
>> Mark.
>>
>>
>>
>> On Tue, 22 Feb 2022 at 08:56, Bogdan-Andrei Iancu
wrote:
>>>
>>> Hi Mark,
>>>
>>> If a gw is defined via FQDN, that will by DNS resolved (NAPTR,
SRV, A records) when DB data is (re)loaded by DR module, and used
later for such checks. All found IPs (from DNS) will be stored on
the GW.
>>>
>>> How do you specify the GW address in DB and what kind of DNS
records do you have for it ?
>>>
>>> Best regards,
>>>
>>> Bogdan-Andrei Iancu
>>>
>>> OpenSIPS Founder and Developer
>>> https://www.opensips-solutions.com
>>> OpenSIPS eBootcamp
>>> https://www.opensips.org/Training/Bootcamp
>>>
>>> On 2/18/22 6:04 PM, Mark Farmer wrote:
>>>
>>> Hi everyone
>>>
>>> I am using is_from_gw() to match against a group of gateways
specified by DNS names which resolve to multiple IP addresses but
it seems to be failing to match.
>>>
>>> Is this supported functionality or do I need to do something
else in this case?
>>>
>>> Thanks and regards
>>> Mark.
>>>
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>> --
>> Mark Farmer
>> farm...@gmail.com
>>
>>
>
>
> --
> Mark Farmer
> farm...@gmail.com
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Re: [OpenSIPS-Users] Managing the implicit subscription and its NOTIFYs within B2BUA REFER handling
Hi Jeff, You can use proxy commands for different traffic flows but keep in mind that once you trigger a B2B session in the main request route (with the b2b_init_request() function), you will only see the rest of the B2B related traffic in the dedicated routes. In those routes it is recommended to not use proxy-like signaling functions. Regards, Vlad .-- Vlad Patrascu OpenSIPS Core Developer http://www.opensips-solutions.com On 28.01.2022 17:30, Jeff Pyle wrote: Great. Is it still recommended to keep a particular OpenSIPS instance either all proxy commands or all B2B commands? Or, do we have some flexibility to mix them since B2B commands live in the script now? - Jeff On Wed, Jan 26, 2022 at 9:50 AM Vlad Patrascu wrote: Hi Jeff, Sending NOTIFYs is supported by using the "n" flag for the b2b_bridge() [1] function. Just note that some mechanisms from RFC 3515 2.4.4 <https://datatracker.ietf.org/doc/html/rfc3515#section-2.4.4> are not implemented though, such as the ability to terminate the subscription prematurely by unsubscribing or refreshing the subscription. [1] https://opensips.org/docs/modules/3.2.x/b2b_logic.html#func_b2b_bridge Regards, -- Vlad Patrascu OpenSIPS Core Developer http://www.opensips-solutions.com On 24.01.2022 17:38, Jeff Pyle wrote: OpenSIPS 3.2 supports some slick in-script B2B functions, documented here [1] with an example for handling REFER. Does this approach include support for the subscription and the subsequent NOTIFYs required by RFC 3515 2.4.4 [2] <https://datatracker.ietf.org/doc/html/rfc3515#section-2.4.4>? I have a need to write some REFER handling functions into an existing OpenSIPS SBC stack, and I need to send accurate NOTIFYs back to the REFER-er. Terminating the subscription with the first NOTIFY isn't good enough, sadly. I'm hoping this is the right way to do it. [1] https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2#toc10 [2] https://datatracker.ietf.org/doc/html/rfc3515#section-2.4.4 - Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Managing the implicit subscription and its NOTIFYs within B2BUA REFER handling
Hi Jeff, Sending NOTIFYs is supported by using the "n" flag for the b2b_bridge() [1] function. Just note that some mechanisms from RFC 3515 2.4.4 <https://datatracker.ietf.org/doc/html/rfc3515#section-2.4.4> are not implemented though, such as the ability to terminate the subscription prematurely by unsubscribing or refreshing the subscription. [1] https://opensips.org/docs/modules/3.2.x/b2b_logic.html#func_b2b_bridge Regards, -- Vlad Patrascu OpenSIPS Core Developer http://www.opensips-solutions.com On 24.01.2022 17:38, Jeff Pyle wrote: OpenSIPS 3.2 supports some slick in-script B2B functions, documented here [1] < [1] https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2#toc10> with an example for handling REFER. Does this approach include support for the subscription and the subsequent NOTIFYs required by RFC 3515 2.4.4 [2] <https://datatracker.ietf.org/doc/html/rfc3515#section-2.4.4>? I have a need to write some REFER handling functions into an existing OpenSIPS SBC stack, and I need to send accurate NOTIFYs back to the REFER-er. Terminating the subscription with the first NOTIFY isn't good enough, sadly. I'm hoping this is the right way to do it. [1] https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2#toc10 <https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2#toc10> [2] https://datatracker.ietf.org/doc/html/rfc3515#section-2.4.4 <https://datatracker.ietf.org/doc/html/rfc3515#section-2.4.4> - Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Python and avp
Hi Alberto, You should be able to call any opensips script function from the python code, including pv_printf() as suggested in #1893, by using the /call_function()/ method [1]. This has been made possible from 3.1 onwards, but indirectly, by porting all opensips core functions to the same calling interface as module functions. [1] https://opensips.org/docs/modules/3.2.x/python.html#overview Regards, -- Vlad Patrascu OpenSIPS Core Developer http://www.opensips-solutions.com On 17.01.2022 20:14, Alberto wrote: Hi, I'm using opensips 3.4 and I'm looking to set avp variables from a python script, looking at https://github.com/OpenSIPS/opensips/issues/1893 <https://github.com/OpenSIPS/opensips/issues/1893> seems this feature was available from after 3.0. Is there any documentation on how to do it? Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Compiliing issue OpenSIPS 3.2.4 - WolfSSL on Debian 11
Hi Eugen,
You probably just need to update the wolfssl git submodule, do:
make clean
git submodule update --init
Regards,
--
Vlad Patrascu
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 18.01.2022 15:50, Eugen Prieb via Users wrote:
Hello,
i have a problem with compiling OpenSIPS 3.2.4 on Debian 11. Issue is
on WolfSSL compiling...
make[1]: Entering directory '/usr/src/opensips/modules/tls_wolfssl'
Compiling wolfssl.c
wolfssl.c: In function ‘oss_mutex_cb’:
wolfssl.c:140:7: error: ‘WOLFSSL_USER_MUTEX_INIT’ undeclared (first
use in this function)
140 | case WOLFSSL_USER_MUTEX_INIT:
| ^~~
wolfssl.c:140:7: note: each undeclared identifier is reported only
once for each function it appears in
wolfssl.c:141:4: error: ‘wolfSSL_Mutex’ {aka ‘pthread_mutex_t’} has no
member named ‘mutex’
141 | m->mutex = lock_alloc();
| ^~
wolfssl.c:142:9: error: ‘wolfSSL_Mutex’ {aka ‘pthread_mutex_t’} has no
member named ‘mutex’
142 | if (!m->mutex || !lock_init(m->mutex)) {
| ^~
wolfssl.c:142:32: error: ‘wolfSSL_Mutex’ {aka ‘pthread_mutex_t’} has
no member named ‘mutex’
142 | if (!m->mutex || !lock_init(m->mutex)) {
| ^~
wolfssl.c:147:7: error: ‘WOLFSSL_USER_MUTEX_FREE’ undeclared (first
use in this function)
147 | case WOLFSSL_USER_MUTEX_FREE:
| ^~~
In file included from wolfssl.c:34:
wolfssl.c:149:17: error: ‘wolfSSL_Mutex’ {aka ‘pthread_mutex_t’} has
no member named ‘mutex’
149 | lock_dealloc(m->mutex);
| ^~
../../mem/shm_mem.h:513:38: note: in definition of macro ‘shm_free’
513 | #define shm_free( _ptr ) _shm_free( (_ptr), \
| ^~~~
wolfssl.c:149:3: note: in expansion of macro ‘lock_dealloc’
149 | lock_dealloc(m->mutex);
| ^~~~
wolfssl.c:150:4: error: ‘wolfSSL_Mutex’ {aka ‘pthread_mutex_t’} has no
member named ‘mutex’
150 | m->mutex = NULL;
| ^~
wolfssl.c:152:7: error: ‘WOLFSSL_USER_MUTEX_LOCK’ undeclared (first
use in this function)
152 | case WOLFSSL_USER_MUTEX_LOCK:
| ^~~
In file included from ../../mem/shm_mem.h:50,
from wolfssl.c:34:
wolfssl.c:153:13: error: ‘wolfSSL_Mutex’ {aka ‘pthread_mutex_t’} has
no member named ‘mutex’
153 | lock_get(m->mutex);
| ^~
../../mem/../lock_ops.h:93:34: note: in definition of macro ‘lock_get’
93 | #define lock_get(lock) get_lock(lock)
| ^~~~
wolfssl.c:155:7: error: ‘WOLFSSL_USER_MUTEX_UNLOCK’ undeclared (first
use in this function)
155 | case WOLFSSL_USER_MUTEX_UNLOCK:
| ^
In file included from ../../mem/shm_mem.h:50,
from wolfssl.c:34:
wolfssl.c:156:17: error: ‘wolfSSL_Mutex’ {aka ‘pthread_mutex_t’} has
no member named ‘mutex’
156 | lock_release(m->mutex);
| ^~
../../mem/../lock_ops.h:90:41: note: in definition of macro
‘lock_release’
90 | #define lock_release(lock) release_lock(lock)
| ^~~~
wolfssl.c: In function ‘mod_init’:
wolfssl.c:172:2: warning: implicit declaration of function
‘wolfSSL_SetUserMutexCb’; did you mean ‘wolfSSL_SetHsDoneCb’?
[-Wimplicit-function-declaration]
172 | wolfSSL_SetUserMutexCb(oss_mutex_cb);
| ^~
| wolfSSL_SetHsDoneCb
make[1]: *** [../../Makefile.rules:28: wolfssl.o] Error 1
make[1]: Leaving directory '/usr/src/opensips/modules/tls_wolfssl'
make: *** [Makefile:197: modules] Error 2-
maili
--
Eugen P.
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Re: [OpenSIPS-Users] STIR/SHAKEN E.164 strict mode module parameter not working .
Hello Devang,
The cache_fetch() functions returns -2 when the key is not found so it's
normal when getting the certificate the first time.
STIR/SHAKEN requires that the certificates have the TnAuthList
extension. You can check out some examples on how to generate such
certificates in this document [1] at Appendix A or this script [2].
[1]
https://cstga.ca/wp-content/uploads/2020/07/ATIS-180.v002_SHAKEN-Governance-Model.pdf
[2]
https://github.com/OpenSIPIt/OpenSIPIt_00/blob/master/STIR_SHAKEN/Certgen/gencert.sh
Regards,
--
Vlad Patrascu
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 22.11.2021 14:41, Devang Dhandhalya wrote:
Hello Vlad
Authentication service works fine. Now I am getting an error in the
stir_shaken_verify function , after fetching a certificate, I am
getting below error . I generated private keys and X509 certificates
using prime256v1" EC.
After fetching the certificate I am getting var(found) = -2 can you
please tell me about this too .
OpenSIPS Logs :
*
*
*var(found) = [-2]
INFO:stir_shaken:validate_certificate: The certificate is missing the
TnAuthList extension
INFO:stir_shaken:w_stir_verify: Invalid certificate
return code : [-8] stir_shaken_verify() failed:437, Unsupported Credential
*
below opensips script snippet :
$var(found) = cache_fetch("local",$identity(x5u),$var(cert));
if (!$var(found) || !stir_shaken_check_cert("$var(cert)")) {
rest_get("$identity(x5u)",$var(cert),$var(ctype), $var(http_rc));
if ($rc<0 ) {
send_reply(436, "Bad Identity Info");
exit;
}
cache_store("local", $identity(x5u),"$var(cert)");
}
$var(rc_verify)= stir_shaken_verify($var(cert),
$var(err_sip_code),$var(err_sip_reason),"$var(orig)","$var(dest)");
if ($var(rc_verify) < 0) {
send_reply(
$var(err_sip_code),$var(err_sip_reason));
exit;
}
Kindly inform me why I am getting this error . Please suggest a
solution to this .
Regards
Devang Dhandhalya
On Fri, Nov 19, 2021 at 3:16 PM Devang Dhandhalya
mailto:devang.dhandha...@ecosmob.com>>
wrote:
Hello Vlad
Thanks for your response . Authentication service works .
Regards ,
Devang Dhandhalya
On Wed, Nov 17, 2021 at 8:57 PM Devang Dhandhalya
mailto:devang.dhandha...@ecosmob.com>> wrote:
hello all
Above E.164 Error still getting .Right now I'm getting the below error .
Can anyone tell me why I am getting this error ? as far as i know this
error for x5u parameter in stir_shaken_auth function , this issue coming
for certificate path or certificate file format .
I check the certificate file with .der and .cer format also .
Here is the code snippet used .
$var(rc_auth)=stir_shaken_auth("A",
"GWID-123456","$var(cert)",
"$var(pkey)","http://localhost/certificate.pem
<http://localhost/certificate.pem>","$var(orig)","$var(dest)");
Below Error i am getting .
ERROR:stir_shaken:add_identity_hf: Failed to convert from DER to
internal format
ERROR:stir_shaken:w_stir_auth: Failed to add Identity header
STIR_SHAKEN AUTHENTICATION SERVICE return code : -1
Kindly let me know if there is something wrong that I could be
doing. Many Thanks Devang Dhandhalya
On Wed, Nov 17, 2021 at 11:37 AM Devang Dhandhalya
mailto:devang.dhandha...@ecosmob.com>> wrote:
Hi All
I configured the e164 strict mode module parameter as 0
(disabled) . but still i am getting errors related to its
e164 format .While if orig/dest number is not in e164
format then also opensips have to accept it but it is not
accepting . I have a user like extension123 for this
function I have to perform authentication service . if i
have a user extension123 is it possible to perform
authenticate service for this kind of user ?
I think this is a bug for the e164 strict mode module
parameter . I am getting the below error .
opensips version : 3.2.2
ERROR :
ERROR:stir_shaken:check_passport_phonenum: number is not
in E.164 format: extension123
ERROR:stir_shaken:w_stir_auth: failed to validate
Originator number (extension123)
loadmodule "stir_shaken.so"
modparam("stir_shaken", "auth_date_freshness", 300)
modparam("stir_shaken", "verify_date_freshness", 300)
modpar
Re: [OpenSIPS-Users] TLS Handshake fail issue
Hi Devang, The tls_mgm module configuration looks OK. Have you also tried to connect with openssl s_client to the TLS endpoint? Regards, -- Vlad Patrascu OpenSIPS Core Developer http://www.opensips-solutions.com On 17.11.2021 10:33, Devang Dhandhalya wrote: Hello Bogdan I Just casually write , My intention was not rude or anything else , still it's my bad if any one feeling hurts . as you say the party you are trying to connect to (1.2.3.4:40945 <http://1.2.3.4:40945/>) is not accepting your connection. i check that but can you please tell me what type of thing i have to check . means from opensips side or blink user configuration i have add tls certificate also in user configuration . In the opensips module parameter any configuration wrong for TLS service ? Many Thanks Devang *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] STIR/SHAKEN E.164 strict mode module parameter not working .
Hi Devang,
The error:
ERROR:stir_shaken:add_identity_hf: Failed to convert from DER to internal format
may be a bit misleading as the private key and certificate must be
provided in PEM format to the stir_shaken_auth() function. The DER
format that the error refers to is used only internally, for an
intermediary step. On the other hand, you may get the same error if the
private key is not generated as an "prime256v1" EC key.
Regarding the e164_strict_mode module parameter, the orig/dest number
should still be in E.164 format, except the leading '+' sign, which is
not required when the parameter is turned off. The parameter is also
related to the type of URI required (tel URI / sip URI with the
"user=phone" parameter).
Regards,
Vlad
--
Vlad Patrascu
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 17.11.2021 17:27, Devang Dhandhalya wrote:
hello all
Above E.164 Error still getting .Right now I'm getting the below error .
Can anyone tell me why I am getting this error ? as far as i know this
error for x5u parameter in stir_shaken_auth function , this issue coming
for certificate path or certificate file format .
I check the certificate file with .der and .cer format also .
Here is the code snippet used .
$var(rc_auth)=stir_shaken_auth("A", "GWID-123456","$var(cert)",
"$var(pkey)","http://localhost/certificate.pem
<http://localhost/certificate.pem>","$var(orig)","$var(dest)");
Below Error i am getting .
ERROR:stir_shaken:add_identity_hf: Failed to convert from DER to internal format
ERROR:stir_shaken:w_stir_auth: Failed to add Identity header
STIR_SHAKEN AUTHENTICATION SERVICE return code : -1
Kindly let me know if there is something wrong that I could be doing.
Many Thanks Devang Dhandhalya
On Wed, Nov 17, 2021 at 11:37 AM Devang Dhandhalya
mailto:devang.dhandha...@ecosmob.com>>
wrote:
Hi All
I configured the e164 strict mode module parameter as 0 (disabled)
. but still i am getting errors related to its e164 format .While
if orig/dest number is not in e164 format then also opensips have
to accept it but it is not accepting . I have a user like
extension123 for this function I have to perform authentication
service . if i have a user extension123 is it possible to perform
authenticate service for this kind of user ?
I think this is a bug for the e164 strict mode module parameter .
I am getting the below error .
opensips version : 3.2.2
ERROR :
ERROR:stir_shaken:check_passport_phonenum: number is not in E.164
format: extension123
ERROR:stir_shaken:w_stir_auth: failed to validate Originator
number (extension123)
loadmodule "stir_shaken.so"
modparam("stir_shaken", "auth_date_freshness", 300)
modparam("stir_shaken", "verify_date_freshness", 300)
modparam("stir_shaken", "require_date_hdr", 0)
modparam("stir_shaken", "e164_strict_mode", 0)
$var(orig) = $fU;
$var(dest) = $tU
$var(rc_auth)=stir_shaken_auth("A", "GWID-123456","$var(cert)",
"$var(pkey)","http://localhost/certificate.pem
<http://localhost/certificate.pem>","$var(orig)","$var(dest)");
Please suggest a solution to this .
Many Thanks
Devang
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Re: [OpenSIPS-Users] 3.2.3 TLS issue
Hi Gregory, It is difficult to draw any conclusions without a backtrace so I'm hoping you'll be able to get the proper core dumps again in order to investigate this further. Regards, -- Vlad Patrascu OpenSIPS Core Developer http://www.opensips-solutions.com On 12.11.2021 14:44, Gregory Massel wrote: Sorry, I do have the core files, however, my /usr/sbin/opensips is now 3.2.2 and the core dumps were taken against 3.2.3. I should have thought to get the backtrace before downgrading. This is a production server, however, I will try and schedule some downtime and then restore 3.2.3 to generate the backtrace and will try out the nightly release. --Greg ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 3.2.0 TLS MGM module does not load 3.1.0 domain configuration
Hi Adrian,
The wolfSSL implementation does not support a TLS method range, such as
"TLSv1-", so that could be one of the causes. What seems strange is that
there is no warning message: "WARNING:tls_wolfssl:tls_get_method:
wolfSSL does not support method range specification" which should be
thrown in such cases.
Regards,
--
Vlad Patrascu
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 12.08.2021 20:12, Adrian Georgescu wrote:
Hi,
I am using the latest 3.2.0 build with the old TLS configuration, with the aim
to try out Wolf SSL stack.
But while the config check passed, the server does not start with the old
configuration:
loadmodule “tls_mgm.so"
loadmodule “tls_wolfssl.so"
modparam("tls_mgm", "client_tls_domain_avp", "tls_client_domain")
modparam("tls_mgm", "tls_library", "auto”)
modparam("tls_mgm", "server_domain","ag-projects-server")
modparam("tls_mgm", "match_ip_address", "[ag-projects-server]*")
modparam("tls_mgm", "match_sip_domain", "[ag-projects-server]ag-projects.com")
modparam("tls_mgm", "tls_method", "[ag-projects-server]TLSv1-")
modparam("tls_mgm", "certificate",
"[ag-projects-server]/etc/opensips/tls/ag-projects.crt")
modparam("tls_mgm", "private_key",
"[ag-projects-server]/etc/opensips/tls/ag-projects.key")
modparam("tls_mgm", "ca_list",
"[ag-projects-server]/etc/opensips/tls/ca-list.pem")
modparam("tls_mgm", "ca_dir", "[ag-projects-server]/etc/ssl/certs")
modparam("tls_mgm", "verify_cert", "[ag-projects-server]1")
modparam("tls_mgm", "require_cert", "[ag-projects-server]0")
modparam("tls_mgm", "client_domain","ag-projects-client")
modparam("tls_mgm", "match_ip_address", "[ag-projects-client]*")
modparam("tls_mgm", "match_sip_domain", "[ag-projects-client]ag-projects.com")
modparam("tls_mgm", "tls_method", "[ag-projects-client]TLSv1-")
modparam("tls_mgm", "certificate",
"[ag-projects-client]/etc/opensips/tls/ag-projects.crt")
modparam("tls_mgm", "private_key",
"[ag-projects-client]/etc/opensips/tls/ag-projects.key")
modparam("tls_mgm", "ca_list",
"[ag-projects-client]/etc/opensips/tls/ca-list.pem")
modparam("tls_mgm", "ca_dir", "[ag-projects-client]/etc/ssl/certs")
modparam("tls_mgm", "verify_cert", "[ag-projects-client]1")
modparam("tls_mgm", "require_cert", "[ag-projects-client]0”)
Aug 12 18:51:14 live01 opensips[6455]: Aug 12 18:51:14 [6455]
DBG:core:set_mod_param_regex: tls_mgm matches module tls_mgm
Aug 12 18:51:14 live01 opensips[6455]: Aug 12 18:51:14 [6455]
DBG:core:set_mod_param_regex: found in module tls_mgm
[/usr/lib/x86_64-linux-gnu/opensips/modules/]
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455]
DBG:core:solve_module_dependencies: solving dependency tls_mgm -> module
tls_wolfssl
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455]
DBG:core:solve_module_dependencies: solving dependency tls_mgm -> module
tls_openssl
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455]
DBG:core:solve_module_dependencies: module tls_mgm soft-depends on module
tls_openssl, and it was not loaded -- continuing
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455]
DBG:core:solve_module_dependencies: solving dependency proto_tls -> module
tls_mgm
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455]
DBG:core:init_mod: initializing module tls_mgm
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455]
INFO:tls_mgm:mod_init: initializing TLS management
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455]
DBG:tls_mgm:load_info: 0 rows found in tls_mgm
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455]
DBG:tls_mgm:load_info: 0 records found in tls_mgm
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455]
INFO:tls_mgm:init_tls_dom: Processing TLS domain 'ag-projects-server'
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455]
DBG:tls_mgm:init_tls_dom: no DH params file for tls domain 'ag-projects-server'
defined, using default '(null)'
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455]
DBG:tls_mgm:init_tls_dom: cipher list null ... setting default
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455]
NOTICE:tls_mgm:init_tls_dom: no crl for tls, using none
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455]
ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'ag-projects-server'
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455]
ERROR:core:init_mod: failed to initialize module tls_mgm
Any ideas what am I doing wrong?
Adrian
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Re: [OpenSIPS-Users] Removing ISUP part from sip dialog
Hi Alexey,
The remove_body_part() function should work for any method and replies
as well. So are you saying that you have actually tried the function and
it has no effect ?
Regards,
--
Vlad Patrascu
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 28.07.2021 18:15, Alexey Hamstr wrote:
good day.
Could anybody advise me if it is possible to remove ISUP part from
whole SIP dialogue
Now I have only found that there is no problem to remove it from the
common methods
like
if (is_method("INVITE") ) {
remove_body_part("application/isup");
}
but i see ISUP part in the 180 or 186 messages or in the BYE method
Thanks.
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[OpenSIPS-Users] [BLOG] OpenSIPS 3.2 and the supported SSL/TLS libraries
Hello everyone, Check out this article [1] about your options regarding SSL/TLS libraries in OpenSIPS 3.2. Do not hesitate to try the new wolfSSL-based TLS implementation in OpenSIPS! [1] https://blog.opensips.org/2021/06/03/opensips-3-2-and-the-supported-ssl-tls-libraries/ Cheers, -- Vlad Patrascu OpenSIPS Core Developer http://www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Usage of TLS require_cert and verify_cert settings
Hi Adrian, It is not possible to change those parameters on the fly. But I think you can implement your scenarios by setting a domain with require_cert=0, verify_cert=0 and using the is_peer_verified() script function to check the connections from the remote server. Regards, -- Vlad Patrascu OpenSIPS Core Developer http://www.opensips-solutions.com On 22.04.2021 17:15, Adrian Georgescu wrote: Hello, I have a question. I have the following TLS scenarios: 1) A local user for a domain I own, connects to my server using TLS. If the domain is local, I will authenticate the user against my database and I do not care if the user has a certificate 2) A remote server, connects to my server using TLS and pretends that is domainX.com <http://domainX.com>. In such case, the only way to verify that this is true is by requiring a certificate and verify it So there is a logic split between when to require and how to verify a certificate depending on the fact that we deal with a local user or a foreign domain. I would like to know if is possible to set require_cert and verify_cert on the fly, while routing packets, instead of configuring them statically per domain. Regards, Adrian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] $b2b_logic.entity(id) in [reply_route] B2B module opensips 3.2
HI Xaled,
The variable should normally be available in b2b_logic's reply route. Do
you get the same result for the 200 OK reply or other replies? Also, is
the other information about the entity available
($b2b_logic.entity(callid), $b2b_logic.entity(key)) ?
Regards,
--
Vlad Patrascu
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 27.04.2021 23:28, xa...@web.de wrote:
Hi,
I'm not getting the $b2b_logic.entity(id) in B2B [reply_route]. The log shows
Is this variable not available in the B2B [reply_route]?
route[b2b_announce_response] {
if ($rs != 500)
{
xlog("B2B REPLY: [$rs $rm] from entity: $b2b_logic.entity(id)\n");
b2b_handle_reply();
exit;
}
B2B REPLY: [183 INVITE] from entity:
Thanks,
Xaled
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[OpenSIPS-Users] [Blog] Clustering Engine Improvements in OpenSIPS 3.2
Hi all, Take a look at the latest blog post [1] that presents the new Clustering support improvements brought by the upcoming OpenSIPS 3.2 release. [1] https://blog.opensips.org/2021/04/22/clustering-engine-improvements-in-opensips-3-2/ Enjoy! -- Vlad Patrascu OpenSIPS Core Developer http://www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Generate CANCEL on 180
Hello,
In 3.1, create_dialog [1] supports passing it the 'E' flag which will
lead to the call termination if a dialog race condition occurs.
The dialog will be ended after race_condition_timeout [2] seconds.
Some of the most frequent race conditions are documented in RFC 5407
[3], and the dialog module currently supports terminating the call in
case races 3.1.2 and 3.1.3 from the RFC occur.
[1]
https://opensips.org/html/docs/modules/3.1.x/dialog.html#func_create_dialog
<https://opensips.org/html/docs/modules/3.1.x/dialog.html#func_create_dialog>
[2]
https://opensips.org/html/docs/modules/3.1.x/dialog.html#race_condition_timeout
<https://opensips.org/html/docs/modules/3.1.x/dialog.html#race_condition_timeout>
[3] https://tools.ietf.org/html/rfc5407
<https://tools.ietf.org/html/rfc5407>
On 21.04.2021 15:52, Antonis Psaras wrote:
Hello Vlad
I am using 2.4 so race_condition_timeout is not available but I can upgrade if
required.
Actually is not very clear to me what that parameter does. Can you explain a
bit more?
Regards
Antonis Psaras / Managing Director
-Original Message-
From: Users On Behalf Of Vlad Paiu
Sent: Τετάρτη, 21 Απριλίου 2021 15:45
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180
Hello,
Try using the race_condition_timeout dialog param along with the 'E'
flag when creating the dialog.
https://opensips.org/html/docs/modules/3.1.x/dialog.html#race_condition_timeout
<https://opensips.org/html/docs/modules/3.1.x/dialog.html#race_condition_timeout>
Best Regards,
Vlad
On 21.04.2021 12:46, Antonis Psaras wrote:
Hello Bogdan
The flow is the following
INVITE
Trying (instantly)
(after 2sec)
183
(after 3sec)
180
(here I do the process but during that I receive)
200 (instantly)
(here the CANCEL is sent)
CANCEL
And the call is keep going, hence the CANCEL was ignored by the carrier by the
carrier because the call was answered.
Regards
Antonis Psaras
-Original Message-
From: Bogdan-Andrei Iancu
Sent: Τετάρτη, 21 Απριλίου 2021 12:32
To: apsa...@microbase.gr; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180
Hi Antonis,
What exactly does not work ? sending the CANCEL out? or the callee "refuses" to
cancel and sends a 200 OK ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Bootcamp 2021 online
https://opensips.org/training/OpenSIPS_eBootcamp_2021/
On 4/20/21 11:13 PM, Antonis Psaras wrote:
I did the following
if (t_check_status("180"))
{
t_cancel_branch();
drop;
}
But there is an issue.
When 180 is followed by 200 instantly, the CANCEL is not working as expected.
When I add a delay on Answer ie 1sec then CANCEL works.
Any suggestion?
Antonis Psaras
-Original Message-
From: Users On Behalf Of Kingsley
Tart
Sent: Τρίτη, 20 Απριλίου 2021 20:10
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180
Firstly, I'm new to OpenSIPS so treat my comments accordingly.
But, can you do something in an onreply route?
eg, in a test setup I have, when I get an INVITE I do this:
create_dialog("pPB");
t_on_reply("doodle");
(I can't remember whether the dialog is needed for this)
and then I have this:
onreply_route[doodle] {
# expect $T_reply_code to likely first be 100
# then 180 or 183 for a progressing call
# 200 when call is answered
# or failure code (eg 4xx) or whatever
if (t_check_status("^1[0-9][0-9]$")) {
switch ($T_reply_code) {
case 180: $acc_extra(t_ringing) = $Ts; break;
case 183: $acc_extra(t_progress) = $Ts; break;
}
} else if (t_check_status("^2[0-9][0-9]$")) {
$acc_extra(t_answer) = $Ts;
} else {
xlog("Something else\n");
}
}
so when a 180 is received, it calls the above route function. Could you send a
CANCEL from there?
Cheers,
Kingsley.
On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote:
Dear all
I am trying to create a service which will generate missed calls. In
order to be more accurate, I want to CANCEL the request when 180 is
received.
The scenario is the following
Asterisk Invite -> OpenSIPs -> Carrier
Carrier 183 -> OpenSIPs -> Asterisk
Carrier 180 -> OpenSIPs
OpenSIPs Cancel -> Carrier
Is that possible to be done from script without external app?
Regards
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Re: [OpenSIPS-Users] Generate CANCEL on 180
Hello,
Try using the race_condition_timeout dialog param along with the 'E'
flag when creating the dialog.
https://opensips.org/html/docs/modules/3.1.x/dialog.html#race_condition_timeout
<https://opensips.org/html/docs/modules/3.1.x/dialog.html#race_condition_timeout>
Best Regards,
Vlad
On 21.04.2021 12:46, Antonis Psaras wrote:
Hello Bogdan
The flow is the following
INVITE
Trying (instantly)
(after 2sec)
183
(after 3sec)
180
(here I do the process but during that I receive)
200 (instantly)
(here the CANCEL is sent)
CANCEL
And the call is keep going, hence the CANCEL was ignored by the carrier by the
carrier because the call was answered.
Regards
Antonis Psaras
-Original Message-
From: Bogdan-Andrei Iancu
Sent: Τετάρτη, 21 Απριλίου 2021 12:32
To: apsa...@microbase.gr; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180
Hi Antonis,
What exactly does not work ? sending the CANCEL out? or the callee "refuses" to
cancel and sends a 200 OK ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Bootcamp 2021 online
https://opensips.org/training/OpenSIPS_eBootcamp_2021/
On 4/20/21 11:13 PM, Antonis Psaras wrote:
I did the following
if (t_check_status("180"))
{
t_cancel_branch();
drop;
}
But there is an issue.
When 180 is followed by 200 instantly, the CANCEL is not working as expected.
When I add a delay on Answer ie 1sec then CANCEL works.
Any suggestion?
Antonis Psaras
-Original Message-
From: Users On Behalf Of Kingsley
Tart
Sent: Τρίτη, 20 Απριλίου 2021 20:10
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180
Firstly, I'm new to OpenSIPS so treat my comments accordingly.
But, can you do something in an onreply route?
eg, in a test setup I have, when I get an INVITE I do this:
create_dialog("pPB");
t_on_reply("doodle");
(I can't remember whether the dialog is needed for this)
and then I have this:
onreply_route[doodle] {
# expect $T_reply_code to likely first be 100
# then 180 or 183 for a progressing call
# 200 when call is answered
# or failure code (eg 4xx) or whatever
if (t_check_status("^1[0-9][0-9]$")) {
switch ($T_reply_code) {
case 180: $acc_extra(t_ringing) = $Ts; break;
case 183: $acc_extra(t_progress) = $Ts; break;
}
} else if (t_check_status("^2[0-9][0-9]$")) {
$acc_extra(t_answer) = $Ts;
} else {
xlog("Something else\n");
}
}
so when a 180 is received, it calls the above route function. Could you send a
CANCEL from there?
Cheers,
Kingsley.
On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote:
Dear all
I am trying to create a service which will generate missed calls. In
order to be more accurate, I want to CANCEL the request when 180 is
received.
The scenario is the following
Asterisk Invite -> OpenSIPs -> Carrier
Carrier 183 -> OpenSIPs -> Asterisk
Carrier 180 -> OpenSIPs
OpenSIPs Cancel -> Carrier
Is that possible to be done from script without external app?
Regards
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Re: [OpenSIPS-Users] use of scenario_id in b2b_init_request
Hi Xaled,
The scenario ID is just a transparent identifier for the current B2B
session. You may use it to identify which session belongs to which of
your B2B scenarios. You can see this value in the output of b2b_list MI
function and also you can retrieve it in the script with
$b2b_logic.scenario (this may be useful if you use a single B2B request
route, as in the example here: [1]).
Another purpose of the "scenario_id" parameter of b2b_init_request() is
to be able to choose the internal topology hiding scenario, by providing
the special value "top hiding".
[1]
https://opensips.org/docs/modules/3.2.x/b2b_logic.html#b2b_logic.scenario
Regards,
--
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 09.04.2021 18:49, xa...@web.de wrote:
Hi,
I'm trying to understand the B2B module in 3.2 and don't quite get the meaning
of scenario_id in b2b_init_request function.
There are multiple examples that use different values in scenario_id
https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2
b2b_init_request("prepaid"); or b2b_init_request("refer"); but the values "prepaid" or
"refer" not used to differentiate anything within route[b2b_logic_request] or anywhere else in documentation.
I get the meaning of req_route parameter of b2b_init_request function to
address different B2b logic using different routes, but what does the
scenario_id used for?
Thanks,
Xaled
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[OpenSIPS-Users] [BLOG] Apache Kafka Producer in OpenSIPS 3.2
Hi everyone, Check out our latest blog [1] that shows how you can publish data/events to Apache Kafka! Besides, the post also touches on the improvements made to the event failover capabilities in OpenSIPS. Enjoy this new integration feature in OpenSIPS 3.2! [1] https://blog.opensips.org/2021/03/04/apache-kafka-producer-in-opensips-3-2/ Cheers, -- Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] [BLOG] Exploring SSL/TLS libraries for OpenSIPS 3.2
Hello everyone, Check out this new blog post [1] that recounts our process of searching for and choosing a SSL/TLS library to use in OpenSIPS 3.2, as an alternative to the problematic OpenSSL. [1] https://blog.opensips.org/2021/02/11/exploring-ssl-tls-libraries-for-opensips-3-2/ Cheers, -- Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] www_challenge qop type failed with AAA
Hi Michael,
Regarding the qop issue, you should pass for that parameter a string
value of "auth", "auth-int", or both separated by ','.
Regards,
--
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OpenSIPS Developer
http://www.opensips-solutions.com
On 05.01.2021 19:50, Michael Vale via Users wrote:
Hi,
I’m attempting to do auth with AAA,
I was met with an error of no Cisco vendor and no SIP-URI-Host Attribute which
I added to the dictionaries.
Now www_challenge thinks either “0” or “1” is an invalid qop type.
ERROR:auth:fixup_qop: Bad qop type
Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:fix_cmd: Fixup
failed for param [2]
Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:fix_actions: Failed to
fix command
Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:fix_actions:
fixing failed (code=-1) at /etc/opensips/opensips.cfg:216
Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:main: failed to
fix configuration with err code -1
Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: INFO:core:cleanup: cleanup
Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: NOTICE:core:main: Exiting
Jan 06 04:42:14 unispy opensips[29709]: INFO:core:daemonize: pre-daemon proce
if (!aaa_www_authorize("")) {
www_challenge("", "1");
};
Could someone please shed some light on the subject?
Also it would be nice if there was another approach to handling the
SIP-URI-Host and Cisco issue or at least some confirmation that my approach was
the best way to deal with it.
Regards,
Michael.
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Re: [OpenSIPS-Users] too many parameters for command
Hi Elaine,
Unfortunately the documentation is actually misleading and the function
accepts only one extra argument for passing to python.
Regards,
--
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 04.12.2020 11:30, Elaine Huang wrote:
Hi,
I wrote a python function that takes 3 parameters (excluding msg):
class MyClass:
…
my_f(self, msg, param1, param2, param3):
…
…
While the python module readme suggests it can accept extra args (more
than 1), opensips fail to start with error: too many parameters for
command
my config code:
python_exec("my_f", param1, param2, param3)
It can start if I change it to:
python_exec("my_f", param1)
Any idea why?
OpenSIPS version: 3.1
Kind Regards,
Elaine
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Re: [OpenSIPS-Users] Clear local cache collection without OpenSIPS 3.1 restart
I did forget to escape the "*" character in the example I gave, so this
should work:
opensips-cli -x mi cache_remove_chunk "*" other
Regards,
--
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OpenSIPS Developer
http://www.opensips-solutions.com
On 20.11.2020 15:25, Mark Allen wrote:
thanks Vlad - I've tried it but I get
ERROR: command 'cache_remove_chunk' returned: -32602: Invalid params
On Fri, 20 Nov 2020 at 13:21, Vlad Patrascu <mailto:vl...@opensips.org>> wrote:
Hi Mark,
The command you should run is:
opensips-cli -x mi cache_remove_chunk * other
I will also update the docs as the example is outdated so thanks
for pointing it out!
Regards,
--
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 20.11.2020 13:47, Mark Allen wrote:
I have local cache collections setup in my script:
loadmodule "cachedb_local.so"
modparam("cachedb_local", "cache_collections", "general; other;")
modparam("cachedb_local", "cachedb_url",
"local:general:///general")
modparam("cachedb_local", "cachedb_url", "local:other:///other")
I want to be able to use opensips-cli to clear the contents of
the "other" collection from the command line or a script without
needing to restart OpenSIPS or losing the contents of the
"general" cache collection. I've looked at the cachedb_local
documentation for exported MI functions but I don't understand
how this translates into a valid opensips-cli command.
Any help gratefully received! :D
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Re: [OpenSIPS-Users] Clear local cache collection without OpenSIPS 3.1 restart
Hi Mark,
The command you should run is:
opensips-cli -x mi cache_remove_chunk * other
I will also update the docs as the example is outdated so thanks for
pointing it out!
Regards,
--
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 20.11.2020 13:47, Mark Allen wrote:
I have local cache collections setup in my script:
loadmodule "cachedb_local.so"
modparam("cachedb_local", "cache_collections", "general; other;")
modparam("cachedb_local", "cachedb_url", "local:general:///general")
modparam("cachedb_local", "cachedb_url", "local:other:///other")
I want to be able to use opensips-cli to clear the contents of the
"other" collection from the command line or a script without needing
to restart OpenSIPS or losing the contents of the "general" cache
collection. I've looked at the cachedb_local documentation for
exported MI functions but I don't understand how this translates into
a valid opensips-cli command.
Any help gratefully received! :D
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Re: [OpenSIPS-Users] Teams TLS Error
I guess this error, which is indeed not worded very well, threw you off:
ERROR:tls_mgm:tlsp_set_certificate: TLS domain [my.domain.com
<http://my.domain.com>] not defined in '[my.domain.com
<http://my.domain.com>]/usr/local/etc/opensips/tls/my_domain_com.pem'
What it actually means is that the TLS domain "my.domain.com" (the
tls_mgm module's record/structure, ) is not defined. Refer to this [1]
section in docs for a better explanation on the the concept of "TLS
domains" in tls_mgm. The term "TLS domain" might be a little loose and
generate confusion sometimes.
So what you should do is add this in your config:
modparam("tls_mgm", "server_domain", "my.domain.com")
By the way, you can use any arbitrary identifier for the domain part in
the tls_mgm parameters, eg. :
modparam("tls_mgm", "server_domain", "mydom")
modparam("tls_mgm","certificate",
"[mydom]/usr/local/etc/opensips/tls/myCert.pem")
...
[1] https://opensips.org/docs/modules/3.1.x/tls_mgm.html#idp2972192
Regards,
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OpenSIPS Developer
http://www.opensips-solutions.com
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Re: [OpenSIPS-Users] dialog module mi dlg_list not working as expected?
Hi Mark, This is indeed a bug in the MI output for dlg_list when passing callid and/or from_tag params. I have pushed a fix, thanks for reporting! Regards, -- Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 16.11.2020 15:05, M S wrote: Hi, I think something is not working as expected with dialog: I check this callid and it exists: [root@sbc2 ~]# opensips-cli -x mi dlg_list | grep 209eaf197691d8a937eb9d514104679c "callid": "209eaf197691d8a937eb9d514104679c@192.168.20.190:5060 <http://209eaf197691d8a937eb9d514104679c@192.168.20.190:5060>", Then I try it like https://opensips.org/html/docs/modules/3.1.x/dialog.html#mi_dlg_list : [root@sbc2 ~]# opensips-cli -x mi dlg_list 209eaf197691d8a937eb9d514104679c@192.168.20.190:5060 <http://209eaf197691d8a937eb9d514104679c@192.168.20.190:5060> ERROR: command 'dlg_list' returned: -32602: Invalid params (Command only supports named parameters) and if I go by my guess: [root@sbc2 ~]# opensips-cli -x mi dlg_list callid=209eaf197691d8a937eb9d514104679c@192.168.20.190:5060 <http://209eaf197691d8a937eb9d514104679c@192.168.20.190:5060> ERROR: command 'dlg_list' returned: -32000: Server error something broken? Mark ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Teams TLS Error
Hi Mark,
Can you post the actual errors that you get in the OpenSIPS logs, if
that is the case?
Regards,
--
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 16.11.2020 11:04, Mark Farmer wrote:
Good morning all
Can anyone clarify whether the TLS domain in SAN is supported or not
please?
Many thanks
Mark.
On Fri, 13 Nov 2020 at 15:59, Kevin Vines <mailto:kevin.vi...@gmail.com>> wrote:
You got me there... the doc states
OpenSIPS offers SIP service for multiple
219 domains, e.g.atlanta.com <http://atlanta.com> andbiloxi.com
<http://biloxi.com>. Altough both domains
220 will be hosted on a single SIP proxy, the SIP proxy needs 2
221 certificates: One foratlanta.com <http://atlanta.com> and one
forbiloxi.com <http://biloxi.com>. For
222 incoming TLS connections
If you need one cert per domain, maybe it implies that you need to
have the domain as the CN instead of a SAN?
Kevin
*From:* farm...@gmail.com <mailto:farm...@gmail.com>
*Sent:* November 13, 2020 10:43 a.m.
*To:* users@lists.opensips.org <mailto:users@lists.opensips.org>
*Reply to:* users@lists.opensips.org <mailto:users@lists.opensips.org>
*Subject:* Re: [OpenSIPS-Users] Teams TLS Error
OK so now I have this:
modparam("tls_mgm","certificate", "[my.domain.name
<http://my.domain.name>]/usr/local/etc/opensips/tls/myCert.pem
<http://myCert.pem>")
modparam("tls_mgm","private_key", "[my.domain.name
<http://my.domain.name>]/usr/local/etc/opensips/tls/myKey.key
<http://myKey.key>")
modparam("tls_mgm","ca_dir", "/etc/ssl/certs")
modparam("tls_mgm","verify_cert", "[my.domain.name
<http://my.domain.name>]1")
modparam("tls_mgm","require_cert", "[my.domain.name
<http://my.domain.name>]1")
modparam("tls_mgm","tls_method", "[my.domain.name
<http://my.domain.name>]TLSv1_2")
modparam("tls_mgm", "match_sip_domain", "my.domain.name
<http://my.domain.name>")
But now it claims that my.domain.name <http://my.domain.name> is
not defined in myCert.pem <http://myCert.pem>
I know it is - it is in a SAN within the certificate.
Any suggestions?
Many thanks
Mark.
On Fri, 13 Nov 2020 at 15:12, Kevin Vines mailto:kevin.vi...@gmail.com>> wrote:
Hi Mark,
Based on some googling it looks like you need to specify the
domain eg:
modparam("tls_mgm","verify_cert", "[domain.com
<http://domain.com>]1")
https://fossies.org/linux/opensips/modules/tls_mgm/README
Kevin
*From:* farm...@gmail.com <mailto:farm...@gmail.com>
*Sent:* November 13, 2020 9:49 a.m.
*To:* users@lists.opensips.org <mailto:users@lists.opensips.org>
*Reply to:* users@lists.opensips.org
<mailto:users@lists.opensips.org>
*Subject:* [OpenSIPS-Users] Teams TLS Error
Hi everyone
OpenSIPS 3.1.0
I am following the OpenSIPS as Teams SBC guide and have added
the TLS config:
modparam("tls_mgm","verify_cert", "1")
modparam("tls_mgm","require_cert", "1")
modparam("tls_mgm","tls_method", "TLSv1_2")
modparam("tls_mgm","certificate",
"/usr/local/etc/opensips/tls/myCert.pem <http://myCert.pem>")
modparam("tls_mgm","private_key",
"/usr/local/etc/opensips/tls/myKey.key <http://myKey.key>")
modparam("tls_mgm", "ca_dir", "/etc/ssl/certs")
But I am seeing a TLS domain error:
Nov 13 14:36:50 [175314] ERROR:tls_mgm:split_param_val: No TLS
domain name
Nov 13 14:36:50 [175314] Traceback (last included file at the
bottom):
Nov 13 14:36:50 [175314] 0.
/usr/local//etc/opensips/opensips.cfg <http://opensips.cfg>
Nov 13 14:36:50 [175314] CRITICAL:core:yyerror: parse error in
/usr/local//etc/opensips/opensips.cfg:191
<http://opensips.cfg:191>:19-20: Parameter not
found in module - can't set
Nov 13 14:36:50 [175314] #modparam("tls_mgm", "require_cert",
"[dom4]1")
Nov 13 14:36:50 [175314]
Nov 13 14:36:50 [175314] modparam("tls_mgm","verify_cert", "1")
Nov 13 14:36:50 [175314] ^~
Nov 13 14:36:50 [175314] modparam("tls_mgm","require_cert"
Re: [OpenSIPS-Users] how to call ul_dump?brief using jsonrpc
Hi Babak,
This is a bit of a mistake in the documentation, as the parameter's name
is "brief", but as value it actually expects an integer. As such the
jsonrpc request should be:
'{"jsonrpc":"2.0","method":"ul_dump","params":{"brief":1},"id":10}'
or:
'{"jsonrpc":"2.0","method":"ul_dump","params":[1],"id":10}'
Regards,
--
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 11.11.2020 15:08, Babak Yakhchali wrote:
Hi
I'm using opensips 3.1.
What json should be sent as body to http mi which results like
ul_dump?brief
I tried this curl command but it is not working
curl --location --request POST 'http://127.0.0.1:8080/opensips_mi' \
--header 'Content-Type: application/json' \
--data-raw
'{"jsonrpc":"2.0","method":"ul_dump","id":10,*"params":["brief"]*}'
I get
{"jsonrpc":"2.0","error":{"code":-32602,"message":"Invalid params"}
this works but it is not brief :)
curl --location --request POST 'http://127.0.0.1:8080/opensips_mi'
--header 'Content-Tya-raw '{"jsonrpc":"2.0","method":"ul_dump","id":10}'
thanks
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Re: [OpenSIPS-Users] Can't register using TLS - version 3.1
Hello, It seems that you haven't properly configured a TLS server domain. What is your tls_mgm module configuration? Regards, -- Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 20.10.2020 11:37, z...@communitake.com wrote: Hi All What ever configurations I’m trying, I always get the following error Oct 20 07:38:07 zivtest opensips: Oct 20 07:38:07 [1390] DBG:proto_tls:tls_conn_init: Creating a whole new ssl connection Oct 20 07:38:07 zivtest opensips: Oct 20 07:38:07 [1390] DBG:proto_tls:tls_conn_init: looking up TLS server domain [172.30.0.8:5061] Oct 20 07:38:07 zivtest opensips: Oct 20 07:38:07 [1390] ERROR:proto_tls:tls_conn_init: no TLS server domain found Oct 20 07:38:07 zivtest opensips: Oct 20 07:38:07 [1390] ERROR:core:handle_io: failed to do proto 3 specific init for conn 0x7ff49831a488 Oct 20 07:38:07 zivtest opensips: Oct 20 07:38:07 [1390] DBG:core:tcpconn_release: releasing con 0x7ff49831a488, state -3, fd=-1, id=181457494 Oct 20 07:38:07 zivtest opensips: Oct 20 07:38:07 [1390] DBG:core:tcpconn_release: extra_data (nil) Oct 20 07:38:07 zivtest opensips: Oct 20 07:38:07 [1399] DBG:core:handle_tcp_worker: response= 7ff49831a488, -3 from tcp worker 0 (0) Oct 20 07:38:07 zivtest opensips: Oct 20 07:38:07 [1399] DBG:core:tcpconn_destroy: destroying connection 0x7ff49831a488, flags 001c ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS 3.1 - raise_event() crashes OpenSIPS with segmentation fault
Hi Mark,
thanks for the report! I have pushed a fix for the crash.
I've also updated the documentation to be more clear about how
raise_event() should be used and the changes from 3.0. Basically, you
don't have to necessarily pass both AVP parameters in 3.1, but if you
want to pass only the attribute values for the event you should do:
raise_event("E_WFC_REGISTERED", , $avp(values));
That is because the event attribute values can now only be passed
through the corresponding parameter(which is the last one). The way you
did it initially was equivalent to passing only the event attribute
names which in 3.1 results in an event with no attributes at all. This
in turn was unfortunately causing a crash.
Regards,
--
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 28.07.2020 17:42, Mark Allen wrote:
[SOLVED]
Hi Stas - good call! It's a change in behaviour from 3.0.
In 3.0 documentation says...
The next two parameters should be AVPs and they are optional. If
only one is present, it should contain the values attached to the event.
In 3.1 it removes mention of the behaviour if only one AVP is present,
but it's not obvious and perhaps it could be documented in the
https://www.opensips.org/Documentation/Migration-3-0-0-to-3-1-0 guide?
Also, perhaps 3.1 should handle it better if parameter is missing
rather than giving a segfault (something for 3.2?)
Once again I'm indebted to you for your help
thanks very much & all the best
Mark
On Tue, 28 Jul 2020 at 15:22, Stas Kobzar <mailto:staskob...@gmail.com>> wrote:
Hi Allen,
Did you try with two parameters: name, value?
*$avp(keys) = "registered";*
$avp(values) = "true";
xlog("Raised E_WFC_REGISTERED $avp(values)");
raise_event("E_WFC_REGISTERED", *$avp(keys)*, $avp(values));
I know they are said to be optional in the documentation but
probably it is optional for two. Either no params or if you pass
parameters, you have to pass both.
On Tue, Jul 28, 2020 at 9:59 AM Mark Allen mailto:m...@allenclan.co.uk>> wrote:
We're upgrading from 3.0 to 3.1. Everything seems ok except we
get a weird error. We subscribe a dynamic event...
startup_route {
subscribe_event("E_WFC_REGISTERED", "udp:127.0.0.1:
<http://127.0.0.1:>");
}
which we can see works from /var/log/syslog...
event_datagram:mod_init: initializing module ...
core:evi_publish_event: Registered event 8b 46 08 48 85 c0 74
0b 48 83 78 18 00 0f 84 a5 02 00 00 e8 34
INFO:core:handle_sigs: child process 10525 exited by a
signal 11
INFO:core:handle_sigs: core was generated
INFO:core:handle_sigs: terminating due to SIGCHLD
If I comment out the raise_event() line - OpenSIPS seems fine
and doesn't crash when passing through this code.
Running gdb to get core file backtrace we see...
Core was generated by `/usr/sbin/opensips -P
/run/opensips/opensips.pid -f /etc/opensips/opensips.cfg'.
Program terminated with signal SIGSEGV, Segmentation fault.
#0 evi_build_payload (params=0x0, method=0x7f931f5b6f08,
id=id@entry=0, extra_param_k=extra_param_k@entry=0x0,
extra_param_v=extra_param_v@entry=0x0) at
evi/evi_transport.c:159
159 if (params->first && !params->first->name.s) {
(gdb) bt full
#0 evi_build_payload (params=0x0, method=0x7f931f5b6f08,
id=id@entry=0, extra_param_k=extra_param_k@entry=0x0,
extra_param_v=extra_param_v@entry=0x0) at
evi/evi_transport.c:159
param =
param_obj = 0x0
tmp =
ret_obj = 0x7f9323135fe0
payload = 0x0
__FUNCTION__ = "evi_build_payload"
#1 0x7f931b7d934f in datagram_raise (msg=,
ev_name=, sock=0x7f931f5c54c8,
params=)
at event_datagram.c:315
ret =
buf =
__FUNCTION__ = "datagram_raise"
#2 0x55cef82148fb in evi_raise_event_msg
(msg=msg@entry=0x7f9323134890, id=id@entry=20,
params=params@entry=0x0)
at evi/event_interface.c:208
subs = 0x7f931f5c55a8
prev =
now = 1595943308
flags = 1073741838
pflags = 0
ret = 0
__FUNCTION__ = "evi_raise_event_msg"
#3 0x55cef8216afb in evi_raise_script_event
(msg=0x7f9323134890, id=20, _a=, _v=)
at evi/event_interface.c:430
vals =
attrs =
v_avp =
Re: [OpenSIPS-Users] How should we represent empty parameters in MI Datagram JSON
Hi Adrian,
In the cases (like 'pua_publish') where there are multiple optional
parameters that cannot be simply identified by position in the array,
the 'params' member in the JSON-RPC request should be an Object with
named parameters.
As such, you should have:
"params": {
"presentity_uri": "sip:system@192.168.2.132",
"expires": 3600,
[...]
}
Regards,
On 06.07.2020 22:47, Adrian Fretwell wrote:
Hello,
What is the correct way to specify empty parameters in an MI call
using JSON?
For example this call to pua_publish has two unused parameters; ETag
and extra_headers, in the old way they were represented by a dot, but
if I try that now I get an invalid etag message:
Pua_publish:
- body type if body of a type different from
default event content-type or .
- ETag that publish should match or . if no
ETag
- extra headers to be added to the request or .
- may not be present in case of update for expire
Example:
:pua_publish:fifo_reply
sip:system@192.168.2.132
3600
presence
application/pidf+xml
.
.
xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model'... etc.
Would someone point me in the right direction please.
Kind regards,
Adrian Fretwell
Sibthorpe
Nottinghamshire
UK.
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Re: [OpenSIPS-Users] cluster/cassandra cache
Hi Slava, Can you open a ticket on Github for this? And also include there the Cassandra version and other relevant information (as per the issue template). Regards, On 09.06.2020 23:57, Slava Bendersky via Users wrote: Hello Everyone, Opensips v3.1 dev can't connect properly to Cassandra cluster. 1591735692.141 [ERROR] (cluster_connector.cpp:190:void datastax::internal::core::ClusterConnector::on_connect(datastax::internal::core::ControlConnector*)): Unable to establish a control connection to host 10.100.101.9 because of the following error: Underlying connection error: Received error response 'Invalid or unsupported protocol version (66); supported versions are (3/v3, 4/v4, 5/v5-beta)' (0x020A) Any help, thank you. volga629 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Stir_shaken signature length
Hi, Please check the fix on the latest master branch on github and let me know if everything is ok. Regards, Vlad Patrascu On 10.04.2020 12:28, Alexandru Tripon wrote: Hi, I tried to populate the Identity header with the stir_shaken module. The header is populated but when I try to verify the signature using an external tool it fails because of the length. I have the folowing Identity generated by Opensips: ` eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiL2hvbWUvdHJpYWwvTHVjcnUvQ29kZS9zdGlyU2hha2VuL215cHVia2V5LnBlbSJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMDAyIl19LCJpYXQiOjE1ODY1MDMxODcsIm9yaWciOnsidG4iOiIxMDAxIn0sIm9yaWdpZCI6IjEyMzQ1NiJ9.MEYCIQCjIx6w8IeilqHq0jbc6uwIB9v1RDmecoep0gRJJC4EmQIhANH1MO9jwRtqH6jgFH12XqROFv-nUroEgzsRAaMJtAsR;info=\u003c/home/trial/Lucru/Code/stirShaken/mypubkey.pem\u003e;ppt=\"shaken\" ` the lenght of encoded signature(in base64) is 96 and in the decoded one is 72. In the RFC for ES256 algorithm(https://tools.ietf.org/html/rfc7518#section-3.4) the length of the decoded signature is 64. Am I missing something here? Thanks, Alexandru Tripon ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Stir_shaken signature length
Hi Frederico, I'm not really sure I understand your question of "how" to generate the signature. Are you refering to how the scripting should look like or something else ? But anyway, it is not possible with OpenSIPS 2.4.7 as the stir_shaken module is available starting with OpenSIPS 3.1. Regards, Vlad Patrascu On 13.04.2020 18:13, Saint Michael wrote: I am trying to do the same. The question I need to ask here is: how do you generate the signature from the certificate, the caller ID and the destination number? I have the API working in staging mode, but now I need to really sign a call and send it forward with Opensips 2.4.7 Federico On Mon, Apr 13, 2020 at 11:03 AM Vlad Patrascu <mailto:vl...@opensips.org>> wrote: Hi Alexandru, OpenSIPS is using the signature in DER encoded format (as it is directly generated by openssl) but indeed it is not the proper format as per RFC 7518. Thanks for the report, I am working on a fix. Regards, Vlad Patrascu On 10.04.2020 12:28, Alexandru Tripon wrote: Hi, I tried to populate the Identity header with the stir_shaken module. The header is populated but when I try to verify the signature using an external tool it fails because of the length. I have the folowing Identity generated by Opensips: ` eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiL2hvbWUvdHJpYWwvTHVjcnUvQ29kZS9zdGlyU2hha2VuL215cHVia2V5LnBlbSJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMDAyIl19LCJpYXQiOjE1ODY1MDMxODcsIm9yaWciOnsidG4iOiIxMDAxIn0sIm9yaWdpZCI6IjEyMzQ1NiJ9.MEYCIQCjIx6w8IeilqHq0jbc6uwIB9v1RDmecoep0gRJJC4EmQIhANH1MO9jwRtqH6jgFH12XqROFv-nUroEgzsRAaMJtAsR;info=\u003c/home/trial/Lucru/Code/stirShaken/mypubkey.pem\u003e;ppt=\"shaken\" ` the lenght of encoded signature(in base64) is 96 and in the decoded one is 72. In the RFC for ES256 algorithm(https://tools.ietf.org/html/rfc7518#section-3.4) the length of the decoded signature is 64. Am I missing something here? Thanks, Alexandru Tripon ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Stir_shaken signature length
Hi Alexandru, OpenSIPS is using the signature in DER encoded format (as it is directly generated by openssl) but indeed it is not the proper format as per RFC 7518. Thanks for the report, I am working on a fix. Regards, Vlad Patrascu On 10.04.2020 12:28, Alexandru Tripon wrote: Hi, I tried to populate the Identity header with the stir_shaken module. The header is populated but when I try to verify the signature using an external tool it fails because of the length. I have the folowing Identity generated by Opensips: ` eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiL2hvbWUvdHJpYWwvTHVjcnUvQ29kZS9zdGlyU2hha2VuL215cHVia2V5LnBlbSJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMDAyIl19LCJpYXQiOjE1ODY1MDMxODcsIm9yaWciOnsidG4iOiIxMDAxIn0sIm9yaWdpZCI6IjEyMzQ1NiJ9.MEYCIQCjIx6w8IeilqHq0jbc6uwIB9v1RDmecoep0gRJJC4EmQIhANH1MO9jwRtqH6jgFH12XqROFv-nUroEgzsRAaMJtAsR;info=\u003c/home/trial/Lucru/Code/stirShaken/mypubkey.pem\u003e;ppt=\"shaken\" ` the lenght of encoded signature(in base64) is 96 and in the decoded one is 72. In the RFC for ES256 algorithm(https://tools.ietf.org/html/rfc7518#section-3.4) the length of the decoded signature is 64. Am I missing something here? Thanks, Alexandru Tripon ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [BLOG] Real-Time Rating and Cost Based Routing in OpenSIPS 3.1
Hello, Currently the module can only be used for a basic postpaid type of rating. Regards, Vlad On 09.04.2020 19:47, Mehdi Shirazi wrote: Hi Thanks for new features. Is it possible(or planed) to use this module for basic prepaid billing ? Regards M.Shirazi >While there are numerous external rating and billing engines available >in the wild, having a quick and easy way of putting a price for a call, >without relying on external applications, can be a valuable asset to have. >https://blog.opensips.org/2020/04/07/real-time-rating-and-cost-based-routing-in-opensips-3-1/ <https://blog.opensips.org/2020/04/07/real-time-rating-and-cost-based-routing-in-opensips-3-1/> >Thank you Vlad Paiu for the valuable contribution and post ! >Best Regards, >Bogdan-Andrei Iancu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] lua_exec in local_route
Hi Johan, There is no workaround unfortunately so indeed you should open a feature request for this. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 2/21/20 12:48 PM, johan wrote: Hello, when I use lua_exec in local_route, I have below error: Feb 21 10:31:26 hendrix opensips[26811]: Not starting opensips: invalid configuration file! Feb 21 10:31:26 hendrix opensips[26811]: Feb 21 10:31:26 [26823] Traceback (last included file at the bottom): Feb 21 10:31:26 hendrix opensips[26811]: Feb 21 10:31:26 [26823] 0. /data/opensips/etc/opensips/opensips.cfg Feb 21 10:31:26 hendrix opensips[26811]: Feb 21 10:31:26 [26823] CRITICAL:core:yyerror: parse error in /data/opensips/etc/opensips/opensips.cfg:1165:23-24: Command cannot be used in the block Context: when I receive an sms in ucs2, proto_smpp returns the message in utf16 big endian. I need to to convert this to utf8. You find the body in local_route as the message is created by tm. Therefore I thought to do the decoding using lua. so I assume that lua_exec is there forbidden. If this is true, should I open a feature request to add this functionality or is there a workaround ? BR, Johan. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips lua
You get the newline from the print() function in the python script that
you call. I've tested with your script and for example this fixed it:
print(testing(),end="")
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 1/14/20 3:32 PM, volga629 via Users wrote:
Hello Vlad,
In lua script we use \n only in xlog.
this repository
https://github.com/VoIP-SAAS/opensips-smpp-lua
On Tue, Jan 14, 2020 at 13:37, Vlad Patrascu wrote:
I still don't think it is AVP_set function's fault for the
whitespaces. Make sure that the string you produce in lua and want to
return to opensips doesn't have a newline character at the end. It
seems that syslog prints a newline as "#012" sometimes (if it's not
actually at the end of a message).
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 1/13/20 3:26 PM, volga629 via Users wrote:
Hello Vlad,
Yes, it still in issue, but we have work around. #012 it insert
white spaces.
$avp(sms-out) = $(avp(formatted-msg){s.trimr});
On Mon, Jan 13, 2020 at 12:26, Vlad Patrascu wrote:
Hi Volga,
Sorry for getting to this so late. Do you still encounter this
issue? I have tried to reproduce this myself but it seems that
AVP_set() does set the value correctly.
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 12/14/19 7:12 AM, volga629 via Users wrote:
Hello Everyone,
Having some issue get lua to set proper avp. When it set in
insert some extra characters into value of avp
Here are log
Dec 14 05:53:41 dev1-fr /usr/sbin/opensips[12985]: siplua:
test::w nah.uy/u5bmnc
Dec 14 05:53:41 dev1-fr /usr/sbin/opensips[12985]: siplua: Tested
string ~>0
Dec 14 05:53:41 dev1-fr /usr/sbin/opensips[12985]: SMS_ROUTE_IN:
Test string ~> [0#012]
It insert #012
Lua script
local cmd_var = handle:read("*all")
handle:close()
xlog("Tested string ~> " .. cmd_var .. "\n")
r eturn AVP_set("test-str", cmd_var)
Any help thank you
volga629
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Re: [OpenSIPS-Users] opensips lua
I still don't think it is AVP_set function's fault for the whitespaces.
Make sure that the string you produce in lua and want to return to
opensips doesn't have a newline character at the end. It seems that
syslog prints a newline as "#012" sometimes (if it's not actually at the
end of a message).
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 1/13/20 3:26 PM, volga629 via Users wrote:
Hello Vlad,
Yes, it still in issue, but we have work around. #012 it insert white
spaces.
$avp(sms-out) = $(avp(formatted-msg){s.trimr});
On Mon, Jan 13, 2020 at 12:26, Vlad Patrascu wrote:
Hi Volga,
Sorry for getting to this so late. Do you still encounter this issue?
I have tried to reproduce this myself but it seems that AVP_set()
does set the value correctly.
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 12/14/19 7:12 AM, volga629 via Users wrote:
Hello Everyone,
Having some issue get lua to set proper avp. When it set in insert
some extra characters into value of avp
Here are log
Dec 14 05:53:41 dev1-fr /usr/sbin/opensips[12985]: siplua:
test::w nah.uy/u5bmnc
Dec 14 05:53:41 dev1-fr /usr/sbin/opensips[12985]: siplua: Tested
string ~>0
Dec 14 05:53:41 dev1-fr /usr/sbin/opensips[12985]: SMS_ROUTE_IN:
Test string ~> [0#012]
It insert #012
Lua script
local cmd_var = handle:read("*all")
handle:close()
xlog("Tested string ~> " .. cmd_var .. "\n")
r eturn AVP_set("test-str", cmd_var)
Any help thank you
volga629
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Re: [OpenSIPS-Users] opensips lua
Hi Volga,
Sorry for getting to this so late. Do you still encounter this issue? I
have tried to reproduce this myself but it seems that AVP_set() does set
the value correctly.
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 12/14/19 7:12 AM, volga629 via Users wrote:
Hello Everyone,
Having some issue get lua to set proper avp. When it set in insert
some extra characters into value of avp
Here are log
Dec 14 05:53:41 dev1-fr /usr/sbin/opensips[12985]: siplua: test::w
nah.uy/u5bmnc
Dec 14 05:53:41 dev1-fr /usr/sbin/opensips[12985]: siplua: Tested
string ~>0
Dec 14 05:53:41 dev1-fr /usr/sbin/opensips[12985]: SMS_ROUTE_IN: Test
string ~> [0#012]
It insert #012
Lua script
local cmd_var = handle:read("*all")
handle:close()
xlog("Tested string ~> " .. cmd_var .. "\n")
r eturn AVP_set("test-str", cmd_var)
Any help thank you
volga629
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Re: [OpenSIPS-Users] Dialog profiles replication issues
Hi Sammy,
I have pushed the fix for the crash, please check the latest commits.
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 1/7/20 10:40 PM, SamyGo wrote:
Hi Vlad,
Update: it wasn't the issue with dialog module or bin_proto. When I
used avp_db_query() inside async() that led to script processing to
halt, I've removed the async() and its working fine now, however the
crash for adding a new profile variable is still there and needs
resolution.
Best Regards,
Sammy
On Tue, Jan 7, 2020 at 11:38 AM SamyGo <mailto:govoi...@gmail.com>> wrote:
Thanks Vald,
The problem I'm facing with profile sharing is strange, when I
make a call the whole flow freeze in that route with dialog
profile operations i.e get/set. I can wait for as many as 15+
seconds and nothing happens. CANCEL the call and resend INVITE and
it completes the route in normal way. Which is beyond my
understanding as why each alternate call is able to process the
route.
Here is the core dump trace: https://pastebin.com/XqiaMWnV
Best regards,
Sammy
On Tue, Jan 7, 2020 at 10:03 AM Vlad Patrascu mailto:vl...@opensips.org>> wrote:
Hi Sammy,
The behavior you described when replicating profiles is not
actually a loop but instead the normal operation. Dialog
profile counters are periodically broadcasted by all nodes in
the cluster to all other nodes. Also, an update is also
broadcasted when a counter decreases. As such there indeed
should not be any direct correlation between the number of SIP
invites and binary packets (containing profile counters) sent.
But if the SIP traffic contains BYEs or if you end dialogs you
might see a surge of binary packets.
As for the crash, can you post a full backtrace from gdb?
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 1/6/20 1:05 PM, Bogdan-Andrei Iancu wrote:
reposting
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Summit, Amsterdam, May 2020
https://www.opensips.org/events/Summit-2020Amsterdam/
OpenSIPS Bootcamp, Miami, March 2020
https://opensips.org/training/OpenSIPS_Bootcamp_2020/
On 1/6/20 4:21 AM, SamyGo wrote:
Hi Bogdan,
Thanks for taking time out, here are the related config params.
listen=bin:OPENSIPS_IP:6677
### dialog module for profiles/limits
loadmodule "dialog.so"
modparam("dialog", "db_mode", 0)
modparam("dialog", "profiles_with_value",
"userConcurrent/b;userInbound/b;userOutbound/b;userInternational/b;userTotalCalls/b;inboundDomain/b;outboundDomain/b;globalDomain/b;prefixx
/b;countryPrefix/b")
modparam("dialog", "profile_replication_cluster", 1)
modparam("dialog", "replicate_profiles_timer", 3000)
modparam("dialog", "replicate_profiles_check", 10)
### Clustering Module
loadmodule "clusterer.so"
modparam("clusterer", "db_mode", 1)
modparam("clusterer", "my_node_id", 3)
modparam("clusterer", "sharing_tag", "gw1/1=active")
modparam("clusterer", "node_timeout", 20)
modparam("clusterer",
"db_url","DB_MODULE://DB_USER:DB_PASS@DB_HOST/DB_NAME")
Routing logic:
https://pastebin.com/6whjkhCs
Related Logs are here.
https://pastebin.com/a9a3iDQh
Wireshark graph showing storm of binary packets when SIP
invite initiates the related route.
*BONUS CRASH: *When in a cluster I add a new
profile_with_value , say someVariable/b and restart the
first box with this variable, the other boxes in cluster
crash by saying something along the lines of "unexpected
someVariable received"
I'm trying to get hold of the core files and as soon as
possible will share that here.
Regards,.
Sammy.
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Re: [OpenSIPS-Users] Dialog profiles replication issues
Hi Sammy,
The behavior you described when replicating profiles is not actually a
loop but instead the normal operation. Dialog profile counters are
periodically broadcasted by all nodes in the cluster to all other nodes.
Also, an update is also broadcasted when a counter decreases. As such
there indeed should not be any direct correlation between the number of
SIP invites and binary packets (containing profile counters) sent. But
if the SIP traffic contains BYEs or if you end dialogs you might see a
surge of binary packets.
As for the crash, can you post a full backtrace from gdb?
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 1/6/20 1:05 PM, Bogdan-Andrei Iancu wrote:
reposting
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Summit, Amsterdam, May 2020
https://www.opensips.org/events/Summit-2020Amsterdam/
OpenSIPS Bootcamp, Miami, March 2020
https://opensips.org/training/OpenSIPS_Bootcamp_2020/
On 1/6/20 4:21 AM, SamyGo wrote:
Hi Bogdan,
Thanks for taking time out, here are the related config params.
listen=bin:OPENSIPS_IP:6677
### dialog module for profiles/limits
loadmodule "dialog.so"
modparam("dialog", "db_mode", 0)
modparam("dialog", "profiles_with_value",
"userConcurrent/b;userInbound/b;userOutbound/b;userInternational/b;userTotalCalls/b;inboundDomain/b;outboundDomain/b;globalDomain/b;prefixx
/b;countryPrefix/b")
modparam("dialog", "profile_replication_cluster", 1)
modparam("dialog", "replicate_profiles_timer", 3000)
modparam("dialog", "replicate_profiles_check", 10)
### Clustering Module
loadmodule "clusterer.so"
modparam("clusterer", "db_mode", 1)
modparam("clusterer", "my_node_id", 3)
modparam("clusterer", "sharing_tag", "gw1/1=active")
modparam("clusterer", "node_timeout", 20)
modparam("clusterer",
"db_url","DB_MODULE://DB_USER:DB_PASS@DB_HOST/DB_NAME")
Routing logic:
https://pastebin.com/6whjkhCs
Related Logs are here.
https://pastebin.com/a9a3iDQh
Wireshark graph showing storm of binary packets when SIP invite
initiates the related route.
*BONUS CRASH: *When in a cluster I add a new profile_with_value , say
someVariable/b and restart the first box with this variable, the
other boxes in cluster crash by saying something along the lines of
"unexpected someVariable received"
I'm trying to get hold of the core files and as soon as possible will
share that here.
Regards,.
Sammy.
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Re: [OpenSIPS-Users] Set AVP variables from Python script ?
Hi Alexandro, Unfortunately, you cannot do this in Python, but as you've noticed you can use Lua or Perl. Starting from OpenSIPS 3.1 though, it will be possible to call OpenSIPS core script functions from Python, so 'pv_printf()' would facilitate writing AVPs from Python. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 11/26/19 11:43 PM, Alexandro Conde wrote: Hi all, Is there a way to set a AVP variable from a Python script ? I need to authenticate the endpoints with the username & password stored and managed on other system DB, I already have an API that can return the user information, so it will be great to create a Python script that consume the API and return the username & password on a AVP variable to do the Authorization using the pv_www_authorize command. I prefer to program that on Python but if it's not possible it can be on LUA or Perl Right? (AVP_set & OpenSIPS::AVP) Thanks in Advance ! -- *Alexandro Conde Mtz.* Voip Specialist & Developer aco...@perfectpitchtech.com <mailto:aco...@perfectpitchtech.com> Facebook <https://www.facebook.com/perfectpitchtech> Twitter <https://twitter.com/PerfectPitchTM> https://perfectpitchtech.com/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 3.x LTS
Hi Mark, Indeed 3.0 is not LTS, the next LTS version will be 3.1. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 11/8/19 12:03 PM, Mark Farmer wrote: According to the versions page only 2.4.6 is currently LTS if I understood correctly :) https://www.opensips.org/About/AvailableVersions On Fri, 8 Nov 2019 at 09:46, Giovanni Maruzzelli <mailto:gmar...@gmail.com>> wrote: 3.0 IS LTS (if I understood correctly) On Fri, Nov 8, 2019 at 10:26 AM Mark Farmer mailto:farm...@gmail.com>> wrote: Good morning I have some upcoming projects for which I will be using OpenSIPS and I would really like to use features from 3.x but I need to stick to LTS releases. Is there any kind of timeline for the next LTS release please? Best regards Mark. ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Mark Farmer farm...@gmail.com <mailto:farm...@gmail.com> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] segfault in dialog.so
Hello,
Can you open a ticket on Github and also extract a full backtrace of the
crash?
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 11/7/19 10:19 AM, Антон Ершов wrote:
Hello friends,
I already wrote about this problem and there was no answer.
http://lists.opensips.org/pipermail/users/2019-October/041771.html
I still want to process call profiles in event_route, but opensips
falls into error at the same time.
opensips -V
version: opensips 3.0.1 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.
git revision: 3d2bd5318
main.c compiled on 02:53:35 Nov 7 2019 with gcc 4.8.5
my event_route
event_route[E_DLG_STATE_CHANGED] {
xlog("L_INFO", "[$param(callid)] - new state dialog changed
$param(new_state) old state $param(old_state) \n");
cache_raw_query("redis:group3", "HGET callid
$param(callid)","$avp(reknum)");
if ($avp(reknum) != NULL) {
xlog("L_INFO", "[$param(callid)] - get reknum $avp(reknum) \n");
if ( get_dialogs_by_profile("reknumber", $avp(reknum),
$avp(dlg_jsons), $avp(callcount)) ) {
xlog("L_INFO", "[$param(callid)] - advertising number $avp(reknum)
has $avp(callcount) other calls \n");
} else {
xlog("L_INFO", "[$param(callid)] - this profile does not have
active dialogs \n");
$avp(callcount) = 0;
}
switch($param(new_state)) {
case 4:
xlog("L_INFO", "[$param(callid)] - write in REDIS \"callcount
$avp(reknum) $avp(callcount) \n");
if ($avp(reknum) != "" && $avp(callcount) != "") {
cache_raw_query("redis:group3","HSET callcount $avp(reknum)
$avp(callcount)");
cache_raw_query("redis:group3","EXPIRE callconut 360");
xlog("L_INFO", "[$param(callid)] - call established. Now on number
$avp(reknum) $avp(callcount) calls \n");
}
break;
case 5:
xlog("L_INFO", "[$param(callid)] - write in REDIS \"callcount
$avp(reknum) $avp(callcount) \n");
if ($avp(reknum) != "" && $avp(callcount) != "") {
cache_raw_query("redis:group3","HSET callcount $avp(reknum)
$avp(callcount)");
cache_raw_query("redis:group3","EXPIRE callcount 360");
xlog("L_INFO", "[$param(callid)] - call end. Now on number
$avp(reknum) $avp(callcount) calls \n");
cache_raw_query("redis:group3","HDEL callid $param(callid)");
}
break;
}
} else {
xlog("L_INFO", "[$param(callid)] - reknum vareable is NULL");
exit;
}
}
when we try to get a profile of an already dead dialog opensips falls.
please tell me is it possible to implement what I want or is it worth
looking for another solution?
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Re: [OpenSIPS-Users] opensipsctl fifo get_statistics - dialogs stats synced?
The term "processed" should be read more generically in this context to include to act of receiving the replicated dialog and preparing it to become active at any time. Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 10/21/2019 02:57 PM, solarmon wrote: Hi Vlad, That was my understanding too, but why would processed_dialogs be incrementing on the standby (State=0) node? On Mon, 21 Oct 2019 at 12:51, Vlad Patrascu <mailto:vl...@opensips.org>> wrote: Hi solarmon, 'failed_dialogs' and 'processed_dialogs' represent the total number of failed/processed dialogs on that OpenSIPS instance since startup. If one of the nodes has been running and handling traffic for a longer time, it's normal for the stats to be different. On the other hand 'active_dialogs' should indeed be always in sync. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 10/21/2019 01:07 PM, solarmon wrote: Hi, Are dialogs stats meant to be totally synced between to openSIPS nodes in a cluster - i.e. when using clusterer? I am monitoring/graphing the active_dialogs, failed_dialogs and processed_dialogs for each node, using 'opensipsctl fifo get_statistics' and they are not fully synced: *Node 1:* active_dialogs = 6 failed_dialogs = 175 processed_dialogs = 605 *Node 2:* active_dialogs = 6 failed_dialogs = 0 processed_dialogs = 430 It seems active_dialogs is synced, but failed_dialogs and processed_dialogs are not. processed_dialogs figures are increasing on both, but they are not synced. Please can somebody explain and/or confirm this behaviour. Thanks ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensipsctl fifo get_statistics - dialogs stats synced?
Hi solarmon, 'failed_dialogs' and 'processed_dialogs' represent the total number of failed/processed dialogs on that OpenSIPS instance since startup. If one of the nodes has been running and handling traffic for a longer time, it's normal for the stats to be different. On the other hand 'active_dialogs' should indeed be always in sync. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 10/21/2019 01:07 PM, solarmon wrote: Hi, Are dialogs stats meant to be totally synced between to openSIPS nodes in a cluster - i.e. when using clusterer? I am monitoring/graphing the active_dialogs, failed_dialogs and processed_dialogs for each node, using 'opensipsctl fifo get_statistics' and they are not fully synced: *Node 1:* active_dialogs = 6 failed_dialogs = 175 processed_dialogs = 605 *Node 2:* active_dialogs = 6 failed_dialogs = 0 processed_dialogs = 430 It seems active_dialogs is synced, but failed_dialogs and processed_dialogs are not. processed_dialogs figures are increasing on both, but they are not synced. Please can somebody explain and/or confirm this behaviour. Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] need some help using opensips 3.0 cluster module .
Hi Sasmita,
The functions that allow you the send custom messages in the cluster at
script level are still there in opensips 3.0, so yes, you can implement
something similar to that blog post.
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 09/06/2019 11:25 AM, Sasmita Panda wrote:
Ok . Thank you .
I got a document for geo distributed clustering which uses redis as
cache . This uses opensips 2.4 . I had tested this architecture also .
This is working fine .
https://saevolgo.blogspot.com/2018/01/opensips-geo-distributed-clustering.html
Can I use this kind of scripting with opensips 3.0 to use redis cache ?
If not , then to use redis I have to use opensips2.4 with clustering .
Am I right ?
*/Thanks & Regards/*
/Sasmita Panda/
/Senior Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Fri, Sep 6, 2019 at 1:31 PM Giovanni Maruzzelli <mailto:gmar...@gmail.com>> wrote:
it cannot work
documentation say that explicitly
"Currently, the metadata information may be published to NoSQL
databases which support key/multi-value column-like associations.
Example known backends to support these abstractions at the time
of writing are MongoDB and Cassandra."
https://opensips.org/html/docs/modules/3.0.x/usrloc.html
On Fri, Sep 6, 2019 at 9:53 AM Sasmita Panda mailto:spa...@3clogic.com>> wrote:
Hi All ,
I have tested opensips 3.0 federated cachedb cluster with
mongo db as cache db .
But in my company we are using redis as cache db.
loadmodule "cachedb_redis.so"
loadmodule "db_cachedb.so"
modparam("cachedb_redis",
"cachedb_url","redis:group1://192.168.0.213:6379/
<http://192.168.0.213:6379/>")
modparam("db_cachedb","cachedb_url","redis:group1://192.168.0.213:6379/
<http://192.168.0.213:6379/>")
modparam("usrloc", "cachedb_url",
"redis:group1://192.168.0.213:6379/ <http://192.168.0.213:6379/>")
In place of mongodb I have replace redis . But its not working
. I am not getting any data the redis cache . Please help me .
*/Thanks & Regards/*
/Sasmita Panda/
/Senior Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Mon, Sep 2, 2019 at 1:07 PM Sasmita Panda
mailto:spa...@3clogic.com>> wrote:
*/Thanks & Regards/*
/Sasmita Panda/
/Senior Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
-- Forwarded message -
From: *Giovanni Maruzzelli* mailto:gmar...@gmail.com>>
Date: Mon, Sep 2, 2019 at 12:40 PM
Subject: Re: [OpenSIPS-Users] need some help using
opensips 3.0 cluster module .
To: OpenSIPS users mailling list mailto:users@lists.opensips.org>>
Sasmita, how possibly you can do that?
How you then associate a machine with an id?
Would you create a new database field for the MAC of the
machine?
-giovanni
On Mon, Sep 2, 2019 at 8:54 AM Sasmita Panda
mailto:spa...@3clogic.com>> wrote:
Hi All ,
In clustering , every opensips node has its own node id .
modparam("clusterer", "my_node_id", 2)
I want to read this value from from the DB . Is this
possible or not ? If yes , how will I do this ?
Please do help me .
*/Thanks & Regards/*
/Sasmita Panda/
/Senior Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Thu, Aug 8, 2019 at 12:44 PM Sasmita Panda
mailto:spa...@3clogic.com>> wrote:
Thank you so much for the explanation .
*/Thanks & Regards/*
/Sasmita Panda/
/Senior Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Wed, Aug 7, 2019 at 6:46 PM Liviu Chircu
mailto:li...@opensips.org>>
wrote:
Sasmita,
Dialog replication is not meant to work on
such a cluster. Since dialogs
retain more
state than registration, they are more
sensitive to the IP of the
current node.
If you want high-availability for dialogs, the
basic unit of deployment
sh
Re: [OpenSIPS-Users] OpenSIPS 2.4 HA Seed Node Dialog Sync
Hi Pasan, Indeed, it can help to have the DB as an alternative for restart persistency in case it can not be achieved through cluster syncing. But this is not _required_ and the concept of "seed" node is not linked to the DB. Rather, this is more of a consequence of the fact that the seed does not automatically perform a sync itself. Nevertheless, it can be forced to sync by running an MI command. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 08/29/2019 01:41 PM, Pasan Meemaduma via Users wrote: Hi Steve, you need a db backend to have restart persistence. From what I understood seed node require a db backend to restore data that require restart persistence. On Wednesday, 28 August 2019, 9:25:27 am SGT, Steve Sharad Kumar wrote: Hi, We have 2 openSIPS instances in HA and one instance is seed node with seed flag in the DB. So each time backup node reboots and it always syncs the dialog from seed node. But how we can make a seed node to sync the dialog from other backup node is seed node reboots? I tried to make both nodes seed but primary node (seed) never syncs the dialog. So i execute the dialog sync command using keepalived everytime seed node reboots. Is there any other way to make seed node sync too please. Any help will be appreciated. Steve Sharad Kumar ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS 2.4 HA Seed Node Dialog Sync
Hi,
If you are using a 2.4 build newer than 8 July, you can set the the
following modparam to configure the seed node to also try to find a sync
donor within a given interval:
...
modparam("clusterer", "seed_fallback_interval", 5)
...
Otherwise, this improvement is not included in the latest 2.4.6 release,
and the only option is triggering the sync MI command.
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 08/27/2019 11:41 PM, Steve Sharad Kumar wrote:
Hi,
We have 2 openSIPS instances in HA and one instance is seed node with
seed flag in the DB. So each time backup node reboots and it always
syncs the dialog from seed node. But how we can make a seed node to
sync the dialog from other backup node is seed node reboots?
I tried to make both nodes seed but primary node (seed) never syncs
the dialog. So i execute the dialog sync command using keepalived
everytime seed node reboots. Is there any other way to make seed node
sync too please.
Any help will be appreciated.
Steve Sharad Kumar
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Re: [OpenSIPS-Users] Function add_isup_part() parsing error in opensips 3.0
It's on 3.1 and 3.0.
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 07/18/2019 09:52 AM, sagar malam wrote:
Hi Vlad,
Thanks for fix.Is it pushed to 3.0 ? or it will be released with 3.1 ?
On Wed, Jul 17, 2019 at 8:41 PM Vlad Patrascu <mailto:vl...@opensips.org>> wrote:
Hi Sagar,
This is a bug introduced when porting the script functions
interface to 3.0. I've just pushed a fix, thanks for reporting!
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 07/17/2019 05:05 PM, sagar malam wrote:
Hello,
Can anyone please help me on this ?
On Tue, Jul 16, 2019 at 3:22 PM sagar malam mailto:sagarma...@gmail.com>> wrote:
Hello,
I am using *add_isup_part([isup_msg_type][,extra_headers])*
function from request and failure route to local route in
*Opensips 3.0* version script but I am getting parsing error
as below . I AM NOT GETTING THIS ERROR IN OPENSIPS 2.x.
Please help me on this issue.
Usage :
if(!isflagset(11) && $avp(tsi_types)=="SIP-I" ) {
$ru = $ru + ";user=phone";
xlog("L_INFO","Adding isup part ");
if(add_isup_part("Initial address")) {
setflag(11);
}
} else {
xlog("L_INFO","Removing isup part ");
remove_body_part("application/sdp","revert");
}
--
Error :
Jul 15 10:46:35 sbcsbc opensips: CRITICAL:core:yyerror: parse
error in /usr/local//etc/opensips/opensips.cfg:1282:28-29:
*Command cannot be used in the block#012*
Jul 15 10:46:35 sbcsbc opensips: xlog("L_INFO","Adding
isup part ");
Jul 15 10:46:35 sbcsbc opensips:
Jul 15 10:46:35 sbcsbc opensips:
#011#011if(add_isup_part("Initial address")) {
Jul 15 10:46:35 sbcsbc opensips: #011#011^~
Jul 15 10:46:35 sbcsbc opensips: #011 #011setflag(11);
Jul 15 10:46:35 sbcsbc opensips: #011#011}
Jul 15 10:46:35 sbcsbc opensips:
ERROR:core:parse_opensips_cfg: bad config file (1 errors)
Jul 15 10:46:35 sbcsbc opensips: ERROR:core:main: failed to
parse config file (null)
Jul 15 10:46:35 sbcsbc opensips: NOTICE:core:main: Exiting
--
--
Thanks,
Sagar
--
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Sagar
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Re: [OpenSIPS-Users] Function add_isup_part() parsing error in opensips 3.0
Hi Sagar,
This is a bug introduced when porting the script functions interface to
3.0. I've just pushed a fix, thanks for reporting!
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 07/17/2019 05:05 PM, sagar malam wrote:
Hello,
Can anyone please help me on this ?
On Tue, Jul 16, 2019 at 3:22 PM sagar malam <mailto:sagarma...@gmail.com>> wrote:
Hello,
I am using *add_isup_part([isup_msg_type][,extra_headers])*
function from request and failure route to local route in
*Opensips 3.0* version script but I am getting parsing error as
below . I AM NOT GETTING THIS ERROR IN OPENSIPS 2.x. Please help
me on this issue.
Usage :
if(!isflagset(11) && $avp(tsi_types)=="SIP-I" ) {
$ru = $ru + ";user=phone";
xlog("L_INFO","Adding isup part ");
if(add_isup_part("Initial address")) {
setflag(11);
}
} else {
xlog("L_INFO","Removing isup part ");
remove_body_part("application/sdp","revert");
}
--
Error :
Jul 15 10:46:35 sbcsbc opensips: CRITICAL:core:yyerror: parse
error in /usr/local//etc/opensips/opensips.cfg:1282:28-29:
*Command cannot be used in the block#012*
Jul 15 10:46:35 sbcsbc opensips: xlog("L_INFO","Adding isup part ");
Jul 15 10:46:35 sbcsbc opensips:
Jul 15 10:46:35 sbcsbc opensips: #011#011if(add_isup_part("Initial
address")) {
Jul 15 10:46:35 sbcsbc opensips: #011#011^~
Jul 15 10:46:35 sbcsbc opensips: #011 #011setflag(11);
Jul 15 10:46:35 sbcsbc opensips: #011#011}
Jul 15 10:46:35 sbcsbc opensips: ERROR:core:parse_opensips_cfg:
bad config file (1 errors)
Jul 15 10:46:35 sbcsbc opensips: ERROR:core:main: failed to parse
config file (null)
Jul 15 10:46:35 sbcsbc opensips: NOTICE:core:main: Exiting
--
--
Thanks,
Sagar
--
Thanks,
Sagar
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Re: [OpenSIPS-Users] need some help using opensips 3.0 cluster module .
Hi,
I would advise to directly rely on the usrloc module for configuring
distributed user location, instead of manually scripting this using the
clusterer script functions.
You can read more about this here:
- usrloc module documentation [1]
- federated user location tutorial [2] - this is for 2.4 but aside from
syntax changes, the config should be quite similar.
[1]
https://opensips.org/html/docs/modules/3.0.x/usrloc.html#distributed-sip-user-location
[2]
https://opensips.org/Documentation/Tutorials-Distributed-User-Location-Federation
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 07/09/2019 02:08 PM, Sasmita Panda wrote:
Hi ,
I have followed the below link while configuring opensips 2.4
https://saevolgo.blogspot.com/2018/01/opensips-geo-distributed-clustering.html
Is there any document which I can follow for opensips 3.0 also .
Please let me know or provide any sample config file if possible .
*/Thanks & Regards/*
/Sasmita Panda/
/Senior Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Tue, Jul 9, 2019 at 4:27 PM Sasmita Panda <mailto:spa...@3clogic.com>> wrote:
Hi Vlad ,
I got that and modified . Thanks for your reply .
But I am facing some other problem now . Wherever I have written
xlog to print some user defined logs these logs are not getting
printed . I think I am doing some other mistakes in the config
also .
Can I get a sample config with cluster module enable for opensips
3.0 ? Earlier I have tested with 2.4 and it was working fine . But
lots of things changed in 3.0 and so I am stuck .
*/Thanks & Regards/*
/Sasmita Panda/
/Senior Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Tue, Jul 9, 2019 at 4:05 PM Vlad Patrascu mailto:vl...@opensips.org>> wrote:
Hi Sasmita,
The actual name of the parameter, as stated in the docs for
E_CLUSTERER_REQ_RECEIVED, is "msg". So you should access it by
/$param(msg)/.
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 07/08/2019 02:29 PM, Sasmita Panda wrote:
event_route[E_CLUSTERER_REQ_RECEIVED]{
$avp(state) = $(param(rcv_msg){s.select,0,:});
$avp(user) = $(param(rcv_msg){s.select,1,:});
$avp(proxy) = $(param(rcv_msg){s.select,2,:});
xlog("L_INFO","[CLUSTER-TALK] {MSG:$avp(rcv_msg)} Sender
Proxy:$avp(proxy) SRC_ID:$avp(source_id) User: $avp(user)
State: $avp(state) ClusterId: $param(cluster_id)\n")
if($avp(state) =~ "REGOFFLINE") {
cache_raw_query("redis:group1","HDEL REGONLINE $avp(user)");
}else if($avp(state) =~ "REGONLINE") {
cache_raw_query("redis:group1","HSET $avp(state) $avp(user)
$avp(proxy)");
}else if($avp(state) =~ "WHEREIS") {
route(QUERY_LOCAL_CACHE);
}else if($avp(state) =~ "HACKER"){
$avp(useragent) = $avp(user);
$avp(hacker_ip) = $avp(proxy);
xlog("L_INFO","[CLUSTER-TALK] HACKER IP
DETECTED BY CLUSTER ADDING $avp(hacker_ip) & User-Agent:
$(avp(useragent){s.b64decode}) Into Watch-List\n");
cache_raw_query("redis:group1","HSET HACKER_USERAGENT_LIST
$avp(useragent) $avp(hacker_ip)");
cache_raw_query("redis:group1","HSET HACKER_IP_LIST
$avp(hacker_ip) $avp(useragent)");
}
exit;
}
This is giving warning in opensips 3.0 as below .
*WARNING:event_route:event_route_param_get: Parameter
not found for event *
**
* Due to this the data is not getting saved in
the subsequent node in the cluster . How I will fix this ? *
*
*
*/Thanks & Regards/*
/Sasmita Panda/
/Senior Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Thu, Jul 4, 2019 at 5:35 PM Sasmita Panda
mailto:spa...@3clogic.com>> wrote:
Thank you for your quick response . I resolved it .
*/Thanks & Regards/*
/Sasmita Panda/
/Senior Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Thu, Jul 4, 2019 at 5:12 PM Liviu Chircu
mailto:li...@opensips.org>> wrote:
Hey Sasmita,
I had a similar problem yesterday. In 3.0,
fetch_event_params() was removed, and you are
simply supposed to access your params via:
$param(cluster_id), $param(src_id), $param(
Re: [OpenSIPS-Users] need some help using opensips 3.0 cluster module .
Hi Sasmita,
The actual name of the parameter, as stated in the docs for
E_CLUSTERER_REQ_RECEIVED, is "msg". So you should access it by
/$param(msg)/.
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 07/08/2019 02:29 PM, Sasmita Panda wrote:
event_route[E_CLUSTERER_REQ_RECEIVED]{
$avp(state) = $(param(rcv_msg){s.select,0,:});
$avp(user) = $(param(rcv_msg){s.select,1,:});
$avp(proxy) = $(param(rcv_msg){s.select,2,:});
xlog("L_INFO","[CLUSTER-TALK] {MSG:$avp(rcv_msg)} Sender
Proxy:$avp(proxy) SRC_ID:$avp(source_id) User: $avp(user) State:
$avp(state) ClusterId: $param(cluster_id)\n")
if($avp(state) =~ "REGOFFLINE") {
cache_raw_query("redis:group1","HDEL REGONLINE
$avp(user)");
}else if($avp(state) =~ "REGONLINE") {
cache_raw_query("redis:group1","HSET $avp(state)
$avp(user) $avp(proxy)");
}else if($avp(state) =~ "WHEREIS") {
route(QUERY_LOCAL_CACHE);
}else if($avp(state) =~ "HACKER"){
$avp(useragent) = $avp(user);
$avp(hacker_ip) = $avp(proxy);
xlog("L_INFO","[CLUSTER-TALK] HACKER IP DETECTED BY
CLUSTER ADDING $avp(hacker_ip) & User-Agent:
$(avp(useragent){s.b64decode}) Into Watch-List\n");
cache_raw_query("redis:group1","HSET
HACKER_USERAGENT_LIST $avp(useragent) $avp(hacker_ip)");
cache_raw_query("redis:group1","HSET HACKER_IP_LIST
$avp(hacker_ip) $avp(useragent)");
}
exit;
}
This is giving warning in opensips 3.0 as below .
*WARNING:event_route:event_route_param_get: Parameter not
found for event *
**
* Due to this the data is not getting saved in the
subsequent node in the cluster . How I will fix this ? *
*
*
*/Thanks & Regards/*
/Sasmita Panda/
/Senior Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Thu, Jul 4, 2019 at 5:35 PM Sasmita Panda <mailto:spa...@3clogic.com>> wrote:
Thank you for your quick response . I resolved it .
*/Thanks & Regards/*
/Sasmita Panda/
/Senior Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Thu, Jul 4, 2019 at 5:12 PM Liviu Chircu mailto:li...@opensips.org>> wrote:
Hey Sasmita,
I had a similar problem yesterday. In 3.0,
fetch_event_params() was removed, and you are
simply supposed to access your params via:
$param(cluster_id), $param(src_id), $param(rcv_msg) and
$param(rcv_tag)
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 04.07.2019 14:22, Sasmita Panda wrote:
Hi All ,
Previously I have used cluster module with opensips 2.4 and
that was working fine . Now I want to upgrade to opensips 3.0
. And there are lots of parameter get changed in opensips 3.0 .
Is there any documentation for cluster module in 3.0 . I need
an example for config file which I can use with opensips 3.0 .
Now I am facing problem in bellow lines .
event_route[E_CLUSTERER_RPL_RECEIVED]{
fetch_event_params("$avp(cluster_id);$avp(src_id);$avp(rcv_msg);$avp(rcv_tag)");
$avp(state) = $(avp(rcv_msg){s.select,0,:});
$avp(user) = $(avp(rcv_msg){s.select,1,:});
$avp(proxy) = $(avp(rcv_msg){s.select,2,:});
xlog("L_INFO","[CLUSTER-TALK] Replying
Proxy:$avp(proxy) User: $avp(user) State: $avp(state)
ClusterId: $avp(cluster_id) TAG: $avp(rcv_tag) \n");
}
fetch_event_params this is not present in opensips 3.0 . How
I will convert this so that it will support in opensips 3.0.
Please do help me .
*/Thanks & Regards/*
/Sasmita Panda/
/Senior Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
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Re: [OpenSIPS-Users] How to add ISUP part to 403 via send_reply function?
Hi Oleg,
Unfortunately, at the moment there is no way to do this, as the
/add_isup_part()/ function appends an ISUP body to the SIP message
exposed in the current script route. So the only option is to add the
ISUP body to a received reply, not a locally generated one.
Also, it's not an easy/quick feature to implement, but if you don't find
a way around it and think it would be useful, please open a feature
request on Github.
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 06/12/2019 07:42 PM, Олег Подгуйко via Users wrote:
My opensips works as state-full-proxy. At the one side it works via
SIP-I, at another side via SIP. Everything is ok.
In one of my business-cases I get some INVITE with ISUP body with
certain parameters from SIP-I side I need to send 403 with ISUP body
including ISUP reason.
To do this I try to use function: send_reply("403","Some reason").
It looks like
# Encapsulate a REL
add_isup_part("REL");
$isup_param(Cause Indicators | Cause value) = 21;
send_reply("403","The reason is 21");
xlog("L_WARN","The call was rejected with 21");
exit;
But actually 403 is sending without any ISUP body.
Is it possible to do such action?
--
Oleg Podguyko
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Re: [OpenSIPS-Users] dialog-dlg-repl capability State=not synced in a 2-node cluster, but dialog-prof-repl State=Ok
Hi Hari, I notice that you use 'db_mode' 0 and set the 'current_info'/'neighbor_info' modparams. In this case, there is no need to provision anything else in the DB, the nodes will automatically discover the topology. Usually this is useful when you want nodes dynamically joining the cluster. But the problem you're facing is probably caused by not setting any node in the cluster with the "seed" flag. You can read more about this here [1]. If everything is configured correctly, setting 'dialog_replication_cluster' should be enough in order to replicate dialogs. [1] https://opensips.org/html/docs/modules/3.0.x/clusterer.html#capabilities Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] cluster dialog synchronization - 200Ok/180Ringing ?
Yes it is . But even if the dialog specific data would have been replicated sooner, transactions are not replicated. As such, when the backup node takes over it cannot make sense of the 200 OK anyway. Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 05/07/2019 05:06 PM, Konrad Malewski wrote: so if there is like 60 sec after 180ringing and 200ok - during that time this dialog is vulnerable to failover ? On Tue, May 7, 2019 at 3:59 PM Vlad Patrascu <mailto:vl...@opensips.org>> wrote: Hi Konrad, The dialog is replicated after the 200 OK is received. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com <https://urldefense.proofpoint.com/v2/url?u=http-3A__www.opensips-2Dsolutions.com&d=DwMDaQ&c=q3cDpHe1hF8lXU5EFjNM_A&r=Q2-P3NgorS0ZH4_aMeg5NL1og3Rsx6yNgM-aaPfa8xzzIkVlEcHJTs2jlyabbp9e&m=Fj_nyRYH9UHn3Dx85pRDIF29KpGk8e8xgDGDS9NQfe0&s=bj9ef4taHUsG059fUfifgY1l66gNZXwXrF5Oj5x83-Y&e=> On 05/07/2019 04:43 PM, Konrad Malewski wrote: Hello, I have a question regarding dialog synchronization in opensips cluster. Does dialog synchronization happen after initial transaction is complete or before ? (When backup is able to take over failed active node?) I am asking because it may be a lot of time after 180Ringing before 200Ok arrives. thanks, -- Konrad Malewski ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <https://urldefense.proofpoint.com/v2/url?u=http-3A__lists.opensips.org_cgi-2Dbin_mailman_listinfo_users&d=DwMDaQ&c=q3cDpHe1hF8lXU5EFjNM_A&r=Q2-P3NgorS0ZH4_aMeg5NL1og3Rsx6yNgM-aaPfa8xzzIkVlEcHJTs2jlyabbp9e&m=Fj_nyRYH9UHn3Dx85pRDIF29KpGk8e8xgDGDS9NQfe0&s=zNjisP_DWAX5QHdaLvr5SOM8oZ1MfdElR_5vfGOJ4ls&e=> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> https://urldefense.proofpoint.com/v2/url?u=http-3A__lists.opensips.org_cgi-2Dbin_mailman_listinfo_users&d=DwIGaQ&c=q3cDpHe1hF8lXU5EFjNM_A&r=Q2-P3NgorS0ZH4_aMeg5NL1og3Rsx6yNgM-aaPfa8xzzIkVlEcHJTs2jlyabbp9e&m=Fj_nyRYH9UHn3Dx85pRDIF29KpGk8e8xgDGDS9NQfe0&s=zNjisP_DWAX5QHdaLvr5SOM8oZ1MfdElR_5vfGOJ4ls&e= -- -- Konrad Malewski ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] cluster dialog synchronization - 200Ok/180Ringing ?
Hi Konrad, The dialog is replicated after the 200 OK is received. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 05/07/2019 04:43 PM, Konrad Malewski wrote: Hello, I have a question regarding dialog synchronization in opensips cluster. Does dialog synchronization happen after initial transaction is complete or before ? (When backup is able to take over failed active node?) I am asking because it may be a lot of time after 180Ringing before 200Ok arrives. thanks, -- Konrad Malewski ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] b2b top hiding with opensips 3.0.0-beta gives "Wrong Scenary ID"
Hi,
Thanks for the report, this was a bug and I've pushed the fix for it, it
should be fine now.
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 04/23/2019 01:54 PM, Konrad Malewski wrote:
Hello,
I switched to opensips3 and b2b internal scenario stopped working:
Apr 23 10:33:29 [55] ERROR:b2b_logic:fixup_b2b_logic: Wrong Scenary
ID. No scenario with this ID [�Ͷ�s]
Apr 23 10:33:29 [55] ERROR:core:fix_cmd: Fixup failed for param [1]
Apr 23 10:33:29 [55] ERROR:core:fix_actions: Failed to fix command
Apr 23 10:33:29 [55] ERROR:core:fix_actions: fixing failed (code=-1)
at /etc/opensips/opensips.cfg:185
Apr 23 10:33:29 [55] ERROR:core:main: failed to fix configuration with
err code -1
Line 185 of configuration contains only
-> b2b_init_request("top hiding");
did scenario name changed ?
--
--
Konrad Malewski
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Re: [OpenSIPS-Users] how to migrate statistics to 3.x ?
The opensipsctl tool does not work anymore with the get_statistics
command and will be dropped soon anyway. Please try the new opensips-cli
tool.
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 04/11/2019 12:54 PM, Konrad Malewski wrote:
Hello,
I am using 3.0.0-dev version and I am having problem with opensipsctl
command for statistics:
$opensipsctl fifo get_statistics load-all
{
"jsonrpc": "2.0",
"id": "a7cbfd4e",
"error": {
"message": "Invalid params",
"code": -32602
}
}
$opensipsctl fifo get_statistics load:
{
"jsonrpc": "2.0",
"id": "177e12c6",
"error": {
"message": "Invalid params",
"code": -32602
}
}
the log says: ERROR:core:get_mi_array_param: Parameter: statistics is
not an array.
What is the new syntax ?
--
--
Konrad Malewski
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Re: [OpenSIPS-Users] sip-i ?
Hi Sean, You can set a custom value for any ISUP parameter that appears in the specification. In your case, you can do: $isup_param(User Service Information) = "0x8090A2" Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 03/26/2019 04:22 PM, Sean Watkins wrote: Hi Bogan I got some more success yesterday with it -- I'll post my configs after, for a SIP-I to SIP gateway. Where I'm stuck now - there seems to be some differences between ANSI + ITU. I've got the invite coming into Opensips, then I'm successfully adding the SDP info for ISUP. (Whomever came up with this... crazy!) The far end carrier always drops the calls 483 Incomplete address. Working with the vendor the carrier, they suspect its in the User service information - They want something like this: User Service Information Mandatory Parameter: User service information (29) Pointer to Parameter: 3 Parameter Length: 3 User service information (-> Q.931 Bearer_capability): 8090a2 1... = Extension indicator: last octet .00. = Coding standard: ITU-T standardized coding (0x0) ...0 = Information transfer capability: Speech (0x00) 1... = Extension indicator: last octet .00. = Transfer mode: Circuit mode (0x0) ...1 = Information transfer rate: 64 kbit/s (0x10) 1... = Extension indicator: last octet .01. = Layer identification: Layer 1 identifier (0x1) ...0 0010 = User information layer 1 protocol: Recommendation G.711 u-law (0x02) Is there any way to customize this in OpenSIPS? Sean Sean On Tue, Mar 26, 2019 at 1:28 AM Bogdan-Andrei Iancu wrote: Hi Sean, Where have you got stuck ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 03/22/2019 10:47 PM, Sean Watkins wrote: Hi Are there any SIP-I experts on here that are willing todo some consulting? I need to get a SIP-I to SIP gateway working... I've got partial call connectivity working, but I don't seem to know what I'm doing.. Don't get all the ins and outs of reply etc.. Sean ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- -- Sean Watkins 403-629-6152 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] clusterer.so settings in 2.4.x
Yes, that is correct.
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 03/15/2019 12:32 PM, Alexey Kazantsev via Users wrote:
Hi Vlad, thank you for the reply.
So I have to configure it in this way:
# main node, which is active if everything's OK
modparam("clusterer",
"current_info","cluster_id=1,url=bin:10.45.144.77:,flags=seed")
modparam("clusterer",
"neighbor_info","cluster_id=1,node_id=2,url=bin:10.45.144.79:")
# backup node, which is standby and will work in case of node1 failure
modparam("clusterer",
"current_info","cluster_id=1,url=bin:10.45.144.79:")
modparam("clusterer",
"neighbor_info","cluster_id=1,node_id=1,url=bin:10.45.144.77:")
Is it correct?
So, node with BIN IP 10.45.144.77 will be a seed, a source of ready cluster
info/data.
---
BR, Alexey
http://alexeyka.zantsev.com/
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Re: [OpenSIPS-Users] clusterer.so settings in 2.4.x
Hi Alexey,
For your clustering scenario you should use the seed flag only in one of
the nodes' configs.
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 03/15/2019 11:13 AM, Alexey Kazantsev via Users wrote:
Hi list!
I have a working active-backup cluster of two OpenSIPS 2.4.5 nodes.
(listening on virtual IP being moved by Keepalived,
and Redis instances for RTPEngines).
I configured clusterer module without DB support.
I'm a little bit confused if I have to use 'flags=seed' only in one
config or in both nodes's configs?
As for now, it is so:
# OpenSIPS1
modparam("clusterer",
"current_info","cluster_id=1,url=bin:10.145.213.63:,flags=seed")
modparam("clusterer",
"neighbor_info","cluster_id=1,node_id=2,url=bin:10.145.213.155:")
# OpenSIPS2
modparam("clusterer", "current_info",
"cluster_id=1,url=bin:10.145.213.155:,flags=seed")
modparam("clusterer", "neighbor_info",
"cluster_id=1,node_id=1,url=bin:10.145.213.63:")
https://blog.opensips.org/2018/03/23/clustering-ongoing-calls-with-opensips-2-4/
https://opensips.org/html/docs/modules/2.4.x/clusterer.html
---
BR, Alexey
http://alexeyka.zantsev.com/
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Re: [OpenSIPS-Users] opensips 2.4.4: bug in tls_mgm
Hi Johan,
The issue is probably caused by the fact that OpenSIPS tries to
initialize a 'default' client and server domain and the certificate file
is inexistent at the default path
('/usr/local/opensips//etc/opensips/tls/cert.pem'). So even if you
define your "sv_dom" custom server domain, OpenSIPS still tries to
create default domains for fallback purposes. When you omit the
'[sv_dom]' prefix for the domain settings in your second test, you
overwrite the default ones (with a valid path for the certificate this
time).
In conclusion, either specify your custom domain _and_ default domains
separately (set the modparams multiple times) or make sure the
certificate is found at the default path. Also, note that it's not
necessary to define a custom server domain if you only intend to use a
single one, as the default will match any socket.
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 02/17/2019 01:35 PM, johan de clercq wrote:
Hi,
I believe that I have found a bug in tls_mgm:
Using opensips default certificates:
/usr/local/opensips/etc/opensips/tls# ls -lu
total 24
-rw-r--r-- 1 root staff 2049 Feb 17 12:13 ca.conf
-rw-r--r-- 1 root staff 1048 Feb 17 12:13 README
-rw-r--r-- 1 root staff 1127 Feb 17 12:13 request.conf
drwxr-sr-x 4 root staff 4096 Feb 17 12:16 rootCA
drwxr-sr-x 2 root staff 4096 Feb 17 12:13 user
-rw-r--r-- 1 root staff 591 Feb 17 12:13 user.conf
/usr/local/opensips/etc/opensips/tls/rootCA# ls
cacert.pem certs index.txt private serial
Tls params:
loadmodule "tls_mgm.so"
modparam("tls_mgm", "server_domain", "sv_dom=5.135.140.139:5061")
modparam("tls_mgm", "require_cert", "[sv_dom]0")
modparam("tls_mgm", "verify_cert", "[sv_dom]0")
modparam("tls_mgm", "tls_method", "[sv_dom]SSLv23")
modparam("tls_mgm", "certificate",
"[sv_dom]/usr/local/opensips/etc/opensips/tls/rootCA/cacert.pem")
modparam("tls_mgm", "private_key",
"[sv_dom]/usr/local/opensips/etc/opensips/tls/rootCA/private/cakey.pem")
modparam("tls_mgm", "ca_list",
"[sv_dom]/usr/local/opensips/etc/opensips/tls/rootCA/cacert.pem")
PROTO_TLS module
loadmodule "proto_tls.so"
modparam("proto_tls", "trace_destination", "hep_dest")
modparam("proto_tls", "trace_on", 1)
I removed the passphrase:
mv etc/tls/rootCA/private/cakey.pem
etc/tls/rootCA/private/cakey.pem.protected
openssl rsa -in etc/tls/rootCA/private/cakey.pem.protected -out
etc/tls/rootCA/private/cakey.pem
and then tried to run opensips from cmdline : ./opensips -f
/usr/local/opensips/etc/opensips/opensips.cfg
syslog output:
Feb 17 12:22:01 ns3012072 ./opensips[28673]:
ERROR:tls_mgm:load_certificate: unable to load certificate file
'/usr/local/opensips//etc/opensips/tls/cert.pem'
Feb 17 12:22:01 ns3012072 ./opensips[28673]:
ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'default'
Feb 17 12:22:01 ns3012072 ./opensips[28673]: ERROR:core:init_mod:
failed to initialize module tls_mgm
Feb 17 12:22:01 ns3012072 ./opensips[28673]: ERROR:core:main: error
while initializing modules
Feb 17 12:22:01 ns3012072 ./opensips[28673]: CRITICAL:core:sig_usr:
segfault in attendant (starter) process!
Feb 17 12:22:01 ns3012072 kernel: [ 4024.678398] opensips[28673]:
segfault at 7fcb76dbf850 ip 7fcb76546f69 sp 7ffe803ac150 error
4 in libcrypto.so.1.1[7fcb763df000+265000]
Next I tried with:
loadmodule "tls_mgm.so"
#modparam("tls_mgm", "server_domain", "sv_dom=5.135.140.139:5061")
modparam("tls_mgm", "require_cert", "0")
modparam("tls_mgm", "verify_cert", "0")
modparam("tls_mgm", "tls_method", "SSLv23")
modparam("tls_mgm", "certificate",
"/usr/local/opensips/etc/opensips/tls/rootCA/cacert.pem")
modparam("tls_mgm", "private_key",
"/usr/local/opensips/etc/opensips/tls/rootCA/private/cakey.pem")
modparam("tls_mgm", "ca_list",
"/usr/local/opensips/etc/opensips/tls/rootCA/cacert.pem")
PROTO_TLS module
loadmodule "proto_tls.so"
modparam("proto_tls", "trace_destination", "hep_dest")
modparam("proto_tls", "trace_on", 1)
and then opensips starts…
Can you please explain what I am doing wrong ?
cid:F3100D46-F00D-4610-87ED-3E91DA790A82
Johan De Clercq, Managing Director
Democon bvba - Ooigemstraat 41 - 8780 Oostrozebeke
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Re: [OpenSIPS-Users] Query regarding opensips cluster module .
Hi, You should issue the 'clusterer_reload' MI command on all running nodes after adding a new one in the DB and no restart is necessary. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 12/28/2018 09:56 AM, Sasmita Panda wrote: version: opensips 2.2.4 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svn revision: 3247:3606M main.c compiled on 09:50:32 Nov 12 2018 with gcc 4.8.3 This is the entire version . Can anyone let me know , whether I am in the latest version or I need to update my code ?? */Thanks & Regards/* /Sasmita Panda/ /Senior Network Testing and Software Engineer/ /3CLogic , ph:07827611765/ On Thu, Dec 27, 2018 at 6:33 PM Sasmita Panda <mailto:spa...@3clogic.com>> wrote: Hi All, I am using below version of opensips . version: opensips 2.2.4 I have configured cluster module and I have 2 node in a single cluster . Lets say I am adding another node into the same cluster . Each time I have to update the config file with the new node information or add the new entry in the DB and restart the process . I don't want to disturb the running setup while adding new node . Is there a way through which I can do it in runtime . for example : when I am updating something in dynamic routing , I am adding that through command line and reload the db to have the updated information . I am not force to restart the process . Can cluster module also work in the same way ? There is no option to add the DB entry through command line also . Is that's also a limitation in cluster module ? Please do help me . I am not getting anything regarding this in the module documentation . */Thanks & Regards/* /Sasmita Panda/ /Senior Network Testing and Software Engineer/ /3CLogic , ph:07827611765/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] I deleted an entry from clusterer table , but still opensips try to ping that node .
Hi Sasmita,
By default, there is no clusterer replication if "replicate_contacts_to"
parameter is not set in usrloc. Also, even if another node is sending
replication packets, they will no get processed on the receiving node
unless "accept_replicated_contacts" is set. On a typical setup, both
these parameter should be set on all nodes.
Are you getting any other errors in the logs besides that "parameter not
found" ?
Btw, I strongly suggest updating to 2.4 as it has received major
upgrades in terms of clustering.
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 01/02/2019 12:10 PM, Sasmita Panda wrote:
Hi,
I have another doubt . Please do help me .
When I am reading usrloc module document , its saying in a cluster if
we want to replicate the contacts across the cluster then we have to
set a parameter as below .
modparam("usrloc", "replicate_contacts_to", 1)
The default value is 0 , where no cluster id is mentioned .
I have not set this , I have a cluster having 2 node . While I am registering a
user , the contact is getting replicated between 2 nodes .
If I am trying to mention this parameter , then opensips is not getting started
. Its saying
*Parameter not found in module *
*So , my question is , if this parameter is not set , still how contact
replication is happening ? Is this the default behavior of cluster
module ? *
*May be my question is foolish ,it will be great if anybody will
explain this . *
**
**
*/Thanks & Regards/*
/Sasmita Panda/
/Senior Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Wed, Jan 2, 2019 at 12:22 PM Sasmita Panda <mailto:spa...@3clogic.com>> wrote:
Hi Sammy,
Yes , you are right . I need to reload the cluster data through MI
command . After reloading its seems fine .
I was not aware about the fact that the cluster data also get
shared with all nodes when I am adding that in 1 node only .
Thank you for your explanation . Its really helpful .
*/Thanks & Regards/*
/Sasmita Panda/
/Senior Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Mon, Dec 31, 2018 at 10:25 PM SamyGo mailto:govoi...@gmail.com>> wrote:
Hi,
Did you restart OpenSIPS process on both node1, and 2
simultaneously ? The way I look at this is one of the two
nodes kept the 3rd one in the memory and restarting both nodes
one at a time resulted in both sharing their node structure
and hence node3 stayed visible.
I think possible way to remove a node gracefully would be to
disable the node via the MI command and then remove from DB. I
will try doing this on my test setup as well.
Regards,
Sammy
On Fri, Dec 28, 2018 at 6:40 AM Sasmita Panda
mailto:spa...@3clogic.com>> wrote:
Hi All,
I have a cluster of 2 nodes . Both in working condition .
Then I added another node in the same cluster which is
down .
I restarted the opensips process , so it starts pinging
the new node to check its status . As the new node is
down , other nodes in the cluster wont get any reply for
the ping . Then I remove the 3rd node from the cluster
table and restart the opensips process .
Now what I am getting in logs is , still the 2 working
node in the cluster try to ping the 3rd node which is not
in the DB .
Is this an issue on the cluster module or I am doing
something wrong ?? Please help me .
*/Thanks & Regards/*
/Sasmita Panda/
/Senior Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
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Re: [OpenSIPS-Users] SIP-I and CANCEL
Hi Giovanni, Although I'm not claiming to be an expert at all in isup-sip interworking, the ITU Q.1912.5 document seems to say that it is not necessary to encapsulate REL in CANCEL. You can take a look at the example in Figure III.13/Q.1912.5 from Appendix III section of the document. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 12/07/2018 07:18 PM, Giovanni Maruzzelli wrote: On Fri, Dec 7, 2018 at 6:14 PM Bogdan-Andrei Iancu mailto:bog...@opensips.org>> wrote: Hi Giovanni, The CANCEL is hop-by-hop, which means each SIP hop will generate a completely new CANCEL (to be sent further) - there is not proxying, so the changes you do over the incoming CANCEL will not propagate into the outgoing CANCEL. Ciao Bogdan, yep, saw that I cannot recall it, but is there a standard case of pushing ISUP into CANCEL ? Not sure at all, I am looking around in various (too much) documentation around the intertube, not yet sure if you must put isup REL into CANCEL or not... Have a nice wend! -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Clustered User Location Full Sharing persistency and Cassandra
Hi Jock, The Cassandra module hasn't been well maintained for quite some time up until relatively recently (the code wasn't even compiling anymore) so there is actually no RPM for it (yet). Although the issues have been fixed and the module is now up to date with the latest Cassandra drivers, the features required for the usrloc clustering have only been implemented on the devel branch for Opensips 3.0. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 10/09/2018 06:40 PM, Jock McKechnie wrote: Good morning; We are attempting to implement a Full Sharing topology User Location cluster and I'm trying to make sense of whether this can work with Cassandra or not. At present we have a four-node cluster (across two datacentres) working in an in-memory state, with the nodes seeding from each other (by setting the clusterer node definition flags field to 'NULL' for all nodes). We'd like to consider having the data set stored to a Cassandra cluster should we require a complete OpenSIPS cluster restart, however the following line in the USRLOC documentation has me wondering: "Currently, registrations may optionally be fully managed inside NoSQL databases which support key/multi-value column-like associations. Example known backends to support these abstractions at the time of writing are MongoDB and Cassandra. Of these two, only the MongoDB OpenSIPS driver has been so far extended to implement the required NoSQL API endpoints." I read this to say that while both MongoDB and Cassandra can do what is required, only the MongoDB OpenSIPS modules are currently in a state to support it. Is this correct? I can't even work out what RPM the Cassandra module is in, which doesn't help with me attempting to test. :) My thanks for the clarification; - Jock ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] tls_mgm
Hi, Are you running the latest revision of the devel branch? There was a fix about a month ago regarding this issue. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 10/07/2018 08:10 AM, Slava Bendersky wrote: Hello Everyone, I am trying load certificate from database and set filter for ip and sip domain to *, but give me error Oct 7 01:06:05 canldev01-fedora27 /usr/sbin/opensips[17640]: ERROR:tls_mgm:parse_domain_address: TLS domain address has to be in [IP:port] format Oct 7 01:06:05 canldev01-fedora27 /usr/sbin/opensips[17640]: ERROR:tls_mgm:parse_domain_address: invalid TLS domain address [*] Oct 7 01:06:05 canldev01-fedora27 /usr/sbin/opensips[17640]: ERROR:tls_mgm:parse_match_addresses: Failed to parse address filter: * Oct 7 01:06:05 canldev01-fedora27 /usr/sbin/opensips[17640]: ERROR:tls_mgm:db_add_domain: Failed to parse address matching filters Oct 7 01:06:05 canldev01-fedora27 /usr/sbin/opensips[17640]: ERROR:tls_mgm:load_info: failed to add TLS domain 'default' id: 3, skipping... What is missing ? Thank you. volga629 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] tls_mgm modparam
Hi Dan,
This was indeed a parsing bug, I've pushed the fix. Thanks for spotting it!
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 09/10/2018 08:39 PM, Dan Pascu wrote:
Tried to use the following setting:
modparam("tls_mgm", "match_ip_address", "[default]*”)
This resulted in the following error:
Sep 10 20:19:07 [5987] ERROR:tls_mgm:parse_domain_address: TLS domain address
has to be in [IP:port] format
Sep 10 20:19:07 [5987] ERROR:tls_mgm:parse_domain_address: invalid TLS domain
address [*]
Sep 10 20:19:07 [5987] ERROR:tls_mgm:parse_match_addresses: Failed to parse
address filter: *
Sep 10 20:19:07 [5987] ERROR:tls_mgm:tlsp_set_match_addr: Failed to parse
domain matching filters for domain [default]
Sep 10 20:19:07 [5987] CRITICAL:core:yyerror: parse error in config file o.cfg, line 132,
column 20-21: Parameter not found in module -
can't set
Sep 10 20:19:07 [5987] ERROR:core:main: bad config file (1 errors)
Sep 10 20:19:07 [5987] NOTICE:core:main: Exiting
According to the documentation * should be accepted. Is this a bug in parsing
or is the documentation wrong?
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Re: [OpenSIPS-Users] tls_mgm
Hi Volga,
You are probably getting these errors because you are not providing the
TLS domain name in the parameter specifications. So all parameters
should look something like this:
modparam("tls_mgm", "tls_method", "[dom_name]TLSv1")
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 08/31/2018 05:15 PM, volga...@networklab.ca wrote:
Hello Everyone,
Recent versions of opensips tls_mgm module give this error.
opensips-3.0.0.b33b7a7e7-2.fc27.x86_64
[root@vprx00 ~]# opensips -V
version: opensips 3.0.0-dev (x86_64/linux)
flags: STATS: On, SHM_EXTRA_STATS, DISABLE_NAGLE, USE_MCAST, SHM_MMAP,
PKG_MALLOC, QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.
git revision: b33b7a7e7
main.c compiled on 12:56:54 Aug 14 2018 with gcc 7
Aug 31 09:08:26 [14440] DBG:core:load_module: loading module
/usr/lib64/opensips/modules/proto_hep.so
Aug 31 09:08:26 [14440] DBG:core:load_module: loading module
/usr/lib64/opensips/modules/proto_tls.so
Aug 31 09:08:26 [14440] DBG:core:add_module_dependency: adding type 2
dependency proto_tls - (module tls_mgm)
Aug 31 09:08:26 [14440] DBG:core:add_module_dependency: adding type 0
dependency proto_tls - (module proto_hep)
Aug 31 09:08:26 [14440] DBG:core:load_module: loading module
/usr/lib64/opensips/modules/tls_mgm.so
Aug 31 09:08:26 [14440] DBG:core:register_module: register_pv: tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: tls_mgm matches
module tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: found
in module tls_mgm [/usr/lib64/opensips/modules/]
Aug 31 09:08:26 [14440] ERROR:tls_mgm:split_param_val: No TLS domain name
Aug 31 09:08:26 [14440] CRITICAL:core:yyerror: parse error in config
file /etc/opensips/opensips.cfg, line 205, column 18-19: Parameter
not found in module - can't set
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: tls_mgm matches
module tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: found
in module tls_mgm [/usr/lib64/opensips/modules/]
Aug 31 09:08:26 [14440] ERROR:tls_mgm:split_param_val: No TLS domain name
Aug 31 09:08:26 [14440] CRITICAL:core:yyerror: parse error in config
file /etc/opensips/opensips.cfg, line 206, column 18-19: Parameter
not found in module - can't set
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: tls_mgm matches
module tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: found
in module tls_mgm [/usr/lib64/opensips/modules/]
Aug 31 09:08:26 [14440] ERROR:tls_mgm:split_param_val: No TLS domain name
Aug 31 09:08:26 [14440] CRITICAL:core:yyerror: parse error in config
file /etc/opensips/opensips.cfg, line 207, column 18-19: Parameter
not found in module - can't set
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: tls_mgm matches
module tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: found
in module tls_mgm [/usr/lib64/opensips/modules/]
Aug 31 09:08:26 [14440] ERROR:tls_mgm:split_param_val: No TLS domain name
Aug 31 09:08:26 [14440] CRITICAL:core:yyerror: parse error in config
file /etc/opensips/opensips.cfg, line 208, column 18-19: Parameter
not found in module - can't set
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: tls_mgm matches
module tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: found
in module tls_mgm [/usr/lib64/opensips/modules/]
Aug 31 09:08:26 [14440] ERROR:tls_mgm:split_param_val: No TLS domain name
Aug 31 09:08:26 [14440] CRITICAL:core:yyerror: parse error in config
file /etc/opensips/opensips.cfg, line 209, column 18-19: Parameter
not found in module - can't set
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: tls_mgm matches
module tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: found
in module tls_mgm [/usr/lib64/opensips/modules/]
Aug 31 09:08:26 [14440] ERROR:tls_mgm:split_param_val: No TLS domain name
Aug 31 09:08:26 [14440] CRITICAL:core:yyerror: parse error in config
file /etc/opensips/opensips.cfg, line 210, column 18-19: Parameter
not found in module - can't set
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: tls_mgm matches
module tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: found
in module tls_mgm [/usr/lib64/opensips/modules/]
Aug 31 09:08:26 [14440] ERROR:tls_mgm:split_param_val: No TLS domain name
Aug 31 09:08:26 [14440] CRITICAL:core:yyerror: parse error in config
file /etc/opensips/opensips.cfg, line 211, column 18-19: Parameter
not found in module - can't set
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: tls_mgm matches
module tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: found
in module tls_mgm [/usr/lib64/opensips/modules/]
Aug 31 09:08:26 [14440] ERROR:tls_mgm:split_param_val: No TLS domain name
Aug 31 09:08:2
Re: [OpenSIPS-Users] Help with my OpenSips architecture
Hi Dominic, The state listed by 'clusterer_list_cap' refers to the startup synchronization mechanism. When you register a sip device on an instance, that event is replicated on the spot to the other instances in the cluster, but older registrations are only pulled from other nodes at startup. You probably did not set the "seed" flag for any of the nodes, so the syncing process is unable to bootstrap correctly. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 08/08/2018 09:55 PM, Dominic wrote: There's something I'm not getting though, I defined my cluster with 2 opensips instances: # opensipsctl fifo clusterer_list Cluster:: 1 Node:: 2 DB_ID=-1 URL=bin:10.0.20.163:5566 <http://10.0.20.163:5566> Enabled=1 Link_state=Up Next_hop=2 Description=none # opensipsctl fifo clusterer_list_topology Cluster:: 1 Node:: 1 Neighbours=2 Node:: 2 Neighbours=1 However it seems usrloc is not synced as seen in the state below: # opensipsctl fifo clusterer_list_cap Cluster:: 1 Capability:: usrloc-contact-repl State=not synced However if I register 1 sip device on instance 1, I can see the device on both instances if I run "opensipsctl ul show". So it looks as though the usrloc is replicated properly so I wonder why "opensipsctl fifo clusterer_list_cap" has the "not synced" state. Dominic On Wed, Aug 8, 2018 at 8:00 AM, Dominic <mailto:wallnut.monk...@gmail.com>> wrote: On Wed, Aug 8, 2018 at 4:52 AM, Bogdan-Andrei Iancu mailto:bog...@opensips.org>> wrote: Hi Dominic, Please see inline. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com <http://www.opensips-solutions.com> OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ <http://opensips.org/training/OpenSIPS_Bootcamp_2018/> On 08/07/2018 10:36 PM, Dominic wrote: Hi all, just to make sure I'm one the right track here I'd like to share a few thoughts of where I'm currently going with my architecture, any feedback is really appreciated specially if it can help me avoid any pitfalls in the future. Here is what I'm trying to do: I currently have a a few asterisk servers directly facing the public internet, with sip devices registering directly with a specific asterisk. So my first goal was to have the sip devices register directly with OpenSips so I used the mid_registrar module for that, and forward the registers to the right asterisk. This seems to work very well so far. Secondly, I setup rtp proxy, on the same box as opensips, this also seems to work great so far. At this point I should be able to not have the asterisk facing public internet. Do you intend to move the * boxes to a private network ? or to keep them with public IP, but behind a firewall ? Yes I indend to move them to a private network behind a firewall with no public IP. The third thing I now want to do, and this is where I am currently stuck is to figure out what my options are in terms of high availability of the opensips. I figured I would go with a VIP setup with 2 identical opensips box. So in this type of setup i would have to replicate usrloc and dialogs between the 2 opensips instances right? Am I forgetting anything? yes, you are on the right tracks - use the dialog replication (with a sharing tag) and usrloc full sharing mode - this will give you the HA you are looking for (in combination with a VIP) Perfect, I'll give that a shot Am I on the right track with this type of setup or am I missing something big here? It's also worth mentioning that I am using OpenSIPS 2.4 Any advice/links with info on similar setup/code examples are very welcome For the HA, see https://blog.opensips.org/2018/03/15/scaling-distributing-and-partitioning-registrations-with-opensips-2-4/ <https://blog.opensips.org/2018/03/15/scaling-distributing-and-partitioning-registrations-with-opensips-2-4/> https://blog.opensips.org/2018/03/23/clustering-ongoing-calls-with-opensips-2-4/ <https://blog.opensips.org/2018/03/23/clustering-ongoing-calls-with-opensips-2-4/> Thanks Dominic ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> Thanks a lot for the replie
Re: [OpenSIPS-Users] Using LetsEncrypt certs with v2.4
Hi John, You conclusion about the modparam provisioning is not completely accurate. If you don't have any modparam statements in your script, the default preset values are only used as a backup for the default domains. And this is necessary in case you don't define the default domains in the database. So in this case, in order to always have some settings for the default domains, we do a fallback to the preset values. As such, if you do have a default client and server domain defined in the DB and no modparam statements, no script provisioning should be taken into account. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 08/08/2018 05:36 PM, John Quick wrote: Hi Vlad, Thanks for replying. Yes, I must have been looking at the wrong version in the documentation - the old version was stored in my browser history and I failed to notice. Even in the correct v2.4, the documentation for DB provisioning is more complete when read alongside the extra information in your reply. It is a pity that there is no way to completely disable provisioning from modparam parameters when you want only to use the DB. Removing the modparam statements does not disable them, but merely makes them use default preset values. John Quick Smartvox Limited Web: www.smartvox.co.uk ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Using LetsEncrypt certs with v2.4
Hi John, You are probably looking over the documentation for the wrong OpenSIPS version. The issues that you've mentioned appear in the 2.2 docs. The 2.4 docs should mostly cover your questions, but nevertheless: a) The domain field is only an identifier for the virtual TLS domain, but for default domains, indeed there is a special value, 'default'. b) * address - same meaning as the IP:port part of the 'server_domain' parameter * type - TLS client(1) or server(2) domain and 0 for defining both a client and server default domain with the same attributes * crl_check_all - check all files in the 'crl_dir' * crl_dir - path to directory containing Certificate Revocation Lists c) Both DB and script domains can be defined at the same time, but they should be seen as different sets of domains, so you should set a modparam only for a script defined domain. The blob database fields indeed should contain the contents of the certificates. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 08/01/2018 06:55 PM, John Quick wrote: Hi Bogdan, Thanks for your response to my earlier query. Im now trying to convert from modparam based definitions to provisioning certs from the DB. I cannot find a published example of a populated DB record in the tls_mgm table. Furthermore, the online documentation has gaps regarding DB Provisioning and it also contains this error: Section 1.7.14 describes a parameter db_mode, but if you try adding this it generates an error "parameter not found in module" Can you please help with an example record or at least answer these questions: a) What to put in the 'domain' field if I only want to set up one default domain. Should it be "default"? b) What are the following fields. I am not sure what they should contain: 'address', 'type', 'crl_check_all', 'crl_dir' c) How does provisioning from DB interact with provisioning from static modparam values? I got errors when I commented out modparam statements for "certificate" and "private_key" because the module was still looking for the "default" files, even though I am now provisioning from the DB. This means there is now ambiguity - certificates are defined both in files in modparam and also in blob fields in the DB. I assume the blob fields 'certificate', 'private_key' and 'ca_list' must contain the contents of the certificate, not the path to the file. This means I'll need to write a script to copy these data from the renewed LetsEncrypt certificates before issuing the MI reload command. By the way, the online module documentation for tls_mgm has a duplicate section - 1.7.18 is same as 1.7.19 John Quick Smartvox Limited Bogdan-Andrei Iancu bogdan at opensips.org Thu Jul 26 07:56:18 EDT 2018 Hi John, When the cert is configured via modparam, the cert is loaded on startup by OpenSIPS, so any renewal of the cert will have 0 impact on OpenSIPS - so you will have to restart after each renewal. I suggest you to provision the certs via DB (and not script), so you can do a reload after renewal, with any need to restart opensips. Regards, Bogdan-Andrei Iancu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 2.4 and AnyCast Setup
Hi Jonathan, OpenSIPS doesn't do that automatically because the setting should depend on what roles the nodes have when you first boot up your cluster. As such, on the node that first holds the VIP, and will generally be your primary node, OpenSIPS should start with the sharing tag as active in order to correctly process the traffic as soon as possible. And on the backup node, the opposite. From then on, when a switchover happens at runtime, the tag state will be changed by running the MI command "dlg_set_sharing_tag_active" when the VRRPD state transitions (this is where you will need some extra logic). Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 21.06.2018 17:13, Jonathan Mabrito wrote: Thanks! So it does sound like AnyCast is at the routing protocol level...I was thinking it was multicast for some reason. Alrighty then...I know what needs to be done on my end and a few scenarios I can try and get it working. My main roadblock is that we are running Cisco EIGRP setup (not that this is your issue...just more summarizing on what I found for anyone else who might be reading this)...so either I can redistribute some static routes into our EIGRP process or setup a new OSPF process and a new network address space (say another /24) and redistribute that into our EIGRP process. OSPF would be the more elegant and reliable way (as each server will be advertising itself when its up) but static is going to be easier to get going and less overhead of not having to setup OSPF (but with static, it wont technically know when a server is down and I am then relying on longer time outs before the other static route metric kicks in). Anywho...I have some trial and error on my side now. I do have one more question. Right now we use VRRPD between our two 2.3 nodes. If I was to stay with this method for 2.4, but do the dialog replication, that would also be an acceptable solution for us. I am confused though on this statement in the Clustering blog: "When starting OpenSIPS, you need to check the HA status of the node (by inspecting the HA system) and to decide which node will start as owner of the tag:" Is that something OpenSIPS does automatically to determine if that VRRPD VIP IP is active on itself and set the "dlg_sharing_tag" to active or do I have to add extra logic somewhere? On Thu, Jun 21, 2018 at 6:24 AM Răzvan Crainea <mailto:raz...@opensips.org>> wrote: Hi, Jonathan! The guys from Irontech have written a very nice blogpost regarding how they set up their environment. You should definitely give it a try: https://blog.irontec.com/testing-opensips-anycast-support/ Hope it helps! Best regards, Răzvan On 06/20/2018 10:36 PM, Jonathan Mabrito wrote: > Hoping I can get some guidance/pointed in the right direction on this > subject. Looking at standing up a 2.4 AnyCast 2 node cluster and try out > the new 2.4 dialog cluster features (will be a upgrade/migration from > 2.3). Not entirely sure what is needed to make this work from the > AnyCast perspective > > When setting up AnyCast, I know the network and routers need some config > to support AnyCast. I know this question falls more in the topics of > networking, but hoping I can get some guidance on what was done maybe on > the test servers when it was developed? Not looking for specific > configs, just xyz you should look at this or that type responses if > possible. > > Right now, I have the two nodes stood up, but the AnyCast IP I am trying > to bring online is not reachable yet. I know I need to do some network > statements, just trying to grasp what they are. > > Any advice/help to point me in the right direction would be appreciated. > -- > -Jonathan > > > ___ > Users mailing list > Users@lists.opensips.org <mailto:Users@lists.opensips.org> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- -Jonathan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to extract SIP-I ISUP 'Redirecting number' parameter
Hi, I've pushed the fix, but bear in mind that you could still extract the redirecting number without full subfield access support. By just reading from "$isup_param(Redirecting number)" you get, as a string, the full ISUP parameter value in hex representation. From that, knowing the parameter format from the ISUP specification, you could parse the number yourself in the OpenSIPS script. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 29.03.2018 13:52, Eugene Prokopiev wrote: Btw, is it possible to extract ISUP part as is in binary form and read by some external script? Can anybody advice me any library to parse ISUP binary data? 2018-03-29 12:50 GMT+03:00 Eugene Prokopiev : Thank you! I submitted https://github.com/OpenSIPS/opensips/issues/1323 - maybe your patch can be applied? 2018-03-29 10:53 GMT+03:00 Arto Kuiri : Hello, I had same problem and made my own patch. See attached diff, it adds support for: - redirection info (reason in same format as in SIP diversion header, commented ITU Q.763 version) - original called number - redirection number Meant to make feature request/patch, but somehow forgot it. Best regards, Arto Kuiri Lähettäjä: Users käyttäjän Vlad Patrascu puolesta Lähetetty: 28. maaliskuuta 2018 20:48 Vastaanottaja: users@lists.opensips.org Aihe: Re: [OpenSIPS-Users] How to extract SIP-I ISUP 'Redirecting number' parameter Hi, Unfortunately "Redirecting Number" is not currently supported, but you could open up a feature request regarding this, as it would be not very difficult to implement. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com Home — OpenSIPS Solutions www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 28.03.2018 18:32, Eugene Prokopiev wrote: Hi, I need to extract ISUP parameter 'Redirecting number' from SIP-I INVITE request and add it as SIP header with 'X-' prefix to process in FreeSWITCH. Can't found any about 'Redirecting number' in http://www.opensips.org/html/docs/modules/2.3.x/sip_i.html, so how to extract it? 'Redirecting number' parameter value is visible in Wireshark now as ISUP parameter with t=11 and l=7 ___ 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 -- WBR, Eugene Prokopiev ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to extract SIP-I ISUP 'Redirecting number' parameter
Hi, Unfortunately "Redirecting Number" is not currently supported, but you could open up a feature request regarding this, as it would be not very difficult to implement. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 28.03.2018 18:32, Eugene Prokopiev wrote: Hi, I need to extract ISUP parameter 'Redirecting number' from SIP-I INVITE request and add it as SIP header with 'X-' prefix to process in FreeSWITCH. Can't found any about 'Redirecting number' in http://www.opensips.org/html/docs/modules/2.3.x/sip_i.html, so how to extract it? 'Redirecting number' parameter value is visible in Wireshark now as ISUP parameter with t=11 and l=7 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] dialog replication
Hi Pasan, You could have another OpenSIPS instance that exposes the service IP, in front of the 2 clustered HA nodes. Calls would first land on this box and then would be dispatched to one of the 2 floating IPs. You could for example hash over the From tag or Callid in order to send to the same IP. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 01.03.2018 03:21, Pasan Meemaduma via Users wrote: Hi Vlad, Thanks for your response. I think I'll have to wait until 2.4 transaction replication come in to action if I am to go with anycast ip. I didn't understand how it would solve with 2 moveable ips. how do i make sure dialog will tie to a particular ip ? On Wednesday, 28 February 2018, 17:04, Vlad Patrascu wrote: Hi, Yes, you are correct, this is currently a limitation. But it could also work in an "active/active" sort of setup if you have 2 movable IPs and each dialog is tied to one of the IPs. Full anycast support though is coming up in OpenSIPS 2.4. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com <http://www.opensips-solutions.com/> On 28.02.2018 04:46, Pasan Meemaduma via Users wrote: Hi Vlad, On a second thought its how anycast should work, packets could get to the closest node so basically with 2.3.3 we can't replicate transactions across mulitple opensips servers ? dialog replication would only work for an active/passive setup doesn't it ? On Wednesday, 28 February 2018, 7:36, Pasan Meemaduma via Users <mailto:users@lists.opensips.org> wrote: Hi Vlad, That explains the issue then, my sip listener ip is an anycast one, and for some weired reason replies for INVITE that leaves node2 receive by node1 hence it can't create the dialog. I checked with node2 being down and recovered and calls init via node1 always received final reply hence dialog replication works. So it means my anycast configuration is broken right ? Thanks for you time to look in to it. On Tuesday, 27 February 2018, 21:42, Vlad Patrascu <mailto:vl...@opensips.org> wrote: Hi Pasan, I don't see anything in the logs or your cluster configuration which could indicate that the dialogs don't replicate. Are you sure that the problem is consistently reproducible and that the INVITE for the call in question receives a final reply? Dialogs are replicated when they get to the confirmed state, and judging from the logs, this doesn't appear to be happening for that call. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com <http://www.opensips-solutions.com/> On 27.02.2018 04:15, Pasan Meemaduma via Users wrote: Hi Vlad, I have sent you the full debug logs as requested. Also clusterer_list mi command gives following output when dialog replication stopped working. node1>>opensipsctl fifo clusterer_list Cluster:: 1 Node:: 2 DB_ID=5 URL=bin:10.3.1.137:5566 Enabled=1 Link_state=Up Next_hop=2 Description=Node 2 node2>> opensipsctl fifo clusterer_list Cluster:: 1 Node:: 1 DB_ID=4 URL=bin:10.3.1.136:5566 Enabled=1 Link_state=Up Next_hop=1 Description=Node 1 1 On Monday, 26 February 2018, 18:02, Vlad Patrascu <mailto:vl...@opensips.org> wrote: Hi, Can you send the full logs for both nodes from the time that node1 restarts onwards? Also, what is the output of 'clusterer_list' mi command on the instances? Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com <http://www.opensips-solutions.com/> On 26.02.2018 08:38, Pasan Meemaduma via Users wrote: Hi Guys, Its me again :). I'm using dialog replication in opensips 2.3.3 and it appears it doesn't work after a node goes down and come back online. The recovered node doesn't seems to receiving dialog info via binary interface. I have node1 and node2 with dialog replication on. everything works fine and if I shutdown node1 and bring it back online after a while node2 doesn't send new call dialog info via binary interface. I have attach the debug trace from both nodes, if you need anything else let me know. I'm also using an anycast ip as the service ip. on node2 for new call Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:build_new_dlg: new dialog 0x7f188afd3bf8 (c=ktQ0Pkdwz50qGYjED6Brpw..,f=sip:X@somedomain;transport=UDP,t=sip:+YYY@somedomain;transport=UDP,ft=2b161508) on hash 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:init_leg_info: route_set , contact sip:X@192.168.27.11:56419;transport=UDP <mailto:sip:X@192.168.27.11:56419;transport=UDP>, cseq 2 and bind_addr udp:10.3.3.1:5060 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_add_leg_info: set leg 0 for 0x7f188afd3bf8: tag=<2b161508> rcseq=<0> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips
Re: [OpenSIPS-Users] dialog replication
Hi, Yes, you are correct, this is currently a limitation. But it could also work in an "active/active" sort of setup if you have 2 movable IPs and each dialog is tied to one of the IPs. Full anycast support though is coming up in OpenSIPS 2.4. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 28.02.2018 04:46, Pasan Meemaduma via Users wrote: Hi Vlad, On a second thought its how anycast should work, packets could get to the closest node so basically with 2.3.3 we can't replicate transactions across mulitple opensips servers ? dialog replication would only work for an active/passive setup doesn't it ? On Wednesday, 28 February 2018, 7:36, Pasan Meemaduma via Users wrote: Hi Vlad, That explains the issue then, my sip listener ip is an anycast one, and for some weired reason replies for INVITE that leaves node2 receive by node1 hence it can't create the dialog. I checked with node2 being down and recovered and calls init via node1 always received final reply hence dialog replication works. So it means my anycast configuration is broken right ? Thanks for you time to look in to it. On Tuesday, 27 February 2018, 21:42, Vlad Patrascu wrote: Hi Pasan, I don't see anything in the logs or your cluster configuration which could indicate that the dialogs don't replicate. Are you sure that the problem is consistently reproducible and that the INVITE for the call in question receives a final reply? Dialogs are replicated when they get to the confirmed state, and judging from the logs, this doesn't appear to be happening for that call. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com <http://www.opensips-solutions.com/> On 27.02.2018 04:15, Pasan Meemaduma via Users wrote: Hi Vlad, I have sent you the full debug logs as requested. Also clusterer_list mi command gives following output when dialog replication stopped working. node1>>opensipsctl fifo clusterer_list Cluster:: 1 Node:: 2 DB_ID=5 URL=bin:10.3.1.137:5566 Enabled=1 Link_state=Up Next_hop=2 Description=Node 2 node2>> opensipsctl fifo clusterer_list Cluster:: 1 Node:: 1 DB_ID=4 URL=bin:10.3.1.136:5566 Enabled=1 Link_state=Up Next_hop=1 Description=Node 1 1 On Monday, 26 February 2018, 18:02, Vlad Patrascu <mailto:vl...@opensips.org> wrote: Hi, Can you send the full logs for both nodes from the time that node1 restarts onwards? Also, what is the output of 'clusterer_list' mi command on the instances? Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com <http://www.opensips-solutions.com/> On 26.02.2018 08:38, Pasan Meemaduma via Users wrote: Hi Guys, Its me again :). I'm using dialog replication in opensips 2.3.3 and it appears it doesn't work after a node goes down and come back online. The recovered node doesn't seems to receiving dialog info via binary interface. I have node1 and node2 with dialog replication on. everything works fine and if I shutdown node1 and bring it back online after a while node2 doesn't send new call dialog info via binary interface. I have attach the debug trace from both nodes, if you need anything else let me know. I'm also using an anycast ip as the service ip. on node2 for new call Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:build_new_dlg: new dialog 0x7f188afd3bf8 (c=ktQ0Pkdwz50qGYjED6Brpw..,f=sip:X@somedomain;transport=UDP,t=sip:+YYY@somedomain;transport=UDP,ft=2b161508) on hash 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:init_leg_info: route_set , contact sip:X@192.168.27.11:56419;transport=UDP <mailto:sip:X@192.168.27.11:56419;transport=UDP>, cseq 2 and bind_addr udp:10.3.3.1:5060 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_add_leg_info: set leg 0 for 0x7f188afd3bf8: tag=<2b161508> rcseq=<0> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:link_dlg: ref dlg 0x7f188afd3bf8 with 3 -> 3 in h_entry 0x7f188afc3828 - 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:new_dlg_val: inserting =<45ad4c76-1abe-11e8-9410-831894b67d0c> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_onreq: t hash_index = 47425, t label = 2069082013 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_update_contact: Updated dialog 0x7f188afd3bf8 contact to <mailto:sip:X@192.168.27.11:56419;transport=UDP> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry 0x7f188afc3828 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 3 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:ref_dlg: ref dlg
Re: [OpenSIPS-Users] dialog replication
Hi Pasan, I don't see anything in the logs or your cluster configuration which could indicate that the dialogs don't replicate. Are you sure that the problem is consistently reproducible and that the INVITE for the call in question receives a final reply? Dialogs are replicated when they get to the confirmed state, and judging from the logs, this doesn't appear to be happening for that call. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 27.02.2018 04:15, Pasan Meemaduma via Users wrote: Hi Vlad, I have sent you the full debug logs as requested. Also clusterer_list mi command gives following output when dialog replication stopped working. node1>>opensipsctl fifo clusterer_list Cluster:: 1 Node:: 2 DB_ID=5 URL=bin:10.3.1.137:5566 Enabled=1 Link_state=Up Next_hop=2 Description=Node 2 node2>> opensipsctl fifo clusterer_list Cluster:: 1 Node:: 1 DB_ID=4 URL=bin:10.3.1.136:5566 Enabled=1 Link_state=Up Next_hop=1 Description=Node 1 1 On Monday, 26 February 2018, 18:02, Vlad Patrascu wrote: Hi, Can you send the full logs for both nodes from the time that node1 restarts onwards? Also, what is the output of 'clusterer_list' mi command on the instances? Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com <http://www.opensips-solutions.com/> On 26.02.2018 08:38, Pasan Meemaduma via Users wrote: Hi Guys, Its me again :). I'm using dialog replication in opensips 2.3.3 and it appears it doesn't work after a node goes down and come back online. The recovered node doesn't seems to receiving dialog info via binary interface. I have node1 and node2 with dialog replication on. everything works fine and if I shutdown node1 and bring it back online after a while node2 doesn't send new call dialog info via binary interface. I have attach the debug trace from both nodes, if you need anything else let me know. I'm also using an anycast ip as the service ip. on node2 for new call Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:build_new_dlg: new dialog 0x7f188afd3bf8 (c=ktQ0Pkdwz50qGYjED6Brpw..,f=sip:X@somedomain;transport=UDP,t=sip:+YYY@somedomain;transport=UDP,ft=2b161508) on hash 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:init_leg_info: route_set , contact sip:X@192.168.27.11:56419;transport=UDP <mailto:sip:X@192.168.27.11:56419;transport=UDP>, cseq 2 and bind_addr udp:10.3.3.1:5060 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_add_leg_info: set leg 0 for 0x7f188afd3bf8: tag=<2b161508> rcseq=<0> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:link_dlg: ref dlg 0x7f188afd3bf8 with 3 -> 3 in h_entry 0x7f188afc3828 - 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:new_dlg_val: inserting =<45ad4c76-1abe-11e8-9410-831894b67d0c> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_onreq: t hash_index = 47425, t label = 2069082013 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_update_contact: Updated dialog 0x7f188afd3bf8 contact to <mailto:sip:X@192.168.27.11:56419;transport=UDP> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry 0x7f188afc3828 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 3 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 4 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 3 in entry 0x7f188afc3828 Feb 26 17:28:56 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry 0x7f188afc3828 Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 1 to state 5, due event 4 Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:dlg_onreply: dialog 0x7f188afd3bf8 failed (negative reply) Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 1 in entry 0x7f188afc3828 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 5 to state 5, due event 1 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 0 in entry 0x7f188afc3828 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: ref <=0 for dialog 0x7f188afd3bf8 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:destroy_dlg: destroying dialog 0x7f188afd3bf8 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]:
Re: [OpenSIPS-Users] dialog replication
Hi, Can you send the full logs for both nodes from the time that node1 restarts onwards? Also, what is the output of 'clusterer_list' mi command on the instances? Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 26.02.2018 08:38, Pasan Meemaduma via Users wrote: Hi Guys, Its me again :). I'm using dialog replication in opensips 2.3.3 and it appears it doesn't work after a node goes down and come back online. The recovered node doesn't seems to receiving dialog info via binary interface. I have node1 and node2 with dialog replication on. everything works fine and if I shutdown node1 and bring it back online after a while node2 doesn't send new call dialog info via binary interface. I have attach the debug trace from both nodes, if you need anything else let me know. I'm also using an anycast ip as the service ip. on node2 for new call Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:build_new_dlg: new dialog 0x7f188afd3bf8 (c=ktQ0Pkdwz50qGYjED6Brpw..,f=sip:X@somedomain;transport=UDP,t=sip:+YYY@somedomain;transport=UDP,ft=2b161508) on hash 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:init_leg_info: route_set , contact sip:X@192.168.27.11:56419;transport=UDP, cseq 2 and bind_addr udp:10.3.3.1:5060 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_add_leg_info: set leg 0 for 0x7f188afd3bf8: tag=<2b161508> rcseq=<0> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:link_dlg: ref dlg 0x7f188afd3bf8 with 3 -> 3 in h_entry 0x7f188afc3828 - 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:new_dlg_val: inserting =<45ad4c76-1abe-11e8-9410-831894b67d0c> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_onreq: t hash_index = 47425, t label = 2069082013 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_update_contact: Updated dialog 0x7f188afd3bf8 contact to Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry 0x7f188afc3828 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 3 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 4 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 3 in entry 0x7f188afc3828 Feb 26 17:28:56 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry 0x7f188afc3828 Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 1 to state 5, due event 4 Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:dlg_onreply: dialog 0x7f188afd3bf8 failed (negative reply) Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 1 in entry 0x7f188afc3828 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 5 to state 5, due event 1 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 0 in entry 0x7f188afc3828 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: ref <=0 for dialog 0x7f188afd3bf8 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:destroy_dlg: destroying dialog 0x7f188afd3bf8 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:destroy_dlg: dlg expired or not in list - dlg 0x7f188afd3bf8 [2317:677053409] with clid 'ktQ0Pkdwz50qGYjED6Brpw..' and tags '2b161508' 'NULL' on node1 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28
Re: [OpenSIPS-Users] SIP-I Calling Party number being changed
Hi,
I'm not an expert on ITU-T standards but from what I know, the "+" sign
is only an indication that an international prefix is required and thus
cannot be set as a digit in the ISUP number related parameters. Being an
unsupported character, the sip_i module wrote it as 0. Maybe a warning
log message would be helpful.
As such you should probably replace the plus sign with the international
call prefix and only after that set the ISUP parameter.
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 01/15/2018 12:39 PM, Aqs Younas wrote:
Greetings list,
I am using the sip-i module for conversion between sip to sip-i. I
want to set calling party number and called party number in e164
format. Below is how my code looks like.
route{
...
xlog("Here is Called party number: $rU");
xlog("Here is Calling party number: $fU");
do_accounting("log","missed");
add_isup_part("Initial address");
$isup_param(Nature of connection indicators) = "0x10";
$isup_param(Forward Call Indicators) = "0x6001";
$isup_param(Called party number|Nature of address indicator) = 4;
$isup_param(Called party number|Numbering plan indicator) = 1;
$isup_param(Called party number|Address signal) = $rU;
$isup_param(Calling party number|Nature of address indicator) = 4;
$isup_param(Calling party number|Numbering plan indicator) = 1;
$isup_param(Calling party number|Screening indicator) = 4;
$isup_param(Transmission Medium Requirement) = "speech";
$isup_param(Calling party number|Address signal) = $fU;
...
}
branch_route[per_branch_ops] {
xlog("NOA is: $isup_param_str(Called Party Number|Nature of
address indicator)");
xlog("Called party is: $isup_param_str(Called Party
Number|Address signal)");
xlog("NOA is: $isup_param_str(Calling Party Number|Nature of
address indicator)");
xlog("Calling is: $isup_param_str(Calling Party Number|Address
signal)");
}
But I see Calling party and Called party number being changed from
what i had set.
This is what I set.
Jan 15 10:16:01 gridtele-ug-vpn ./opensips[6911]: Here is Called
party number: +256723000422
Jan 15 10:16:01 gridtele-ug-vpn ./opensips[6911]: Here is Calling
party number: +12064248192
And what appears in ISUP body. You can notice *+ *changed to 0.
Jan 15 10:16:01 gridtele-ug-vpn ./opensips[6911]: NOA is: international
*Jan 15 10:16:01 gridtele-ug-vpn ./opensips[6911]: Called party is:
0256723000422*
Jan 15 10:16:01 gridtele-ug-vpn ./opensips[6911]: NOA is: international
*Jan 15 10:16:01 gridtele-ug-vpn ./opensips[6911]: Calling is:
012064248192*
Any help/suggestion is much appreciated.
Thanks.
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Re: [OpenSIPS-Users] Can't add bin_listen interface to script
Hi, The "bin_listen" parameter was removed in OpenSIPS 2.2.X and bin listeners should be defined using "listen = bin:a.b.c.d:". Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 10/30/2017 11:11 AM, Matine Tinthaitae wrote: Hi, I'm trying to use the Binary Interface to replicate dialogs in OpenSIPS 2.2.5. When ever i add "bin_listen = a.b.c.d:" opensips fails to start. Oct 30 10:08:27 [26532] CRITICAL:core:yyerror: parse error in config file /usr/local//etc/opensips/opensips.cfg, line 143, column 1-11: syntax error Oct 30 10:08:27 [26532] CRITICAL:core:yyerror: parse error in config file /usr/local//etc/opensips/opensips.cfg, line 143, column 1-11: Oct 30 10:08:27 [26532] ERROR:core:main: bad config file (2 errors) Oct 30 10:08:27 [26532] NOTICE:core:main: Exiting Is the binary interface not available in 2.2.5 or what am i doing wrong? Thank you in advance! //Matine ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] clusterer module
Hi Denis, For around 6 minutes since startup of the first proxy, the second proxy is either not fully booted up or the clusterer table is not loaded so it can't reply to its pings. The issue occurs from this point on, but the chunk of log messages you've provided ends soon after so I couldn't pinpoint the problem. Also the other logs weren't really helpful. So could you send a slightly longer sequence of logs from startup? Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 09/18/2017 10:54 AM, Denis via Users wrote: Hello! Is there any information about problem? Thank you. -- С уважением, Денис. Best regards, Denis 08.09.2017, 13:46, "Denis via Users" : Hello, Vlad! Sorry for long answer. So, here (https://yadi.sk/i/j9a4CZhS3MiJEm) you can find the log file. In this file i tried to display information in chronological order. In the log file: opensips2.2. - first proxy opensipsproxy1 - second proxy opensipsctl.sh - script for the first proxy opensipsproxyctl.sh - script for the second proxy 1.1.1.1 - IP of the server (as i mentioned below both proxies are on the same server). Thank you for any help. -- С уважением, Денис. Best regards, Denis 30.08.2017, 13:05, "Vlad Patrascu" <mailto:vl...@opensips.org>>: Hi Denis, I can't really tell what the problem is from the information you've provided (the traces aren't much help as the messages between nodes are binary data). Can you provide some logs relevant to the clusterer (including debug level) from around the time when the nodes first try to establish the links ? Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com <http://www.opensips-solutions.com/> On 08/30/2017 11:46 AM, Denis via Users wrote: Hello! Any information about the problem? Thank you. -- С уважением, Денис. Best regards, Denis 25.08.2017, 16:51, "Denis via Users" <mailto:users@lists.opensips.org>: Hello! Back to 2.2.5 resolved the problem. I got replication of, for example, dialog/profile -- С уважением, Денис. Best regards, Denis 24.08.2017, 17:46, "Denis via Users" <mailto:users@lists.opensips.org>>: Hello! I am developing VoIP architecture with two proxies and one LB. Between these proxies i have a cluster, using corresponding module. All instances (proxies and LB) are on the same server. I started from 2.2 and there were no problems with the cluster. Recently, I upgraded Opensips from 2.2. to 2.3.1 and have got such situation ./opensipsctl.sh fifo clusterer_list Cluster:: 1 Node:: 2 DB_ID=2 URL=bin:1.1.1.1:5041 Enabled=1 Link_state=Probe Next_hop=none Description=none ./opensipsproxyctl.sh fifo clusterer_list Cluster:: 1 Node:: 1 DB_ID=1 URL=bin:1.1.1.1:5040 Enabled=1 Link_state=Up Next_hop=1 Description=none tcpdump 10:33:47.101956 IP 1.1.1.1.40428 > 1.1.1.1.5040: Flags [P.], seq 381492:381525, ack 3259636718, win 342, options [nop,nop,TS val 65425778 ecr 65424597], length 33 10:33:47.101984 IP 1.1.1.1.5040 > 1.1.1.1.40428: Flags [.], ack 33, win 6146, options [nop,nop,TS val 65425778 ecr 65425778], length 0 10:33:51.725437 IP 1.1.1.1.40428 > 1.1.1.1.5040: Flags [P.], seq 33:66, ack 1, win 342, options [nop,nop,TS val 65426934 ecr 65425778], length 33 10:33:51.725458 IP 1.1.1.1.5040 > 1.1.1.1.40428: Flags [.], ack 66, win 6146, options [nop,nop,TS val 65426934 ecr 65426934], length 0 and 10:35:12.589476 IP 1.1.1.1.34850 > 1.1.1.1.5041: Flags [P.], seq 1471487774:1471487807, ack 2063326017, win 342, options [nop,nop,TS val 65447150 ecr 65445995], length 33 10:35:12.589491 IP 1.1.1.1.5041 > 1.1.1.1.34850: Flags [.], ack 33, win 342, options [nop,nop,TS val 65447150 ecr 65447150], length 0 10:35:17.158143 IP 1.1.1.1.34850 > 1.1.1.1.5041: Flags [P.], seq 33:66, ack 1, win 342, options [nop,nop,TS val 65448292 ecr 65447150], length 33 10:35:17.158153 IP 1.1.1.1.5041 > 1.1.1.1.34850: Flags [.], ack 66, win 342, options [nop,nop,TS val 65448292 ecr 65448292], length 0 As you can see node 1 cannot see node 2 (Link_state=Probe). The question is why? Thank you. -- С уважением, Денис. Best regards, Denis , ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users , ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users , ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailm
Re: [OpenSIPS-Users] Use of colon in XML element names
Hi xaled,
There should be no problem with colons in element names (libxml will
produce some errors, because namespaces are not handled by the xml
module and the prefix is simply considered as part of the element name,
but they can be ignored), your issue seems to be caused by something
else. What errors or other relevant messages do the logs produce? I do
spot on the other hand a wrong access to the first element when adding
the attribute, which should look like
"$xml(xml_s/tns:messageType.attr/xmlns:tns)" but the document should
still be created fine without that attribute.
As for adding the encoding, you could initialize the $xml variable
directly with the required XML declaration, like this:
$xml(xml_s) = 'encoding="UTF-8"?>';
Formatted output is not possible currently, you can open a feature
request on Github regarding this, but if you really need the XML
document to actually contain the indentation characters, you could do so
explicitly when inserting the nodes (as in the documentation example).
Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
On 08/20/2017 07:08 PM, xaled wrote:
Hi,
Just tried the new xml module and have some issues with colon in
element names. Should I use some escape magic for colons with opensips
xml module?
Another question is how to add encoding="UTF-8" to have it like this -
It would be also great to have formatted XML output. Something like
format parameter in toString method of XML::LibXML::Document in perl.
test:/usr/local/etc/opensips # /usr/local/sbin/opensips -V
version: opensips 2.3.1 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
git revision: 72fa146
main.c compiled on 18:53:05 Aug 16 2017 with gcc 4.8
$xml(xml_s) = "";
$xml(xml_s.attr/xmlns:tns) = "http://test.org/abc/params/xml/abc";;
$xml(xml_s/tns:messageType) = "";
$xml(xml_s/tns:messageType/tns:crtg) =
"";
$xml(xml_s/tns:messageType/tns:crtg/tns:ControlIndicators) =
"5";
xlog("XML_TEST: $xml(xml_s)\n");
2017-08-20T17:46:38.320273+02:00 test /usr/local/sbin/opensips[32422]:
XML_TEST: #012
The final XML should be like this:
http://test.org/abc/params/xml/abc";>
5
Thanks,
xaled
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Re: [OpenSIPS-Users] clusterer module
Hi Denis, I can't really tell what the problem is from the information you've provided (the traces aren't much help as the messages between nodes are binary data). Can you provide some logs relevant to the clusterer (including debug level) from around the time when the nodes first try to establish the links ? Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 08/30/2017 11:46 AM, Denis via Users wrote: Hello! Any information about the problem? Thank you. -- С уважением, Денис. Best regards, Denis 25.08.2017, 16:51, "Denis via Users" : Hello! Back to 2.2.5 resolved the problem. I got replication of, for example, dialog/profile -- С уважением, Денис. Best regards, Denis 24.08.2017, 17:46, "Denis via Users" <mailto:users@lists.opensips.org>>: Hello! I am developing VoIP architecture with two proxies and one LB. Between these proxies i have a cluster, using corresponding module. All instances (proxies and LB) are on the same server. I started from 2.2 and there were no problems with the cluster. Recently, I upgraded Opensips from 2.2. to 2.3.1 and have got such situation ./opensipsctl.sh fifo clusterer_list Cluster:: 1 Node:: 2 DB_ID=2 URL=bin:1.1.1.1:5041 Enabled=1 Link_state=Probe Next_hop=none Description=none ./opensipsproxyctl.sh fifo clusterer_list Cluster:: 1 Node:: 1 DB_ID=1 URL=bin:1.1.1.1:5040 Enabled=1 Link_state=Up Next_hop=1 Description=none tcpdump 10:33:47.101956 IP 1.1.1.1.40428 > 1.1.1.1.5040: Flags [P.], seq 381492:381525, ack 3259636718, win 342, options [nop,nop,TS val 65425778 ecr 65424597], length 33 10:33:47.101984 IP 1.1.1.1.5040 > 1.1.1.1.40428: Flags [.], ack 33, win 6146, options [nop,nop,TS val 65425778 ecr 65425778], length 0 10:33:51.725437 IP 1.1.1.1.40428 > 1.1.1.1.5040: Flags [P.], seq 33:66, ack 1, win 342, options [nop,nop,TS val 65426934 ecr 65425778], length 33 10:33:51.725458 IP 1.1.1.1.5040 > 1.1.1.1.40428: Flags [.], ack 66, win 6146, options [nop,nop,TS val 65426934 ecr 65426934], length 0 and 10:35:12.589476 IP 1.1.1.1.34850 > 1.1.1.1.5041: Flags [P.], seq 1471487774:1471487807, ack 2063326017, win 342, options [nop,nop,TS val 65447150 ecr 65445995], length 33 10:35:12.589491 IP 1.1.1.1.5041 > 1.1.1.1.34850: Flags [.], ack 33, win 342, options [nop,nop,TS val 65447150 ecr 65447150], length 0 10:35:17.158143 IP 1.1.1.1.34850 > 1.1.1.1.5041: Flags [P.], seq 33:66, ack 1, win 342, options [nop,nop,TS val 65448292 ecr 65447150], length 33 10:35:17.158153 IP 1.1.1.1.5041 > 1.1.1.1.34850: Flags [.], ack 66, win 342, options [nop,nop,TS val 65448292 ecr 65448292], length 0 As you can see node 1 cannot see node 2 (Link_state=Probe). The question is why? Thank you. -- С уважением, Денис. Best regards, Denis , ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users , ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips change the order of requests
Hello, I would say this an UA issue - OpenSIPS does not do any enforcing on the order of outgoing SIP packets, so if the UA sends an INVITE immediately followed by an UPDATE, there exists the risk of the UPDATE reaching the endpoint first ( even if the proxy would maintain the the packet order, since UDP is used as transport, there would still be no guarantee that the endpoint receives them in the same order ). Can you configure the UA to only send the UPDATE after the INVITE transaction has finished ? This would be the only way to ensure there are no races. Best Regards, Vlad Paiu OpenSIPS Developer On 22.10.2015 20:05, Patrick Wakano wrote: Hello Opensips list! I already explained my scenario in this e-mail (http://lists.opensips.org/pipermail/users/2015-October/032859.html) where the Ack was propagated with wrong CSeq number when using the in-dialog ping options. Vlad did a great job and already correct this bug! (https://github.com/OpenSIPS/opensips/issues/680) Now in a scenario where I disabled the in-dialog ping, I faced another issue. I mentioned that sometimes, due to unknown reasons, when I received a Re-Invite immediately followed by an Update, the Update gets forwarded first. The problem is that, since CSeq numbers are not altered, when this inversion happens, my peer rejects the Re-Invite because its CSeq number is lower than the Update one. Probably Opensips has to respect the ordering of in-dialog requests and forward them accordingly, instead of handling them asynchronously and forwarding right away... Best regards, Patrick ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Exported Asyncronous functions
Hello,
There is no way to do that.
Just move your 'line after exec;' to the resume route, or put it in a
new route that will get called from within the resume route.
Regards,
Vlad Paiu
OpenSIPS Developer
On 22.10.2015 17:39, Dragomir Haralambiev wrote:
Hello,
I try to test Exported Asyncronous functions in OpenSips 2.1.
What I do to return back in stript (at line after exec) when finished
route[resume] ?
{
async( exec("test.sh.","$ru","$avp(return)"), resume );
line after exec;
}
route[resume] {
xlog("Exec return$avp(return)");
}
Best regards,
PlayMen
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Re: [OpenSIPS-Users] Opensips and CSeq number handling
Hello, Think a SIP trace would help a lot in debugging this situation - my guess is that there is sort of a "race" between the RE-INVITE being sent out, and in-dialog options pings ( eg. Re-INVITE goes out, then OpenSIPS sends in-dialog OPTION pings, then the ACK comes in and it's CSEQ is wrongly increased ). Can you please provide a SIP trace for your scenario ? Best Regards, Vlad Paiu OpenSIPS Developer On 21.10.2015 12:32, Vlad Paiu wrote: Hello, Yes, if you are using in-dialog OPTIONS pings, OpenSIPS will mangle the CSEQs of all in-dialog requests, after it has sent the first pings. Just making sure I understand the scenario... so the client is sending a Re-INVITE and immediately after that it sends an UPDATE, and the ACK for the RE-INVITE gets propagated with the CSEQ of the UPDATE ? Best Regards, Vlad Paiu OpenSIPS Developer On 20.10.2015 21:28, Patrick Wakano wrote: Hi list, An update about this issue. The behavior I mentioned about Opensips incrementing the CSeq values only after second Re-Invite is incorrect. More tests showed me that this happens after the in-dialog Options the dialog module sends (I am creating it with "pP" options). This also explains why Opensips is changing the CSeq number, because it has to couple the CSeq numbering of the local generated Options with the original requests of the dialog! The problem regarding the Ack with wrong CSeq number still occurs anyways. It seems that Opensips is not matching the Ack with the correct Invite transaction... Patrick On Tue, Oct 20, 2015 at 12:31 PM, Patrick Wakano <mailto:pwak...@gmail.com>> wrote: Hello Opensips list! Hope you all doing fine! The purpose of this e-mail is to explain a problem I am facing and to understand a little bit more about the handling done by Opensips over the CSeq number when forwarding messages to the destination. I couldn't find any real good explanation over this subject so I wrote this huge e-mail, sorry for that... I am trying to use the remote ID feature of Asterisk, but in some transfer scenarios the call gets dropped and after digging the problem I think it is related to the CSeq handling done by Opensips. This remote ID feature is configured to use the P-Asserted-Identity header to transmit the callee ID to the caller and it causes the exchange of Re-Invites and/or Updates during the call. The transfer scenario I mentioned is entirely handled by Asterisk, and as a result of the transfer it sends to Opensips the identity of the new peer, using a Re-Invite and an Update. First I would like to know how Opensips handles the CSeq number when proxying the Invite from one side to the other? My tests showed me that Opensips does not change the CSeq for the first Invite and first Re-invite, however for the second Re-invite and for requests after that it is always incrementing the value by one when forwarding it. Although it haven't caused any errors so far, I am not sure if this is correct. Why is Opensips incrementing it? My understanding is that the proxy was not supposed to change this field... Now the problem I am facing: In a blind transfer scenario, the remote ID feature causes Asterisk to send a Re-Invite and right after an Update. Opensips increments the CSeq of both(because this happens to be the second Re-Invite of the dialog) and forward them to the destination. Both messages are answered with 200 Ok. This follows by Asterisk sending an Ack with the same CSeq number used in the Re-Invite. This is the point where Opensips fails, it gets this Ack and forward it using the CSeq number of the Update and not the one of the Re-Invite. Because of this the destination discards this Ack and keeps retransmitting the 200 Ok for the Re-Invite, eventually the call is dropped by timeout or because some other Re-Invite happens without the prior one being properly handled. Useful information: - If the Re-Invite followed by the Update is the first of the dialog, then the problem does not happen. The CSeq numbers are not incremented and the CSeq for the Ack is correct. - If due to unknown timing reasons, the Update gets forwarded before the Re-Invite (even though the Re-Invite is received first) the problem also does not happen. The CSeq numbers are incremented but the CSeq for the Ack gets the correct value. So it seems to me that the Ack is getting the last CSeq used to forward, and not the one of the corresponding Invite. - When I enable more traces(debug=4), I always fall in the case where the Update is forwarded before the Re-Invite and then the problem doesn't happen. - In an attended transfer, Asterisk does not send the Update so the problem does not happen. - Not sure why Asterisk is sending the Re-Invite im
