Re: [SR-Users] DMQ Cluster - hosts loose network connectivity

[Charles Chance]

Hello José,

The issue may have been introduced by the recent multi-notify option.

To test the theory, could you try setting multi_notify to 0 and the
notification address to one of either the A or B server IP addresses?

e.g.

modparam("dmq", "notification_address", "sip:172.112.10.206:5060")
modparam("dmq", "multi_notify", 0)


If the issue is not present in this case then I will look to fix it for
multi-notify scenario.

Either way, please include the server C log for comparison.

Cheers,
Charles

On 7 Feb 2017 07:06, "José Seabra" <joseseab...@gmail.com> wrote:

> Hello Charles,
> 2 of them were active during network failure time period and at the time
> of reconnection.
>
> As the issue were noticed on Production environment i don't have enough
> logs to report you but i have  reproduced the issue on my Lab environment
> that has exactly the same DMQ configurations, except the number of dmq
> hosts that are 3 instead of 4.
>
> Steps to reproduce the issue:
>
> Start all 3 kamailio nodes.
> at this stage all of them are active in dmq.list_nodes.
> Server A
> {
> host: 172.112.10.243
> port: 5060
> resolved_ip: 172.112.10.243
> status: 2
> last_notification: 0
> local: 0
> }
> {
> host: 172.112.10.246
> port: 5060
> resolved_ip: 172.112.10.246
> status: 2
> last_notification: 0
> local: 0
> }
> {
> host: 172.112.10.207
> port: 5060
> resolved_ip: 172.112.10.207
> status: 2
> last_notification: 0
> local: 0
> }
> {
> host: 172.112.10.206
> port: 5060
> resolved_ip: 172.112.10.206
> status: 2
> last_notification: 0
> local: 1
> }
>
> Server B
> {
> host: 172.112.10.243
> port: 5060
> resolved_ip: 172.112.10.243
> status: 2
> last_notification: 0
> local: 0
> }
> {
> host: 172.112.10.206
> port: 5060
> resolved_ip: 172.112.10.206
> status: 2
> last_notification: 0
> local: 0
> }
> {
> host: 172.112.10.207
> port: 5060
> resolved_ip: 172.112.10.207
> status: 2
> last_notification: 0
> local: 1
> }
>
>
> Server C
>
> {
> host: 172.112.10.246
> port: 5060
> resolved_ip: 172.112.10.246
> status: 2
> last_notification: 0
> local: 0
> }
> {
> host: 172.112.10.207
> port: 5060
> resolved_ip: 172.112.10.207
> status: 2
> last_notification: 0
> local: 0
> }
> {
> host: 172.112.10.206
> port: 5060
> resolved_ip: 172.112.10.206
> status: 2
> last_notification: 0
> local: 0
> }
> {
> host: 172.112.10.243
> port: 5060
> resolved_ip: 172.112.10.243
> status: 2
> last_notification: 0
> local: 1
> }
>
>
> Then, after few minutes i inserted an IPTABLES rule on server C to drop
> all packages to 5060 port.
>
> After that the Server A and Server B can see each other:
>
> {
> host: 172.112.10.206
> port: 5060
> resolved_ip: 172.112.10.206
> status: 2
> last_notification: 0
> local: 0
> }
> {
> host: 172.112.10.207
> port: 5060
> resolved_ip: 172.112.10.207
> status: 2
> last_notification: 0
> local: 1
> }
>
> Server B only can see itself:
>
> {
> host: 172.112.10.243
> port: 5060
> resolved_ip: 172.112.10.243
> status: 2
> last_notification: 0
> local: 1
> }
>
> This behavior keeps the same after the network connectivity comes up.
>
>
> Please find out the log files attached on this email for each server.
>
> Server A - 172.112.10.206
> Server B - 172.112.10.207
> Server C- 172.112.10.243
>
> Let me know if do you need further information.
>
> Regards
> José
>
>
>
>
>
> 2017-02-06 14:04 GMT+00:00 Charles Chance <charles.cha...@sipcentric.com>:
>
>> Hello,
>>
>> DMQ will remove nodes from its internal list if they fail to respond to
>> its pings - with the exception of the original notification peer
>> specified in config. This way, if the network connection is lost, DMQ will
>> continue to try the original peer indefinitely until connectivity is
>> restored, and rebuild its list of other nodes from there.
>>
>> Was the original peer (or one of them if multiple defined A/SRV records)
>> still active at the time of reconnection?
>>
>> It would help to diagnose if you can you send your log from around the
>>

Re: [SR-Users] DMQ Cluster - hosts loose network connectivity

[Charles Chance]

Hello,

DMQ will remove nodes from its internal list if they fail to respond to its
pings - with the exception of the original notification peer specified in
config. This way, if the network connection is lost, DMQ will continue to
try the original peer indefinitely until connectivity is restored, and
rebuild its list of other nodes from there.

Was the original peer (or one of them if multiple defined A/SRV records)
still active at the time of reconnection?

It would help to diagnose if you can you send your log from around the time
of disconnection, and also at the time of reconnect.

Regards,

Charles


On 6 February 2017 at 10:03, José Seabra  wrote:

> Hello Daniel,
>
> The parameters that i have configured on my kamailio server are:
>
> #!ifdef ENABLE_REG_SYNC
> modparam("registrar", "sock_flag", 18)
> modparam("registrar", "sock_hdr_name", "Sock-Info")
> ### SIP registrar replication to other nodes ##
> loadmodule "dmq.so"
>
> ###  distributed message queue module paramenters #
> modparam("dmq", "server_address", "sip:MY_IP_ADDRESS:MY_PORT_ADDRESS")
> modparam("dmq", "notification_address", "DMQ_HOSTS")
> modparam("dmq", "multi_notify", 1)
> modparam("dmq", "num_workers", 4)
>
> loadmodule "dmq_usrloc.so"
> modparam("dmq_usrloc", "enable", 1)
> modparam("dmq_usrloc", "sync", 1)
> modparam("dmq_usrloc", "batch_size", DMQ_BATCH_SIZE)
> modparam("dmq_usrloc", "batch_usleep", DMQ_BATCH_USLEEP)
> #!endif
>
>
> Let me know if do you need further information's.
>
> Thank you
> Regards
> José Seabra
>
> 2017-02-06 7:01 GMT+00:00 Daniel-Constantin Mierla :
>
>> Hello,
>>
>> what are the parameters for dmq module?
>>
>> Cheers,
>> Daniel
>>
>> On 01/02/2017 11:32, José Seabra wrote:
>>
>> Hello there,
>> My DMQ cluster  has 4 nodes and by some reason 2 of them lost the network
>> connectivity for long time(~ 4 hours), after the network of these 2 nodes
>> come back they didn't get connected on DMQ  cluster automatically, i had to
>> restart kamailio to get them again on DMQ list.
>> My doubts here are:
>>
>>- Is this an expected behavior of DMQ module?
>>- Is there any way of put them again on DMQ bus without need restart
>>kamailio?
>>
>>
>> Thank you
>> Best Regards
>> José Seabra
>>
>>
>> ___
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>>
>>
>> --
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>> www.linkedin.com/in/miconda
>> Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - 
>> www.asipto.com
>> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com
>>
>>
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>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
>
> --
> Cumprimentos
> José Seabra
>
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>

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Re: [SR-Users] Upgrade from 4.3 to 4.4

[Charles Chance]

Hello,

There were minimal DB changes between 4.3 and 4.4 - if you're using one of
the affected modules then you'll need to update the schema and version as
described here:

https://www.kamailio.org/wiki/install/upgrade/4.3.x-to-4.4.0

(Also note the dialog-ng module name change.)

Other than that, upgrading should be seamless.

Cheers,
Charles

On 18 Jan 2017 20:21, "Laura"  wrote:

> Dear list,
>
> I'm planning the upgrade fromn 4.3.6 tree to 4.4.x version on Debian
> distribution.
>
> Can i run the apt-get update and upgrade without problem.. or is possible
> that i will found some error during upgrade ?
>
> Is the db schema upgraded or not from 4.3 up to 4.4 ?
>
>
> Regards
>
> Laura
>
>
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Re: [SR-Users] Kamailio - DMQ dns srv Question

[Charles Chance]

Hey Olle,

On 11 Oct 2016 10:54, "Olle E. Johansson"  wrote:
>
>
>> On 11 Oct 2016, at 09:46, José Seabra  wrote:
>>
>> Hi Charles,
>> Sorry for my late reply.
>> I have tried that parameter but seems that for the dmq FQDN kamailio
doesn't send a NAPTR query.
>> Even for the SRV query, the kamailio makes it without service
associated(_sip._udp).
>>
> We should propably define a SRV service tag for DMQ.
>

Sorry, not sure I follow you. For what purpose?

Cheers,
Charles

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Re: [SR-Users] Distributed Message Queue Module

[Charles Chance]

Hello,

That depends entirely on what you need to replicate between servers. DMQ is
just a communication channel. How you use it is up to you.

For example, if you enable dmq_usrloc, there's no need to share a location
database. Htable has an option for replicating over dmq too.

The number of direct module integrations like dmq_usrloc/htable is fairly
limited, but you can also share arbitrary data between nodes yourself from
within your config.

I think you'll need to provide a little more information about what you're
trying to achieve, however, in order for us to offer any further advice.

Cheers,

Charles

On 29 Sep 2016 08:08, "Moacir Ferreira"  wrote:

Hi,


When using the DMQueue module what is the best approach, using one SQL
server/database for each member node or having a single SQL server for all
nodes?


Thanks,

Moacir

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Re: [SR-Users] Kamailio - DMQ dns srv Question

[Charles Chance]

Hi José,

On 21 September 2016 at 18:28, José Seabra  wrote:

> Hello,
> I have a doubt related with DMQ dns behavior, I noticed that when kamailio
> starts, it tries to resolve DMQ name configured on parameter
> notification_address as the following sequence:
>
>1. SRV
>2. A
>3. 
>
>
> Isn't supposed kamailio  try first resolve the  NAPTR DMQ name, and then
> SRV?
>
>
Can you confirm you set dns_try_naptr = yes?

https://www.kamailio.org/wiki/cookbooks/4.4.x/core#dns_try_naptr

https://raw.githubusercontent.com/kamailio/kamailio/master/doc/dns.txt

Cheers,
Charles

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Re: [SR-Users] Kamailio - DMQ dns srv Question

[Charles Chance]

Hi,

On 22 September 2016 at 14:08, José Seabra <joseseab...@gmail.com> wrote:

> Hi charles,
> Inline
> Thanks
>
> 2016-09-22 13:51 GMT+01:00 Charles Chance <charles.cha...@sipcentric.com>:
>
>> Hi José,
>>
>> In your config, you need to tell DMQ about a minimum of one other node.
>> This can be in the form of an IP or FQDN - however, where the FQDN resolves
>> to a single IP, it does not make sense to perform anything other than an A
>> lookup, so you should set the parameter to 0 in this case.
>>
>> The number of nodes in your cluster is irrelevant - once started, DMQ
>> will contact the one you have told it about and learn about the others
>> automatically.
>>
>> Optionally, you can tell DMQ about multiple nodes upfront (to account for
>> instances where the first one may be down or unreachable), by giving it an
>> FQDN which resolves to multiple IP addresses - in which case you'll need to
>> set the multi_notify parameter to 1.
>>
>
>
>> >> This is the reason that I'm using multi_notify, the doubt here was why
>> it is querying dns using SRV without service specification.
>>
>
>

Understood, thanks - I misread the following...


>>>>> Another doubt is when we use a FQDN that only resolves as A record,
>>>>> would be better DMQ send only a A query or is there any reason for always
>>>>> try SRV?
>>>>>
>>>>>
...to mean a scenario in which the FQDN resolves to a single IP only. My
mistake!

I'll take a look into adding service specification later today.

Cheers,
Charles

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Re: [SR-Users] Kamailio - DMQ dns srv Question

[Charles Chance]

Hi José,

In your config, you need to tell DMQ about a minimum of one other node.
This can be in the form of an IP or FQDN - however, where the FQDN resolves
to a single IP, it does not make sense to perform anything other than an A
lookup, so you should set the parameter to 0 in this case.

The number of nodes in your cluster is irrelevant - once started, DMQ will
contact the one you have told it about and learn about the others
automatically.

Optionally, you can tell DMQ about multiple nodes upfront (to account for
instances where the first one may be down or unreachable), by giving it an
FQDN which resolves to multiple IP addresses - in which case you'll need to
set the multi_notify parameter to 1.

Does that help?

Cheers,
Charles


On 22 September 2016 at 13:36, José Seabra <joseseab...@gmail.com> wrote:

> Hi Charles
> I have modparam("dmq", "multi_notify", 1) because my A record resolves
> more than one Kamailio addresses, and if i understood well the
> documentation says that we need set it to a value different of 0 to resolve
> all ip addresses associated to the Name configured on DMQ.
> I'm using DMQ with 4 kamailio instances, so i think that i need have it
> set to a value different than 0, or i'm wrong?.
>
> Documentation says:
>
> Enables the ability to resolve multiple IPv4/IPv6 addresses for a single
> notification address.
>
> A value of zero resolves to the first IP address found. A non-zero value
> resolves to all IP addresses associated with the host. This includes
> addresses from DNS SRV records, A and  records.
>
> *Default value is “0”.*
>
> Thank you
> Cheers
> José
>
>
>
> 2016-09-22 13:28 GMT+01:00 Charles Chance <charles.cha...@sipcentric.com>:
>
>> Hey,
>>
>> On 22 September 2016 at 12:18, José Seabra <joseseab...@gmail.com> wrote:
>>
>>> Hello Daniel and Charles,
>>> Thank you for your feedback.
>>> Another doubt is when we use a FQDN that only resolves as A record,
>>> would be better DMQ send only a A query or is there any reason for always
>>> try SRV?
>>>
>>>
>> As far as I recall (it's been a while since I did anything with this),
>> the default behaviour is to query A record only. There is a mod_param to
>> enable multi-record/SRV lookup:
>>
>> modparam("dmq", "multi_notify", 1)
>>
>> But the default is off.
>>
>> Is this not the behaviour you are seeing?
>>
>> Cheers,
>> Charles
>>
>>
>>
>>
>>> Thank you for your great job.
>>>
>>> BR
>>> José Seabra
>>>
>>> 2016-09-22 8:33 GMT+01:00 Charles Chance <charles.cha...@sipcentric.com>
>>> :
>>>
>>>> Hello,
>>>>
>>>> I can take a look today.
>>>>
>>>> Cheers,
>>>> Charles
>>>>
>>>> On 22 Sep 2016 06:20, "Daniel-Constantin Mierla" <mico...@gmail.com>
>>>> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> I guess that the one who did the implementation of the srv query for
>>>>> dmq hasn't "allocated" any service name. I haven't looked at the sources,
>>>>> but I guess it should be a small patch to compose the srv dns string to be
>>>>> queried -- maybe that part can be made a mod param so each can choose its
>>>>> preferred service.
>>>>>
>>>>> Cheers,
>>>>> Daniel
>>>>>
>>>>> On 21/09/16 19:28, José Seabra wrote:
>>>>>
>>>>> Hello,
>>>>> I have a doubt related with DMQ dns behavior, I noticed that when
>>>>> kamailio starts, it tries to resolve DMQ name configured on parameter
>>>>> notification_address as the following sequence:
>>>>>
>>>>>1. SRV
>>>>>2. A
>>>>>3. 
>>>>>
>>>>>
>>>>> Isn't supposed kamailio  try first resolve the  NAPTR DMQ name, and
>>>>> then SRV?
>>>>> I'm asking this because kamailio is trying resolve the SRV record
>>>>> without any transport protocol specified on query, as my dns server only
>>>>> accepts queries on the format "_Service._Proto.Name" the SRV will never be
>>>>> resolved.
>>>>>
>>>>> Thank you.
>>>>> BR
>>>>> --
>>>>> José Seabra
>>>>>
>>>>>
>>>>> ___

Re: [SR-Users] Kamailio - DMQ dns srv Question

[Charles Chance]

Hey,

On 22 September 2016 at 12:18, José Seabra <joseseab...@gmail.com> wrote:

> Hello Daniel and Charles,
> Thank you for your feedback.
> Another doubt is when we use a FQDN that only resolves as A record, would
> be better DMQ send only a A query or is there any reason for always try SRV?
>
>
As far as I recall (it's been a while since I did anything with this), the
default behaviour is to query A record only. There is a mod_param to enable
multi-record/SRV lookup:

modparam("dmq", "multi_notify", 1)

But the default is off.

Is this not the behaviour you are seeing?

Cheers,
Charles




> Thank you for your great job.
>
> BR
> José Seabra
>
> 2016-09-22 8:33 GMT+01:00 Charles Chance <charles.cha...@sipcentric.com>:
>
>> Hello,
>>
>> I can take a look today.
>>
>> Cheers,
>> Charles
>>
>> On 22 Sep 2016 06:20, "Daniel-Constantin Mierla" <mico...@gmail.com>
>> wrote:
>>
>>> Hello,
>>>
>>> I guess that the one who did the implementation of the srv query for dmq
>>> hasn't "allocated" any service name. I haven't looked at the sources, but I
>>> guess it should be a small patch to compose the srv dns string to be
>>> queried -- maybe that part can be made a mod param so each can choose its
>>> preferred service.
>>>
>>> Cheers,
>>> Daniel
>>>
>>> On 21/09/16 19:28, José Seabra wrote:
>>>
>>> Hello,
>>> I have a doubt related with DMQ dns behavior, I noticed that when
>>> kamailio starts, it tries to resolve DMQ name configured on parameter
>>> notification_address as the following sequence:
>>>
>>>1. SRV
>>>2. A
>>>3. 
>>>
>>>
>>> Isn't supposed kamailio  try first resolve the  NAPTR DMQ name, and then
>>> SRV?
>>> I'm asking this because kamailio is trying resolve the SRV record
>>> without any transport protocol specified on query, as my dns server only
>>> accepts queries on the format "_Service._Proto.Name" the SRV will never be
>>> resolved.
>>>
>>> Thank you.
>>> BR
>>> --
>>> José Seabra
>>>
>>>
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>>>
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>>> http://www.linkedin.com/in/miconda
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>>>
>>>
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>>>
>> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered
>> office: Faraday Wharf, Innovation Birmingham Campus, Holt Street,
>> Birmingham Science Park, Birmingham B7 4BB.
>>
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>>
>
>
> --
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> José Seabra
>
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Re: [SR-Users] Kamailio - DMQ dns srv Question

[Charles Chance]

Hello,

I can take a look today.

Cheers,
Charles

On 22 Sep 2016 06:20, "Daniel-Constantin Mierla"  wrote:

> Hello,
>
> I guess that the one who did the implementation of the srv query for dmq
> hasn't "allocated" any service name. I haven't looked at the sources, but I
> guess it should be a small patch to compose the srv dns string to be
> queried -- maybe that part can be made a mod param so each can choose its
> preferred service.
>
> Cheers,
> Daniel
>
> On 21/09/16 19:28, José Seabra wrote:
>
> Hello,
> I have a doubt related with DMQ dns behavior, I noticed that when kamailio
> starts, it tries to resolve DMQ name configured on parameter
> notification_address as the following sequence:
>
>1. SRV
>2. A
>3. 
>
>
> Isn't supposed kamailio  try first resolve the  NAPTR DMQ name, and then
> SRV?
> I'm asking this because kamailio is trying resolve the SRV record without
> any transport protocol specified on query, as my dns server only accepts
> queries on the format "_Service._Proto.Name" the SRV will never be resolved.
>
> Thank you.
> BR
> --
> José Seabra
>
>
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>
> --
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> http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com
>
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Re: [SR-Users] using DMQ , Kamailio shutsdown

[Charles Chance]

cket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list
>
> Mar 25 06:03:09 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG:
> websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list
> [(nil)] with [0] members
>
> Mar 25 06:03:10 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG:
> websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list
>
> Mar 25 06:03:10 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG:
> websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list
> [(nil)] with [0] members
>
> Mar 25 06:03:11 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG:
> websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list
>
> Mar 25 06:03:11 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG:
> websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list
> [(nil)] with [0] members
>
> Mar 25 06:03:12 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG:
> websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list
>
> Mar 25 06:03:12 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG:
> websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list
> [(nil)] with [0] members
>
> Mar 25 06:03:13 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG:
> websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list
>
> Mar 25 06:03:13 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG:
> websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list
> [(nil)] with [0] members
>
> Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG:
> websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list
>
> Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG:
> websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list
> [(nil)] with [0] members
>
> Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14041]: CRITICAL:
>  [pass_fd.c:275]: receive_fd(): EOF on 21
>
> Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG:
>  [tcp_main.c:3448]: handle_ser_child(): dead child 14, pid 14031
> (shutting down?)
>
> Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG:
>  [io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa31e60, 21,
> -1, 0x0) fd_no=26 called
>
> Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: ALERT:
>  [main.c:739]: handle_sigs(): child process 14031 exited by a signal
> 11
>
> Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: ALERT:
>  [main.c:742]: handle_sigs(): core was generated
>
> Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: INFO:
>  [main.c:754]: handle_sigs(): terminating due to SIGCHLD
>
> Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: DEBUG:
>  [main.c:756]: handle_sigs(): terminating due to SIGCHLD
>
> Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14038]: INFO:
>  [main.c:809]: sig_usr(): signal 15 received
>
> Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14040]: INFO:
>  [main.c:809]: sig_usr(): signal 15 received
>
> Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14037]: INFO:
>  [main.c:809]: sig_usr(): signal 15 received
>
> Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14035]: INFO:
>  [main.c:809]: sig_usr(): signal 15 received
>
> There were no UAs that tried to register while this error occured.
>
> Thanking You,
> Sunil More
> Ph : 9503338275
>
> On Sat, Mar 26, 2016 at 1:26 PM, Charles Chance <
> charles.cha...@sipcentric.com> wrote:
>
>> Hello,
>>
>> Which version are you running?
>>
>> Can you provide a gdb backtrace?
>>
>> Cheers,
>> Charles
>> Hello All,
>>
>> I am running two instances of kamailio and using dmq.
>>
>> Kamailio starts good. However one of the node shutsdown unexpectedly.
>> Unsure what is going wrong. Please let me know what could be wrong. below
>> is a the conf detail.
>>
>> loadmodule "dmq.so"
>> modparam("dmq", "server_address", "sip:sunilmore.in:5060")
>> modparam("dmq", "notification_address", "sip:sunilmore.in:5064")
>>
>>
>> loadmodule "usrloc.so"
>> modparam("usrloc", "db_mode", 2)
>> modparam("usrloc", "db_url", DBURL2)
>> modparam("usrloc", "use_domain", 1)
>> modparam("usrloc", "matching_mode", 0)
>> modparam("usrloc", "desc_time_order", 1)
>> modparam("usrloc", "nat_bflag", FLB_NATB)
>> modparam("usrloc", "db_insert_null", 1)
>>
>&g

Re: [SR-Users] DMQ question

[Charles Chance]

I'm not aware of such mechanism existing already, although I may have
overlooked something. It's likely easier though in my opinion, and more
beneficial to the project overall, to add replication directly within the
tm module - similar to that which exists in htable, with a flag to
enable/disable it.

Seems to me the most appropriate solution all round.

Alternatively, if a function is made available to update transaction state
manually on the receiving node, you may then be able to use
dmq_bcast_message() to relay the information yourself with some custom
payload.

Best,
Charles
On 3 Mar 2016 7:00 pm, "Alex Balashov" <abalas...@evaristesys.com> wrote:

> On 03/03/2016 01:58 PM, Charles Chance wrote:
>
>> Hi Alex,
>>
>> The function was primarily intended for onward replication of successful
>> REGISTERs, rather than transactional state.
>>
>> Assuming the function was available for replies, how do you anticipate
>> handling them on the standby node(s)?
>>
>
> Some mechanism would be needed to drop them, e.g. in a stateless default
> onreply_route.
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 303 Perimeter Center North, Suite 300
> Atlanta, GA 30346
> United States
>
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
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Re: [SR-Users] DMQ question

[Charles Chance]

Hi Alex,

The function was primarily intended for onward replication of successful
REGISTERs, rather than transactional state.

Assuming the function was available for replies, how do you anticipate
handling them on the standby node(s)?

Cheers,
Charles
On 3 Mar 2016 17:33, "Alex Balashov"  wrote:

> Charles,
>
> If replication for replies is not available, how would a replicated
> transaction transition through its life cycle on a standby cluster node?
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 303 Perimeter Center North, Suite 300
> Atlanta, GA 30346
> United States
>
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
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>
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Re: [SR-Users] DMQ question

[Charles Chance]

The function is only available to request route right now - it's been a
while so I don't recall the exact reasons behind it, however the same
restriction applies to t_replicate().

I can take a closer look later and let you know for sure.

Charles

On 3 March 2016 at 16:31, Alex Balashov <abalas...@evaristesys.com> wrote:

> On 03/03/2016 11:29 AM, Charles Chance wrote:
>
>> In the first instance, you could use dmq_is_from_node() to determine if
>> the message is a replicated one and if so, don't relay.
>>
>> There's no specific function to check the state of other nodes, although
>> it would be a simple addition. If I understand your question correctly,
>> however, if the message has just been replicated by the primary server
>> then it's safe to assume it's in service, right?
>>
>
> I would think in almost all imaginable modes of operation, yes.
>
> How are replies handled? Can these be dmq_t_replicated() too, so that the
> transaction is properly aged through its life cycle on the other nodes, too?
>
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 303 Perimeter Center North, Suite 300
> Atlanta, GA 30346
> United States
>
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
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>
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Re: [SR-Users] DMQ question

[Charles Chance]

In the first instance, you could use dmq_is_from_node() to determine if the
message is a replicated one and if so, don't relay.

There's no specific function to check the state of other nodes, although it
would be a simple addition. If I understand your question correctly,
however, if the message has just been replicated by the primary server then
it's safe to assume it's in service, right?

Cheers,
Charles


On 3 March 2016 at 16:14, Alex Balashov <abalas...@evaristesys.com> wrote:

> On 03/03/2016 11:09 AM, Charles Chance wrote:
>
> Yes, you are correct. It simply wraps the standard t_replicate()
>> function, replicating the original request to every other node (first
>> appending a new branch for each). Use case is essentially the same as
>> the original but having the benefit of not having to define
>> destination(s) statically in config, since that part is handled by DMQ.
>>
>
> I see! Thank you.
>
> What I've got seems like a good use-case for clustering over DMQ as a
> transport: a number of Kamailio servers that all answer on the same SIP
> domain, and a nondeterministic Layer 3 routing topology where some replies
> occasionally traverse a different Kamailio server to the one that processed
> the request to which they correspond. For a wide variety of reasons, I need
> to handle all this statefully, not statelessly.
>
> So, it seems like I ought to be able to dmq_t_replicate() incoming INVITEs
> to other nodes and, on those nodes, instantiate the transaction with
> t_newtran(). But one would also need to devise a mechanism to inform those
> secondary nodes that the message should not be t_relay()'d unless the
> primary server for which it was intended (itself difficult to identify due
> to same SIP domain everywhere) was determined to be out of service. I don't
> suppose there's some clever shortcut to implementing that kind of
> conditional logic manually?
>
>
> -- Alex
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 303 Perimeter Center North, Suite 300
> Atlanta, GA 30346
> United States
>
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
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>
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Re: [SR-Users] DMQ question

[Charles Chance]

Hi Alex,

Yes, you are correct. It simply wraps the standard t_replicate() function,
replicating the original request to every other node (first appending a new
branch for each). Use case is essentially the same as the original but
having the benefit of not having to define destination(s) statically in
config, since that part is handled by DMQ.

Cheers,
Charles


On 3 March 2016 at 15:37, Alex Balashov  wrote:

> Hi,
>
> If I understand the DMQ documentation correctly, when a SIP request is
> distributed to other Kamailio instances via dmq_t_replicate(), it is not
> merely sent as a payload in a novel way, but actually "rehydrated" by the
> receiver into a mock request on the other end and processed as through it
> were a "real" message that came into the request_route[] in the natural way.
>
> Is this an accurate assumption?
>
> -- Alex
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 303 Perimeter Center North, Suite 300
> Atlanta, GA 30346
> United States
>
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
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Re: [SR-Users] Releasing Kamailio v4.3.2

[Charles Chance]

Hi Daniel,

Do you have a moment to comment on this related question:
http://lists.sip-router.org/pipermail/sr-dev/2015-September/030663.html?

Cheers,
Charles
On 10 Sep 2015 08:21, "Daniel-Constantin Mierla"  wrote:

> Hello,
>
> short note to remind that today kamailio v4.3.1 will be released. If
> there is anything to push to branch 4.3, write first to sr-dev -- once
> the announcement is done, commits to branch 4.3 can go as usual.
>
> Cheers,
> Daniel
>
> On 07/09/15 10:27, Daniel-Constantin Mierla wrote:
> > Hello,
> >
> > I am considering to release a new version from branch 4.3 later this
> > week, on Thursday, September 10. If there are issues not yet reported,
> > then use github tracker to do it:
> >
> >   - https://github.com/kamailio/kamailio/issues
> >
> > Cheers,
> > Daniel
> >
>
> --
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Book: SIP Routing With Kamailio - http://www.asipto.com
> Kamailio Advanced Training, Sep 28-30, 2015, in Berlin -
> http://asipto.com/u/kat
>
>
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 ***

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Re: [SR-Users] Releasing Kamailio v4.3.2

[Charles Chance]

Of course, sorry - the commit is here:
https://github.com/kamailio/kamailio/commit/edc25d7373f9560d09a35640ab19741f19b97040
.

Will also require the second part of:
https://github.com/kamailio/kamailio/commit/c7f94fd94500270828fcc4cb3011c60c67a8936b
.

It is implementing the replication of usrloc "delete" event - which appears
to have been allowed for in the original code, but never completed.

Cheers,
Charles


On 10 September 2015 at 11:43, Daniel-Constantin Mierla <mico...@gmail.com>
wrote:

> Hi Charles,
>
> can you send quickly what is the commit id in question? Being caught on
> some other backports, it will take a while to read the whole thread and
> figure out the details -- sometime looking at code is faster to decide is a
> new feature or not.
>
> Cheers,
> Daniel
>
> On 10/09/15 11:26, Charles Chance wrote:
>
> Hi Daniel,
>
> Do you have a moment to comment on this related question:
> http://lists.sip-router.org/pipermail/sr-dev/2015-September/030663.html?
>
> Cheers,
> Charles
> On 10 Sep 2015 08:21, "Daniel-Constantin Mierla" <mico...@gmail.com>
> wrote:
>
>> Hello,
>>
>> short note to remind that today kamailio v4.3.1 will be released. If
>> there is anything to push to branch 4.3, write first to sr-dev -- once
>> the announcement is done, commits to branch 4.3 can go as usual.
>>
>> Cheers,
>> Daniel
>>
>> On 07/09/15 10:27, Daniel-Constantin Mierla wrote:
>> > Hello,
>> >
>> > I am considering to release a new version from branch 4.3 later this
>> > week, on Thursday, September 10. If there are issues not yet reported,
>> > then use github tracker to do it:
>> >
>> >   - https://github.com/kamailio/kamailio/issues
>> >
>> > Cheers,
>> > Daniel
>> >
>>
>> --
>> Daniel-Constantin Mierla
>> http://twitter.com/#!/miconda - <http://www.linkedin.com/in/miconda>
>> http://www.linkedin.com/in/miconda
>> Book: SIP Routing With Kamailio - http://www.asipto.com
>> Kamailio Advanced Training, Sep 28-30, 2015, in Berlin -
>> <http://asipto.com/u/kat>http://asipto.com/u/kat
>>
>>
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>>
>
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> <http://www.sipcentric.com/2015/07/were-rebranding/> ***
>
> www.sipcentric.com
>
> Follow us on twitter @sipcentric <http://twitter.com/sipcentric>
>
> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered
> office: Faraday Wharf, Innovation Birmingham Campus, Holt Street,
> Birmingham Science Park, Birmingham B7 4BB.
>
>
> --
> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
> http://www.linkedin.com/in/miconda
> Book: SIP Routing With Kamailio - http://www.asipto.com
> Kamailio Advanced Training, Sep 28-30, 2015, in Berlin - 
> http://asipto.com/u/kat
>
>


-- 
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Managing Director

t. 0121 285 4400m. 07932 063 891

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Re: [SR-Users] Releasing Kamailio v4.3.2

[Charles Chance]

Thanks Daniel - will be back to a desk in the next 30 minutes so should be
enough time to get it in before packaging begins.

Cheers,
Charles
On 10 Sep 2015 11:54, "Daniel-Constantin Mierla" <mico...@gmail.com> wrote:

> Btw, if your decision is to backport, go ahead, packaging of 4.3.2 likely
> to start in one hour or so.
>
> Cheers,
> Daniel
>
> On 10/09/15 12:53, Daniel-Constantin Mierla wrote:
>
> It doesn't change the config or database to have a deeper investigation,
> therefore if it solving something that currently doesn't work properly,
> then all is ok to backport. To me it looks like making that dmq location
> replication work, because removing a contact is part of the normal
> registration processing, so personally I am fine backporting.
>
> Cheers,
> Daniel
>
> On 10/09/15 12:49, Charles Chance wrote:
>
> Of course, sorry - the commit is here:
> https://github.com/kamailio/kamailio/commit/edc25d7373f9560d09a35640ab19741f19b97040.
>
>
> Will also require the second part of:
> https://github.com/kamailio/kamailio/commit/c7f94fd94500270828fcc4cb3011c60c67a8936b
> .
>
> It is implementing the replication of usrloc "delete" event - which
> appears to have been allowed for in the original code, but never completed.
>
> Cheers,
> Charles
>
>
> On 10 September 2015 at 11:43, Daniel-Constantin Mierla <mico...@gmail.com
> > wrote:
>
>> Hi Charles,
>>
>> can you send quickly what is the commit id in question? Being caught on
>> some other backports, it will take a while to read the whole thread and
>> figure out the details -- sometime looking at code is faster to decide is a
>> new feature or not.
>>
>> Cheers,
>> Daniel
>>
>> On 10/09/15 11:26, Charles Chance wrote:
>>
>> Hi Daniel,
>>
>> Do you have a moment to comment on this related question:
>> http://lists.sip-router.org/pipermail/sr-dev/2015-September/030663.html?
>>
>> Cheers,
>> Charles
>> On 10 Sep 2015 08:21, "Daniel-Constantin Mierla" <mico...@gmail.com>
>> wrote:
>>
>>> Hello,
>>>
>>> short note to remind that today kamailio v4.3.1 will be released. If
>>> there is anything to push to branch 4.3, write first to sr-dev -- once
>>> the announcement is done, commits to branch 4.3 can go as usual.
>>>
>>> Cheers,
>>> Daniel
>>>
>>> On 07/09/15 10:27, Daniel-Constantin Mierla wrote:
>>> > Hello,
>>> >
>>> > I am considering to release a new version from branch 4.3 later this
>>> > week, on Thursday, September 10. If there are issues not yet reported,
>>> > then use github tracker to do it:
>>> >
>>> >   - https://github.com/kamailio/kamailio/issues
>>> >
>>> > Cheers,
>>> > Daniel
>>> >
>>>
>>> --
>>> Daniel-Constantin Mierla
>>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>> Book: SIP Routing With Kamailio - http://www.asipto.com
>>> Kamailio Advanced Training, Sep 28-30, 2015, in Berlin -
>>> http://asipto.com/u/kat
>>>
>>>
>>> ___
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users@lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>> *** We're rebranding! Learn more here
>> <http://www.sipcentric.com/2015/07/were-rebranding/> ***
>>
>> www.sipcentric.com
>>
>> Follow us on twitter @sipcentric <http://twitter.com/sipcentric>
>>
>> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered
>> office: Faraday Wharf, Innovation Birmingham Campus, Holt Street,
>> Birmingham Science Park, Birmingham B7 4BB.
>>
>>
>> --
>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
>> http://www.linkedin.com/in/miconda
>> Book: SIP Routing With Kamailio - http://www.asipto.com
>> Kamailio Advanced Training, Sep 28-30, 2015, in Berlin - 
>> http://asipto.com/u/kat
>>
>>
>
>
> --
> *Charles Chance*
> Managing Director
>
> t. 0121 285 4400m. 07932 063 891
>
> *** We're rebranding! Learn more here
> <http://www.sipcentric.com/2015/07/were-rebranding/> ***
>
> www.sipcentric.com
>
> Follow us on twitter @sipcentric <http://twitter.com/sipcentric>
>
> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered
> office: Faraday Wha

Re: [SR-Users] dmq_usrloc

[Charles Chance]

This is a different crash to before and is on the receiving node this time.

I have just pushed another fix for this one. Please test again and let me
know what happens this time.

Cheers,
Charles
On 28 Aug 2015 07:23, Kelvin Chua kel...@gmail.com wrote:

 unfortunately it is crashing with current master

 i do not see this on my logs

 if (dmq_ul.get_udomain(location, _d)  0) {
 LM_ERR(Failed to get domain\n);
 goto done;
 }

 here is a backtrace

 #0  0x7f23dc6d92cc in get_urecord (_d=0x0, _aor=0x7ffc8b14cdd0,
 _r=0x7ffc8b14ce18) at udomain.c:1134
 sl = 32547
 i = 32547
 aorhash = 1712986268
 r = 0x7ffc0008
 #1  0x7f23dc49d5c8 in add_contact (aor=..., ci=0x7f23dc6ac2c0 ci) at
 usrloc_sync.c:55
 r = 0x7f23dd96f452 parse_value+371
 _d = 0x0
 c = 0x7f235390dd93
 contact = {s = 0xb93de0 , len = -1961570368}
 res = 1402003237
 __FUNCTION__ = add_contact
 #2  0x7f23dc4a47e6 in usrloc_dmq_handle_msg (msg=0x7f235390d508,
 resp=0x7ffc8b14d0a0, node=0x7f2353908410) at usrloc_sync.c:384
 content_length = 403
 body = {
   s = 0x7f235390dd93
 {\action\:1,\aor\:\kelvin\,\ruid\:\uloc-55dffbe9-258b-1\,\c\:\sip:kelvin@192.168.1.120:5060;alias=4x.1x8.62.213~5060~1\,\received\:\\,\path\:\sip:6x.6x.174.22:6055;lr;received=sip:4x.1x8.62.213:5060...,
 len = 403}
 jdoc = {root = 0xb93de0, flags = 0, buf = {
 s = 0x7f235390dd93
 {\action\:1,\aor\:\kelvin\,\ruid\:\uloc-55dffbe9-258b-1\,\c\:\sip:kelvin@192.168.1.120:5060;alias=4x.1x8.62.213~5060~1\,\received\:\\,\path\:\sip:6x.6x.174.22:6055;lr;received=sip:4x.1x8.62.213:5060...,
 len = 403}, malloc_fn = 0x7f23f082a750 __GI___libc_malloc, free_fn =
 0x7f23f082adf0 __GI___libc_free}
 it = 0x0
 ci = {ruid = {s = 0xb92a10 uloc-55dffbe9-258b-1, len = 20}, c =
 0x7ffc8b14cf60, received = {s = 0xb92b60 , len = 0}, path =
 0x7ffc8b14cf80, expires = 1440745337, q = -1,
   callid = 0x7ffc8b14cf90, cseq = 2, flags = 2, cflags = 0,
 user_agent = 0x7ffc8b14cfa0, sock = 0x0, methods = 2147483648, instance =
 {s = 0xba7fe0 , len = 0}, reg_id = 0,
   server_id = 0, tcpconn_id = -1, keepalive = 0, xavp = 0x0,
 last_modified = 1440742397}
 action = 1
 expires = 1440745337
 cseq = 2
 flags = 0
 cflags = 0
 q = -1
 last_modified = 1440742397
 methods = -2147483648
 reg_id = 0
 aor = {s = 0xb92550 kelvin, len = 6}
 ruid = {s = 0xb92a10 uloc-55dffbe9-258b-1, len = 20}
 c = {s = 0xb92aa0 
 sip:kelvin@192.168.1.120:5060;alias=4x.1x8.62.213~5060~1,
 len = 56}
 received = {s = 0xb92b60 , len = 0}
 path = {s = 0xb92bf0
 sip:6x.6x.174.22:6055;lr;received=sip:4x.1x8.62.213:5060, len = 58}
 callid = {s = 0xb92cb0 157459181@192.168.1.120, len = 23}
 user_agent = {s = 0xb92d40 qutecom/rev-g-trunk, len = 19}
 instance = {s = 0xba7fe0 , len = 0}
 __FUNCTION__ = usrloc_dmq_handle_msg


 Kelvin Chua

 On Tue, Aug 25, 2015 at 1:27 AM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hi,

 I have pushed a change to master which should prevent the crash.

 You'll still need (for now) to use location in save/lookup - I will add
 multi-domain replication later but there needs to be some enhancement to
 the usrloc API first.

 Please confirm this fixes the crash and I will backport to 4.3.

 Cheers,
 Charles


 On 18 August 2015 at 06:00, Kelvin Chua kel...@gmail.com wrote:

 yes, same crash

 Kelvin Chua

 On Mon, Aug 17, 2015 at 3:17 PM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 There were some issues fixed since 4.3.1 - have you tried with latest
 master?

 Cheers,
 Charles
 On 17 Aug 2015 04:50, Kelvin Chua kel...@gmail.com wrote:

 i think we should support custom location tables in the future but
 either way,
 i tried changing location to kam_location in 2 instances
 on usrloc_sync.c -- crash
 i tried changing save(location) -- crash


 Kelvin Chua

 On Fri, Aug 14, 2015 at 7:32 PM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hi,

 I'm not sure why, but the dmq_usrloc module was written with the
 usrloc domain hardcoded as location.

 I intend to address that along with some other improvements but in
 the meantime, try changing your call to save() to use location instead 
 of
 kam_location...

 save(location);

 Please report back the result.

 Cheers,
 Charles
 On 14 Aug 2015 11:03, Kelvin Chua kel...@gmail.com wrote:

 4.3.1

 kamctl ul show shows blank on the other machine. the registrar is ok.

 here is the DMQ message captured using ngrep

 KDMQ sip:usrloc@1.2.3.4:5090 SIP/2.0.
 Via: SIP/2.0/UDP 1.2.3.4:5080
 ;branch=z9hG4bK97d1.f42de484.0.
 To: sip:usrloc@1.2.3.4:5090.
 From: sip:usrloc@1.2.3.4:5080
 ;tag=c5ea0db92ae4ac0758f8c2b12d8addf4-4d17.
 CSeq: 10 KDMQ.
 Call-ID: 40be1e6c728bfe96-27604@1.2.3.4.
 Content-Length: 403.
 User-Agent

Re: [SR-Users] dmq_usrloc

[Charles Chance]

Hi,

I have pushed a change to master which should prevent the crash.

You'll still need (for now) to use location in save/lookup - I will add
multi-domain replication later but there needs to be some enhancement to
the usrloc API first.

Please confirm this fixes the crash and I will backport to 4.3.

Cheers,
Charles


On 18 August 2015 at 06:00, Kelvin Chua kel...@gmail.com wrote:

 yes, same crash

 Kelvin Chua

 On Mon, Aug 17, 2015 at 3:17 PM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 There were some issues fixed since 4.3.1 - have you tried with latest
 master?

 Cheers,
 Charles
 On 17 Aug 2015 04:50, Kelvin Chua kel...@gmail.com wrote:

 i think we should support custom location tables in the future but
 either way,
 i tried changing location to kam_location in 2 instances
 on usrloc_sync.c -- crash
 i tried changing save(location) -- crash


 Kelvin Chua

 On Fri, Aug 14, 2015 at 7:32 PM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hi,

 I'm not sure why, but the dmq_usrloc module was written with the usrloc
 domain hardcoded as location.

 I intend to address that along with some other improvements but in the
 meantime, try changing your call to save() to use location instead of
 kam_location...

 save(location);

 Please report back the result.

 Cheers,
 Charles
 On 14 Aug 2015 11:03, Kelvin Chua kel...@gmail.com wrote:

 4.3.1

 kamctl ul show shows blank on the other machine. the registrar is ok.

 here is the DMQ message captured using ngrep

 KDMQ sip:usrloc@1.2.3.4:5090 SIP/2.0.
 Via: SIP/2.0/UDP 1.2.3.4:5080
 ;branch=z9hG4bK97d1.f42de484.0.
 To: sip:usrloc@1.2.3.4:5090.
 From: sip:usrloc@1.2.3.4:5080
 ;tag=c5ea0db92ae4ac0758f8c2b12d8addf4-4d17.
 CSeq: 10 KDMQ.
 Call-ID: 40be1e6c728bfe96-27604@1.2.3.4.
 Content-Length: 403.
 User-Agent: Sipsurge Registrar/1.0.0.
 Max-Forwards: 1.
 Content-Type: application/json.
 .

 {action:1,aor:kelvin,ruid:uloc-55cdbac7-6bd4-1,c:sip:kelvin@192.168.1.120:5060
 ;alias=2.4.6.8~5060~1,received:,path:sip:1.2.3.4:6055
 ;lr;received=sip:2.4.6.8:5060,callid:559751939@192.168.1.120
 ,user_agent:qutecom/rev-g-trunk,instance:,expires:1439549023,cseq:2,flags:0,cflags:0,q:-1,last_modified:1439546083,methods:4294967295,reg_id:0}


 here are the modparams

 modparam(usrloc, db_mode, 0)
 modparam(usrloc, use_domain, MULTIDOMAIN)
 modparam(dmq, server_address, DMQ_INT)
 modparam(dmq, notification_address, DMQ_EXT)
 modparam(dmq_usrloc, enable, 1)


 sometimes, the registrar crashes, sometimes, the backup, sometimes
 both but i manage to catch kamctl ul show and this is how it looks like:

 Domain:: kam_location table=1024 records=1 max_slot=1
 AOR:: kelvin
 Contact:: 
 sip:kelvin@192.168.1.120:5060;alias=2.4.6.8~5060~1
 Q=
 Expires:: deleted
 Callid:: 559751939@192.168.1.120
 Cseq:: 30
 User-agent:: qutecom/rev-g-trunk
 Path:: sip:1.2.3.4:6055;lr;received=sip:
 2.4.6.8:5060
 State:: CS_DIRTY
 Flags:: 0
 Cflag:: 0
 Socket:: udp:1.2.3.4:5080
 Methods:: 4294967295
 Ruid:: uloc-55cdbac7-6bd4-1
 Reg-Id:: 0
 Last-Keepalive:: 1439546340
 Last-Modified:: 1439546340


 Kelvin Chua

 On Thu, Aug 13, 2015 at 7:54 PM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hello,

 From which version has this been generated?

 Also, can you provide the output of kamctl ul show and the relevant
 modparam sections of your config (usrloc, dmq, dmq_usrloc).

 Cheers,
 Charles


 On 13 August 2015 at 10:23, Kelvin Chua kel...@gmail.com wrote:

 I don't know if this is related, this happens on 4.3.1 as well but
 usrloc crashes once it gets a DMQ

 #0  0x7f2a82a5727c in get_urecord_by_ruid (_d=0x0,
 _aorhash=8118438, _ruid=0x7ffe6028d5c0, _r=0x7ffe6028d568,
 _c=0x7ffe6028d578)
 at udomain.c:1153
 #1  0x7f2a8282029a in usrloc_get_all_ucontact
 (node=0x7f2a01790998) at usrloc_sync.c:166
 #2  0x7f2a828240a7 in usrloc_dmq_handle_msg (msg=0x7f2a01790f78,
 resp=0x7ffe6028d820, node=0x7f2a01790998) at usrloc_sync.c:363
 #3  0x7f2a83959e6f in worker_loop (id=1) at worker.c:105
 #4  0x7f2a8393e566 in child_init (rank=0) at dmq.c:293
 #5  0x00599658 in init_mod_child (m=0x7f2a86955c78, rank=0)
 at sr_module.c:898
 #6  0x00599390 in init_mod_child (m=0x7f2a86956098, rank=0)
 at sr_module.c:895
 #7  0x00599390 in init_mod_child (m=0x7f2a86956a78, rank=0)
 at sr_module.c:895
 #8  0x00599390 in init_mod_child (m=0x7f2a86956e68, rank=0)
 at sr_module.c:895
 #9  0x00599390 in init_mod_child (m=0x7f2a869570f8, rank=0)
 at sr_module.c:895
 #10 0x00599390 in init_mod_child (m=0x7f2a86957498, rank=0)
 at sr_module.c:895
 #11

Re: [SR-Users] dmq_usrloc

[Charles Chance]

There were some issues fixed since 4.3.1 - have you tried with latest
master?

Cheers,
Charles
On 17 Aug 2015 04:50, Kelvin Chua kel...@gmail.com wrote:

 i think we should support custom location tables in the future but either
 way,
 i tried changing location to kam_location in 2 instances on usrloc_sync.c
 -- crash
 i tried changing save(location) -- crash


 Kelvin Chua

 On Fri, Aug 14, 2015 at 7:32 PM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hi,

 I'm not sure why, but the dmq_usrloc module was written with the usrloc
 domain hardcoded as location.

 I intend to address that along with some other improvements but in the
 meantime, try changing your call to save() to use location instead of
 kam_location...

 save(location);

 Please report back the result.

 Cheers,
 Charles
 On 14 Aug 2015 11:03, Kelvin Chua kel...@gmail.com wrote:

 4.3.1

 kamctl ul show shows blank on the other machine. the registrar is ok.

 here is the DMQ message captured using ngrep

 KDMQ sip:usrloc@1.2.3.4:5090 SIP/2.0.
 Via: SIP/2.0/UDP 1.2.3.4:5080
 ;branch=z9hG4bK97d1.f42de484.0.
 To: sip:usrloc@1.2.3.4:5090.
 From: sip:usrloc@1.2.3.4:5080
 ;tag=c5ea0db92ae4ac0758f8c2b12d8addf4-4d17.
 CSeq: 10 KDMQ.
 Call-ID: 40be1e6c728bfe96-27604@1.2.3.4.
 Content-Length: 403.
 User-Agent: Sipsurge Registrar/1.0.0.
 Max-Forwards: 1.
 Content-Type: application/json.
 .

 {action:1,aor:kelvin,ruid:uloc-55cdbac7-6bd4-1,c:sip:kelvin@192.168.1.120:5060
 ;alias=2.4.6.8~5060~1,received:,path:sip:1.2.3.4:6055
 ;lr;received=sip:2.4.6.8:5060,callid:559751939@192.168.1.120
 ,user_agent:qutecom/rev-g-trunk,instance:,expires:1439549023,cseq:2,flags:0,cflags:0,q:-1,last_modified:1439546083,methods:4294967295,reg_id:0}


 here are the modparams

 modparam(usrloc, db_mode, 0)
 modparam(usrloc, use_domain, MULTIDOMAIN)
 modparam(dmq, server_address, DMQ_INT)
 modparam(dmq, notification_address, DMQ_EXT)
 modparam(dmq_usrloc, enable, 1)


 sometimes, the registrar crashes, sometimes, the backup, sometimes both
 but i manage to catch kamctl ul show and this is how it looks like:

 Domain:: kam_location table=1024 records=1 max_slot=1
 AOR:: kelvin
 Contact:: sip:kelvin@192.168.1.120:5060;alias=2.4.6.8~5060~1
 Q=
 Expires:: deleted
 Callid:: 559751939@192.168.1.120
 Cseq:: 30
 User-agent:: qutecom/rev-g-trunk
 Path:: sip:1.2.3.4:6055;lr;received=sip:
 2.4.6.8:5060
 State:: CS_DIRTY
 Flags:: 0
 Cflag:: 0
 Socket:: udp:1.2.3.4:5080
 Methods:: 4294967295
 Ruid:: uloc-55cdbac7-6bd4-1
 Reg-Id:: 0
 Last-Keepalive:: 1439546340
 Last-Modified:: 1439546340


 Kelvin Chua

 On Thu, Aug 13, 2015 at 7:54 PM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hello,

 From which version has this been generated?

 Also, can you provide the output of kamctl ul show and the relevant
 modparam sections of your config (usrloc, dmq, dmq_usrloc).

 Cheers,
 Charles


 On 13 August 2015 at 10:23, Kelvin Chua kel...@gmail.com wrote:

 I don't know if this is related, this happens on 4.3.1 as well but
 usrloc crashes once it gets a DMQ

 #0  0x7f2a82a5727c in get_urecord_by_ruid (_d=0x0,
 _aorhash=8118438, _ruid=0x7ffe6028d5c0, _r=0x7ffe6028d568,
 _c=0x7ffe6028d578)
 at udomain.c:1153
 #1  0x7f2a8282029a in usrloc_get_all_ucontact
 (node=0x7f2a01790998) at usrloc_sync.c:166
 #2  0x7f2a828240a7 in usrloc_dmq_handle_msg (msg=0x7f2a01790f78,
 resp=0x7ffe6028d820, node=0x7f2a01790998) at usrloc_sync.c:363
 #3  0x7f2a83959e6f in worker_loop (id=1) at worker.c:105
 #4  0x7f2a8393e566 in child_init (rank=0) at dmq.c:293
 #5  0x00599658 in init_mod_child (m=0x7f2a86955c78, rank=0) at
 sr_module.c:898
 #6  0x00599390 in init_mod_child (m=0x7f2a86956098, rank=0) at
 sr_module.c:895
 #7  0x00599390 in init_mod_child (m=0x7f2a86956a78, rank=0) at
 sr_module.c:895
 #8  0x00599390 in init_mod_child (m=0x7f2a86956e68, rank=0) at
 sr_module.c:895
 #9  0x00599390 in init_mod_child (m=0x7f2a869570f8, rank=0) at
 sr_module.c:895
 #10 0x00599390 in init_mod_child (m=0x7f2a86957498, rank=0) at
 sr_module.c:895
 #11 0x00599390 in init_mod_child (m=0x7f2a86957838, rank=0) at
 sr_module.c:895
 #12 0x00599390 in init_mod_child (m=0x7f2a86957db0, rank=0) at
 sr_module.c:895
 #13 0x00599390 in init_mod_child (m=0x7f2a86958090, rank=0) at
 sr_module.c:895
 #14 0x00599390 in init_mod_child (m=0x7f2a869582c0, rank=0) at
 sr_module.c:895
 #15 0x00599390 in init_mod_child (m=0x7f2a869589b0, rank=0) at
 sr_module.c:895
 #16 0x00599390 in init_mod_child (m=0x7f2a86958f88, rank=0) at
 sr_module.c

Re: [SR-Users] dmq_usrloc

[Charles Chance]

Hi,

I'm not sure why, but the dmq_usrloc module was written with the usrloc
domain hardcoded as location.

I intend to address that along with some other improvements but in the
meantime, try changing your call to save() to use location instead of
kam_location...

save(location);

Please report back the result.

Cheers,
Charles
On 14 Aug 2015 11:03, Kelvin Chua kel...@gmail.com wrote:

 4.3.1

 kamctl ul show shows blank on the other machine. the registrar is ok.

 here is the DMQ message captured using ngrep

 KDMQ sip:usrloc@1.2.3.4:5090 SIP/2.0.
 Via: SIP/2.0/UDP 1.2.3.4:5080
 ;branch=z9hG4bK97d1.f42de484.0.
 To: sip:usrloc@1.2.3.4:5090.
 From: sip:usrloc@1.2.3.4:5080;tag=c5ea0db92ae4ac0758f8c2b12d8addf4-4d17.
 CSeq: 10 KDMQ.
 Call-ID: 40be1e6c728bfe96-27604@1.2.3.4.
 Content-Length: 403.
 User-Agent: Sipsurge Registrar/1.0.0.
 Max-Forwards: 1.
 Content-Type: application/json.
 .

 {action:1,aor:kelvin,ruid:uloc-55cdbac7-6bd4-1,c:sip:kelvin@192.168.1.120:5060
 ;alias=2.4.6.8~5060~1,received:,path:sip:1.2.3.4:6055
 ;lr;received=sip:2.4.6.8:5060,callid:559751939@192.168.1.120
 ,user_agent:qutecom/rev-g-trunk,instance:,expires:1439549023,cseq:2,flags:0,cflags:0,q:-1,last_modified:1439546083,methods:4294967295,reg_id:0}


 here are the modparams

 modparam(usrloc, db_mode, 0)
 modparam(usrloc, use_domain, MULTIDOMAIN)
 modparam(dmq, server_address, DMQ_INT)
 modparam(dmq, notification_address, DMQ_EXT)
 modparam(dmq_usrloc, enable, 1)


 sometimes, the registrar crashes, sometimes, the backup, sometimes both
 but i manage to catch kamctl ul show and this is how it looks like:

 Domain:: kam_location table=1024 records=1 max_slot=1
 AOR:: kelvin
 Contact:: sip:kelvin@192.168.1.120:5060;alias=2.4.6.8~5060~1
 Q=
 Expires:: deleted
 Callid:: 559751939@192.168.1.120
 Cseq:: 30
 User-agent:: qutecom/rev-g-trunk
 Path:: sip:1.2.3.4:6055;lr;received=sip:
 2.4.6.8:5060
 State:: CS_DIRTY
 Flags:: 0
 Cflag:: 0
 Socket:: udp:1.2.3.4:5080
 Methods:: 4294967295
 Ruid:: uloc-55cdbac7-6bd4-1
 Reg-Id:: 0
 Last-Keepalive:: 1439546340
 Last-Modified:: 1439546340


 Kelvin Chua

 On Thu, Aug 13, 2015 at 7:54 PM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hello,

 From which version has this been generated?

 Also, can you provide the output of kamctl ul show and the relevant
 modparam sections of your config (usrloc, dmq, dmq_usrloc).

 Cheers,
 Charles


 On 13 August 2015 at 10:23, Kelvin Chua kel...@gmail.com wrote:

 I don't know if this is related, this happens on 4.3.1 as well but
 usrloc crashes once it gets a DMQ

 #0  0x7f2a82a5727c in get_urecord_by_ruid (_d=0x0, _aorhash=8118438,
 _ruid=0x7ffe6028d5c0, _r=0x7ffe6028d568, _c=0x7ffe6028d578)
 at udomain.c:1153
 #1  0x7f2a8282029a in usrloc_get_all_ucontact (node=0x7f2a01790998)
 at usrloc_sync.c:166
 #2  0x7f2a828240a7 in usrloc_dmq_handle_msg (msg=0x7f2a01790f78,
 resp=0x7ffe6028d820, node=0x7f2a01790998) at usrloc_sync.c:363
 #3  0x7f2a83959e6f in worker_loop (id=1) at worker.c:105
 #4  0x7f2a8393e566 in child_init (rank=0) at dmq.c:293
 #5  0x00599658 in init_mod_child (m=0x7f2a86955c78, rank=0) at
 sr_module.c:898
 #6  0x00599390 in init_mod_child (m=0x7f2a86956098, rank=0) at
 sr_module.c:895
 #7  0x00599390 in init_mod_child (m=0x7f2a86956a78, rank=0) at
 sr_module.c:895
 #8  0x00599390 in init_mod_child (m=0x7f2a86956e68, rank=0) at
 sr_module.c:895
 #9  0x00599390 in init_mod_child (m=0x7f2a869570f8, rank=0) at
 sr_module.c:895
 #10 0x00599390 in init_mod_child (m=0x7f2a86957498, rank=0) at
 sr_module.c:895
 #11 0x00599390 in init_mod_child (m=0x7f2a86957838, rank=0) at
 sr_module.c:895
 #12 0x00599390 in init_mod_child (m=0x7f2a86957db0, rank=0) at
 sr_module.c:895
 #13 0x00599390 in init_mod_child (m=0x7f2a86958090, rank=0) at
 sr_module.c:895
 #14 0x00599390 in init_mod_child (m=0x7f2a869582c0, rank=0) at
 sr_module.c:895
 #15 0x00599390 in init_mod_child (m=0x7f2a869589b0, rank=0) at
 sr_module.c:895
 #16 0x00599390 in init_mod_child (m=0x7f2a86958f88, rank=0) at
 sr_module.c:895
 #17 0x00599390 in init_mod_child (m=0x7f2a869594d8, rank=0) at
 sr_module.c:895
 #18 0x0059996c in init_child (rank=0) at sr_module.c:924
 #19 0x004aa753 in main_loop () at main.c:1650
 #20 0x004b02b0 in main (argc=13, argv=0x7ffe6028e318) at
 main.c:2533


 Kelvin Chua

 On Sat, Aug 8, 2015 at 12:09 AM, Asgaroth 00asgarot...@gmail.com
 wrote:

 Hi,



 Yes, I do see it replicating contacts to the other 2 servers on the dmq
 bus, however

Re: [SR-Users] dmq_usrloc

[Charles Chance]

Hello,

From which version has this been generated?

Also, can you provide the output of kamctl ul show and the relevant
modparam sections of your config (usrloc, dmq, dmq_usrloc).

Cheers,
Charles


On 13 August 2015 at 10:23, Kelvin Chua kel...@gmail.com wrote:

 I don't know if this is related, this happens on 4.3.1 as well but usrloc
 crashes once it gets a DMQ

 #0  0x7f2a82a5727c in get_urecord_by_ruid (_d=0x0, _aorhash=8118438,
 _ruid=0x7ffe6028d5c0, _r=0x7ffe6028d568, _c=0x7ffe6028d578)
 at udomain.c:1153
 #1  0x7f2a8282029a in usrloc_get_all_ucontact (node=0x7f2a01790998) at
 usrloc_sync.c:166
 #2  0x7f2a828240a7 in usrloc_dmq_handle_msg (msg=0x7f2a01790f78,
 resp=0x7ffe6028d820, node=0x7f2a01790998) at usrloc_sync.c:363
 #3  0x7f2a83959e6f in worker_loop (id=1) at worker.c:105
 #4  0x7f2a8393e566 in child_init (rank=0) at dmq.c:293
 #5  0x00599658 in init_mod_child (m=0x7f2a86955c78, rank=0) at
 sr_module.c:898
 #6  0x00599390 in init_mod_child (m=0x7f2a86956098, rank=0) at
 sr_module.c:895
 #7  0x00599390 in init_mod_child (m=0x7f2a86956a78, rank=0) at
 sr_module.c:895
 #8  0x00599390 in init_mod_child (m=0x7f2a86956e68, rank=0) at
 sr_module.c:895
 #9  0x00599390 in init_mod_child (m=0x7f2a869570f8, rank=0) at
 sr_module.c:895
 #10 0x00599390 in init_mod_child (m=0x7f2a86957498, rank=0) at
 sr_module.c:895
 #11 0x00599390 in init_mod_child (m=0x7f2a86957838, rank=0) at
 sr_module.c:895
 #12 0x00599390 in init_mod_child (m=0x7f2a86957db0, rank=0) at
 sr_module.c:895
 #13 0x00599390 in init_mod_child (m=0x7f2a86958090, rank=0) at
 sr_module.c:895
 #14 0x00599390 in init_mod_child (m=0x7f2a869582c0, rank=0) at
 sr_module.c:895
 #15 0x00599390 in init_mod_child (m=0x7f2a869589b0, rank=0) at
 sr_module.c:895
 #16 0x00599390 in init_mod_child (m=0x7f2a86958f88, rank=0) at
 sr_module.c:895
 #17 0x00599390 in init_mod_child (m=0x7f2a869594d8, rank=0) at
 sr_module.c:895
 #18 0x0059996c in init_child (rank=0) at sr_module.c:924
 #19 0x004aa753 in main_loop () at main.c:1650
 #20 0x004b02b0 in main (argc=13, argv=0x7ffe6028e318) at
 main.c:2533


 Kelvin Chua

 On Sat, Aug 8, 2015 at 12:09 AM, Asgaroth 00asgarot...@gmail.com wrote:

 Hi,



 Yes, I do see it replicating contacts to the other 2 servers on the dmq
 bus, however, I did come across a strange issue with the max contacts
 parameter, which the devs are looking at:



 https://github.com/kamailio/kamailio/issues/265



 Cheers





 *From:* sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On
 Behalf Of *Kelvin Chua
 *Sent:* Friday 7 August 2015 14:56
 *To:* Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.org
 *Subject:* Re: [SR-Users] dmq_usrloc



 have you seen your usrloc actually populate data from another server?

 i can verify my dmq is working properly. just usrloc not populating.


 Kelvin Chua



 On Fri, Aug 7, 2015 at 6:22 PM, Asgaroth 00asgarot...@gmail.com wrote:

 Hi,



 I’m a new user to the dmq subsystem in kamailio, but, once I had a
 properly configured dmq environment and loaded dmq_usrloc after dmq, all I
 had to do was add the following to the module paramters:



 modparam(dmq_usrloc, enable, 1)



 Then in the routing logic of the registrar’s you just need to add the
 appropriate dmq message handling blocks.



 Hope this helps in some way.







 *From:* sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On
 Behalf Of *Kelvin Chua
 *Sent:* Thursday 6 August 2015 07:29
 *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
 Users Mailing List sr-users@lists.sip-router.org
 *Subject:* [SR-Users] dmq_usrloc



 is there any function to trigger replication?



 i enabled the module and dmq is working for my htables

 but usrloc is not replicating.



 kamctl ul show does not show anything



 i used ngrep to trace the KDMQ packets, i can see the frequent exchanges
 but nothing related to usrloc. am i doing something wrong?


 Kelvin Chua


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Re: [SR-Users] htable replication between several kamailio's through dmq

[Charles Chance]

Hi,

Currently not, I'm afraid.

It is on the to-do list, but other things have had to take priority.

If you'd like to add it on GitHub as a feature request, maybe I or someone
else will be able to look at it soon.

Cheers,
Charles


On 22 July 2015 at 15:23, José Seabra joseseab...@gmail.com wrote:

 Hello there,

 I'm using the dmq and htable modules to limit the number of concurrent
 calls in several kamailio's servers.

 Every call that one kamailio server receives, kamailio increments a htable
 variable that is replicated to other kamailios through dmq, but if one of
 these kamailios fails or is restarted it looses the data from htable
 variable  used to count the number of calls and it is not recovered again
 when kamailio starts again.


 Is there any way to  recover the data from htable variable when  kamailio
 starts, from another kamailio instances through dmq module?

 Best Regards
 --
 Cumprimentos
 José Seabra

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-- 
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t. 0121 285 4400m. 07932 063 891

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Re: [SR-Users] dmq_usrloc replication and max_contacts

[Charles Chance]

Hi,

It seems to me like another bug, although again I'm not familiar with the
module so can't be sure without looking that it is not intended behaviour.

I plan to look later this week when I have some time but in the meantime,
you can send me the examples directly if you like.

Cheers,
Charles


On 22 July 2015 at 11:41, Asgaroth 00asgarot...@gmail.com wrote:

 Hi,

 Any thoughts on this, I can provide actual examples for this, however, I
 would rather not send valid subsciber info directly to the list.

 If anyone is available for looking into it I can send the relevent
 information/examples directly to them.

 Thanks

 On 21/07/2015 15:03, Asgaroth wrote:

 Hi All,

 I have a situation where I have 2 devices registering with the same AOR,
 I have the registrar module's max_contacts parameter set to 1 and I use the
 0x04 flag on the save function.

 When I use dmq_usrloc for replication, I can see 2 contacts registered
 for the AOR on the client nodes whereas the node where the actual
 registration took place only has 1 contact for the AOR.

 Is this expected behaviour of the dmq_usrloc module in terms of
 registration or have a come accross another bug?

 Thanks



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Re: [SR-Users] dmq_is_from_node() issue

[Charles Chance]

Hi,

This is expected. The point of DMQ is to enable dynamic scaling without
having to define every node in advance in Kamailio config.

You should not use dmq_is_from_node() with dmq_handle_message() - it is not
intended to be a security check for KDMQ messages, but for other messages
replicated directly in your config (e.g. REGISTER).

For KDMQ, you should listen on a dedicated IP and/or port and lock down at
network/firewall level. Then your DMQ handling block should look something
like:

if (method == KDMQ) {
if ($Rp == 5070) {
dmq_handle_message();
} else {
drop;
}
}

Do this help? Or is there something else you're looking to achieve?

Cheers,
Charles


On 22 July 2015 at 15:22, Asgaroth 00asgarot...@gmail.com wrote:

 Hi All,

 I have an issue with the dmq_is_from_node() function, whereby it does not
 detect a node is on the bus if it has been removed from its table due to
 previously failed, i presume, ping requests.

 I have the following snippet in my main routing block:

 if(is_method(KDMQ)) {
 if (dmq_is_from_node()) {
 dmq_handle_message();
 exit;
 } else {
 xlog(L_INFO, KDMQ message recieved from node not on our
 bus!! ($si:$sp) Dropping.);
 exit;
 }
 }

 I have the following dmq module parameters enabled for dmq:

 modparam(dmq, server_address, DMQ_ADDRESS)
 modparam(dmq, notification_address, sip:lb.domain.com:5060)
 modparam(dmq, multi_notify, 1)
 modparam(dmq, num_workers, 4)

 DMQ_ADDRESS is set to each of the addresses in the lb.domain.com for each
 local kamailio instance

 lb.domain.com resolves to the following addresses:
 10.6.0.109
 10.6.0.110
 10.6.0.111

 If I start up the 1st instance of kamailio and do not start the other 2
 within the default ping timout then I get the expected log message of dmq
 removing the nodes from its table, for example:

 kamailio[26657]: ERROR: dmq [notification_peer.c:588]:
 notification_resp_callback_f(): deleting server sip:10.6.0.111:5060
 because of failed request
 kamailio[26657]: ERROR: dmq [notification_peer.c:588]:
 notification_resp_callback_f(): deleting server sip:10.6.0.110:5060
 because of failed request

 However, if I then start one, or both, of the other nodes, then the KDMQ
 requests are dropped due to my snippet above, I presume, because
 dmq_is_from_node() checks its active table. See messages below as an
 example of the above snippet in action:

 kamailio[26656]: INFO: script: KDMQ message recieved from node not on
 our bus!! (10.6.0.110:5060) Dropping.
 kamailio[26656]: INFO: script: KDMQ message recieved from node not on
 our bus!! (10.6.0.111:5060) Dropping.

 Is this expected behaviour from the dmq_is_from_node() function? I would
 have expected it to always check the addresses resolved in the notification
 address, or am I using this function in a manner not intended.

 Any suggestions/tips would be greatly appreciated.

 Thanks


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Re: [SR-Users] dmq_replicate deserializing?

[Charles Chance]

Hello,

As with your other thread, I suspect dmq_usrloc is at fault here. I've not
looked at this module before, but can take a look over the next day or so
if no one else gets to it first.

Cheers,
On 1 Jul 2015 22:13, Asgaroth 00asgarot...@gmail.com wrote:

  Hi,

 Sorry, the previous sample, was the incorrect timestamp, this one is the
 corresponding message sent:

 U 2015/07/01 20:40:52.446646 10.6.0.173:5060 - 10.6.0.174:5060

 KDMQ sip:usrloc@10.6.0.174:5060 SIP/2.0.
 Via: SIP/2.0/UDP
 10.6.0.173;branch=z9hG4bK9b3c.fe993716.0.
 To: sip:usrloc@10.6.0.174:5060 sip:usrloc@10.6.0.174:5060.
 From: sip:usrloc@10.6.0.173:5060 sip:usrloc@10.6.0.173:5060
 ;tag=390c95c339281829d3cea6f43c8512cb-7e62.
 CSeq: 10 KDMQ.
 Call-ID: 3c502b38465e8d94-8433@10.6.0.173.
 Content-Length: 471.
 User-Agent: kamailio (bfievkrl01).
 Max-Forwards: 1.
 Content-Type: application/json.
 .
 {action:1,aor:subscriber_name@subscriber_domain
 ,ruid:uloc-55941474-20f1-1,c:
 sip:subscriber_name@212.2.160.202:61270;rinstance=7b7c543d11c12134;transport=UDP
 sip:subscriber_name@212.2.160.202:61270;rinstance=7b7c543d11c12134;transport=UDP
 ,received:sip:212.2.160.202:61270
 ,path:sip:10.7.0.109;lr;received=sip:212.2.160.202:61270,callid:rvOjXlyhAGK1aokNbR859w..,user_agent:Z
 3.7.30891
 r30851,instance:,expires:1435779765,cseq:6,flags:0,cflags:64,q:-1,last_modified:1435779652,methods:5087,reg_id:0}



 On 01/07/2015 21:30, Charles Chance wrote:

 Hello,

 Should be no need to load any additional deserializer. The docs simply
 mean if you wish to send/receive your own messages from within a module or
 script, it's up to you to choose the best payload type and method of
 (de)serialization.

 Can you post an example KDMQ message here to look at?

 Cheers,
 Charles
 On 1 Jul 2015 20:49, Asgaroth 00asgarot...@gmail.com wrote:

 Hi All,

 I am playing with the new dmq_replicate module and am banging my head
 against an issue I have come accross. I can see the kamailio registrars
 sending the replication messages to the other node(s) in the dmq bus, and
 the payload looks to be of type json. However, on the recieving nodes, when
 I try to issue a kamctl ul show --brief, the output of the AOR's are
 garbled.

 I had a look at the dmq/dmq_ursloc moduled, and the dmq module docs
 mention that we may need to load our own deserializers, is this the case
 when using the dmq_usrloc module as well? I am only using dmq to replicate
 registration messages.

 Here is a sample of the dmq workers on a recieving node applying the
 update, and you can see that it thinks the contact to add is 'p÷#031#002',
 is was expecting the actual AOR of the subriber to show up here.

 Am I missing something simple here or is there something more sinister at
 play.

 Any pointers would be greatly appreciated.

 Kamailio Version:

 version: kamailio 4.3.0 (x86_64/linux) c6aa95
 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
 DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
 F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
 USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
 id: c6aa95
 compiled on 16:14:27 Jun 23 2015 with gcc 4.4.7

 Debug log below:

 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/msg_parser.c:606]: parse_msg(): SIP Request:
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/msg_parser.c:608]: parse_msg():  method: KDMQ
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/msg_parser.c:610]: parse_msg():  uri: sip:usrloc@10.6.0.174:5060
 
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/msg_parser.c:612]: parse_msg():  version: SIP/2.0
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/parse_via.c:1254]: parse_via_param(): Found param type 232,
 branch = z9hG4bK9b3c.fe993716.0; state=16
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found,
 flags=2
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the
 first via
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [receive.c:134]: receive_msg(): After parse_msg...
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [receive.c:177]: receive_msg(): preparing to run routing scripts...
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: tm
 [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=450 global
 id=449 T start

Re: [SR-Users] dmq_replicate deserializing?

[Charles Chance]

Hello,

Should be no need to load any additional deserializer. The docs simply mean
if you wish to send/receive your own messages from within a module or
script, it's up to you to choose the best payload type and method of
(de)serialization.

Can you post an example KDMQ message here to look at?

Cheers,
Charles
On 1 Jul 2015 20:49, Asgaroth 00asgarot...@gmail.com wrote:

 Hi All,

 I am playing with the new dmq_replicate module and am banging my head
 against an issue I have come accross. I can see the kamailio registrars
 sending the replication messages to the other node(s) in the dmq bus, and
 the payload looks to be of type json. However, on the recieving nodes, when
 I try to issue a kamctl ul show --brief, the output of the AOR's are
 garbled.

 I had a look at the dmq/dmq_ursloc moduled, and the dmq module docs
 mention that we may need to load our own deserializers, is this the case
 when using the dmq_usrloc module as well? I am only using dmq to replicate
 registration messages.

 Here is a sample of the dmq workers on a recieving node applying the
 update, and you can see that it thinks the contact to add is 'p÷#031#002',
 is was expecting the actual AOR of the subriber to show up here.

 Am I missing something simple here or is there something more sinister at
 play.

 Any pointers would be greatly appreciated.

 Kamailio Version:

 version: kamailio 4.3.0 (x86_64/linux) c6aa95
 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
 DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
 F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
 USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
 id: c6aa95
 compiled on 16:14:27 Jun 23 2015 with gcc 4.4.7

 Debug log below:

 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/msg_parser.c:606]: parse_msg(): SIP Request:
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/msg_parser.c:608]: parse_msg():  method: KDMQ
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/msg_parser.c:610]: parse_msg():  uri: sip:usrloc@10.6.0.174:5060
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/msg_parser.c:612]: parse_msg():  version: SIP/2.0
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/parse_via.c:1254]: parse_via_param(): Found param type 232,
 branch = z9hG4bK9b3c.fe993716.0; state=16
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found,
 flags=2
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the
 first via
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [receive.c:134]: receive_msg(): After parse_msg...
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [receive.c:177]: receive_msg(): preparing to run routing scripts...
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: tm
 [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=450 global
 id=449 T start=0x
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached,
 state=10
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: To
 [30]; uri=[sip:usrloc@10.6.0.174:5060]
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [
 sip:usrloc@10.6.0.174:5060#015#012]
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq CSeq:
 10 KDMQ
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body :
 content_length=471
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: core
 [parser/msg_parser.c:89]: get_hdr_field(): found end of header
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: tm
 [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching:
 hash=50105, isACK=0
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: tm
 [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching
 failed
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: tm
 [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no
 transaction found
 Jul  1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: tm
 

Re: [SR-Users] add feature to dmq to allow for HA (heartbeat or pacemaker)

[Charles Chance]

Hello,

Thanks for the suggestion.

Callbacks for change of node state can be added easily enough and I agree
they could be quite useful to other modules.

Development is frozen currently but I can push to a separate branch if
you're interested in testing?

Cheers,
Charles
 On 24 Apr 2015 13:42, Vik Killa vipki...@gmail.com wrote:

 Hello,
 I was looking to implement HA with 2 Kamailio machines.
 I am aware of the dmq module and I thought that would be a great place to
 start.
 The problem I see is that dmq module does not have any failure callback
 mechanism.
 For instance, if KamA sends dmq ping to KamB, and KamB does not respond
 (perhaps because it's down), then nothing happens.
 I looked into using the dmq API and checked the dmq code to see
 how send_message() function works, and I dont see any indication that there
 is a failure callback.
 I would be a great improvement to add this sort of functionality to dmq
 and then other modules or code can be written to use the API to detect when
 another Kam box is up or down. This could lead to another module like
 dmq_ha (I'm open to writing this)

 Thanks,
 V

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Re: [SR-Users] registrar: add optional check for local path during lookup()

[Charles Chance]

Hi Volkan,

It is in the master (development) branch, so will appear in the next major
release (4.3).

The code could not be backported to 4.2 because it was considered a new
feature rather than a fix.

Cheers,

Charles

P.S. please use mailing list in case the question is useful to others :-)


On 21 April 2015 at 12:57, Volkan Hatem vol...@hatem.net wrote:

 Hi Charles,

 I was hoping to see your change (
 http://git.kamailio.org/gitlist/index.php/kamailio/commits/635f23b12eff2431ca9a14bb39f4204dc2a7227b/modules/registrar/lookup.c)
 in 4.2.4

 I could not tell to which branch your code was submitted. Do you know when
 it might find its way to an official release?

 Best,
 -volkan


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Re: [SR-Users] Kamailio Freeswitch TCP call origination

[Charles Chance]

Hello,

Since you are sharing location database, you might consider using Path.

Cheers,

Charles
 On 15 Mar 2015 08:52, Niraj Roy techfa...@gmail.com wrote:

 Hi,
Here is the setup I have configured.
   --Two Kamailio K1, K2 Works as a registrar and call load balancer
   -- Three freeswitch works as the media server.
  -- They follow common MySQL database.
  Regular call flow
   User A--- Kamailio (K1)--- Freeswitch (F1)Kamailio(K1)---User B.
 Everything works fine as long as all users are registered and making call
 with single Kamailio. Now the problem starts when we distribute users on
 more than one kamailio.
   Let us take the case.
UserA -- Registered over TCP with Kamailio (K1)
UserB -- Registered over TCP with Kamailio (K2)
 Now User A initiates a call.. It will follow the same path
 User A--- Kamailio (K1)--- Freeswitch (F1)Kamailio(K1)---User B.
 Since User B is registered with K2. K1 being a public server won't be able
 to initiate any call to User B because there is no TCP connection
 established with it.
  Can anybody shade some light on this?

 Thanks,



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Re: [SR-Users] How to Use Multiple DMQ Notification Addresses

[Charles Chance]

Hi Bob,

Yes, sr-dev would probably be more appropriate. Or if you would like to
send a patch directly or raise a pull request on Github, that would be fine
also.

SRV support is probably the way to go, since adding multiple addresses in
config suggests a prior knowledge of nodes in advance, which doesn't fit
with the DMQ ideology. I did start but got sidetracked with other
commitments, as happens increasingly more often lately!

Cheers,

Charles



On 4 March 2015 at 13:20, Robert Boisvert rdboisv...@gmail.com wrote:

 Charles,

 I am working with the team that requested the SRV enhancement.  I have
 some time now to do some coding and would like to work on a patch to allow
 multiple notification addresses.  Would you be willing to discuss this?  We
 can take this conversation offline or move it to SR-Dev if you like.

 Thank you for considering this,
 Bob

 On Wed, Mar 4, 2015 at 4:17 AM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hello Bob,

 It is only possible to specify one notification address. So it is
 important to make sure it points initially to an active node.

 Once it has joined the cluster, the active node list is shared between
 all servers, so it should not matter which servers are up or down later.

 But for now, the notification address must be active at least on startup.
 It seems we may need to add SRV support to the dmq module at some point,
 when there is some spare time :-)

 Cheers,

 Charles
  On 3 Mar 2015 23:39, Robert Boisvert rdboisv...@gmail.com wrote:

 Hello all,

 I would like to setup a DMQ bus where servers can come on and offline
 and have DMQ continue working with the online servers.  I'm having trouble
 understanding how to do that with the notification address.  Can someone
 point me in the right direction?

 More specifically, suppose I have five servers named A, B, C, D, and E.
 I would like all of them to be on the DMQ bus when active but they may come
 up at different times.  For example, when A, B and C start I want each of
 them to be on the same DMQ bus.  However, the way I understand it, if I
 make E the notification address none will be on the bus because E is not
 active.

 I could create a circular notification chain where A uses B as a
 notification server, B uses C, and so on.  Of course, there are problems
 when one or more servers are missing from the chain.

 Any help on this is greatly appreciated.

 Thanks,
 Bob

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Re: [SR-Users] How to Use Multiple DMQ Notification Addresses

[Charles Chance]

Hello Bob,

It is only possible to specify one notification address. So it is important
to make sure it points initially to an active node.

Once it has joined the cluster, the active node list is shared between all
servers, so it should not matter which servers are up or down later.

But for now, the notification address must be active at least on startup.
It seems we may need to add SRV support to the dmq module at some point,
when there is some spare time :-)

Cheers,

Charles
 On 3 Mar 2015 23:39, Robert Boisvert rdboisv...@gmail.com wrote:

 Hello all,

 I would like to setup a DMQ bus where servers can come on and offline and
 have DMQ continue working with the online servers.  I'm having trouble
 understanding how to do that with the notification address.  Can someone
 point me in the right direction?

 More specifically, suppose I have five servers named A, B, C, D, and E.  I
 would like all of them to be on the DMQ bus when active but they may come
 up at different times.  For example, when A, B and C start I want each of
 them to be on the same DMQ bus.  However, the way I understand it, if I
 make E the notification address none will be on the bus because E is not
 active.

 I could create a circular notification chain where A uses B as a
 notification server, B uses C, and so on.  Of course, there are problems
 when one or more servers are missing from the chain.

 Any help on this is greatly appreciated.

 Thanks,
 Bob

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Re: [SR-Users] dmq_t_replicate fails with ERROR: t_forward_nonack: no branches for forwarding

[Charles Chance]

On 6 Feb 2015 09:27, Leonardo Arena rna...@gmail.com wrote:

 On gio, 2015-02-05 at 09:29 +, Charles Chance wrote:
  Hi,
 
  On 5 Feb 2015 09:18, Leonardo Arena rna...@gmail.com wrote:
   I saw the patch on sr-dev. Thanks!
  
 
  Did you have chance to test it yourself? With it, you do not need to
  call append_branch() in config.

 For me master (f9c248a) does not compile and the patch does not apply to
 4.2.2. I'll wait for the patch to show up in master and retry.


Master does not compile anyway, or only with the patch?

 Thanks!

 - leo

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Re: [SR-Users] dmq_t_replicate fails with ERROR: t_forward_nonack: no branches for forwarding

[Charles Chance]

Ok, you should report the cnxcc issue separately if required.

For now, I have pushed the patch to master branch but will also backport to
4.2 later.

Cheers,

Charles

On 6 February 2015 at 10:59, Leonardo Arena rna...@gmail.com wrote:

 On ven, 2015-02-06 at 09:37 +, Charles Chance wrote:
 
  On 6 Feb 2015 09:27, Leonardo Arena rna...@gmail.com wrote:
  
   On gio, 2015-02-05 at 09:29 +, Charles Chance wrote:
Hi,
   
On 5 Feb 2015 09:18, Leonardo Arena rna...@gmail.com wrote:
 I saw the patch on sr-dev. Thanks!

   
Did you have chance to test it yourself? With it, you do not need
  to
call append_branch() in config.
  
   For me master (f9c248a) does not compile and the patch does not
  apply to
   4.2.2. I'll wait for the patch to show up in master and retry.
  
 
  Master does not compile anyway, or only with the patch?
 

 The error it's unrelated to the patch:

 In file included from cnxcc_check.c:33:0:
 cnxcc_redis.h:28:29: fatal error: hiredis/hiredis.h: No such file or
 directory
  #include hiredis/hiredis.h
  ^
 compilation terminated.
 In file included from cnxcc_mod.c:68:0:
 cnxcc_redis.h:28:29: fatal error: hiredis/hiredis.h: No such file or
 directory
  #include hiredis/hiredis.h
  ^
 compilation terminated.
 ../../Makefile.rules:97: recipe for target 'cnxcc_check.o' failed
 make[1]: *** [cnxcc_check.o] Error 1
 make[1]: *** Waiting for unfinished jobs
 ../../Makefile.rules:97: recipe for target 'cnxcc_mod.o' failed
 make[1]: *** [cnxcc_mod.o] Error 1
 Makefile:511: recipe for target 'modules' failed
 make: *** [modules] Error 1



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Re: [SR-Users] dmq_t_replicate fails with ERROR: t_forward_nonack: no branches for forwarding

[Charles Chance]

Hi,

On 5 Feb 2015 09:18, Leonardo Arena rna...@gmail.com wrote:

 On mer, 2015-02-04 at 21:15 +, Charles Chance wrote:
  Hi,
 
  On 3 Feb 2015 10:07, Leonardo Arena rna...@gmail.com wrote:
  
   On lun, 2015-02-02 at 17:54 +, Charles Chance wrote:

   [snip ]
  
  dmq.my.domain.netresolves to 10.0.0.2 which is proxy2. What I'm
   effectively trying to do with DNS is to avoid having a static entry
  in
   configuration file as notification address. I'd like to build a
  dynamic
   list of notification servers via SRV records and have DMQ honor the
   priority and weight of such list.
  
  
 
  This will not work currently but I may have some time soon to look
  into it. Unless someone else proposes a patch first.
 

 Yeah, this was my conclusion too after few tests. An SRV query is
 actually sent (but only if there's an A record for the the given
 notification address), the higher priority address is chosen, but if
 this fails it never failover to a lower priority address.

 Also if notification server goes down, I haven't seen any attempt to
 re-contacting it in case it goes back on-line. Or may be I didn't wait
 enough time to observe it.


It should do, if the address is configured statically.

 I think it would be nice to have the information of notification servers
 in DNS SRV records. This would help in building a distributed
 infrastructure, rather than having such info in static configuration
 files, which are more suitable for clusters.

   
   
I will try to replicate your scenario and come back with a
  solution
shortly.
   
   
 
  I have a patch which is working in my tests but I am waiting to hear
  of any possible side effects. It should be in master branch shortly.

 I saw the patch on sr-dev. Thanks!


Did you have chance to test it yourself? With it, you do not need to call
append_branch() in config.

 - leonardo

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Re: [SR-Users] dmq_t_replicate fails with ERROR: t_forward_nonack: no branches for forwarding

[Charles Chance]

Hi,

On 3 Feb 2015 10:07, Leonardo Arena rna...@gmail.com wrote:

 On lun, 2015-02-02 at 17:54 +, Charles Chance wrote:
  Hi,
 
 
  Can you send the output of kamcmd dmq.list_nodes (mask IP addresses
  if needed)?

 proxy2:/etc/kamailio# kamcmd dmq.list_nodes
 {
 host: 10.0.0.3
 port: 5060
 resolved_ip: 10.0.0.3
 status: 2
 last_notification: 0
 local: 0
 }
 {
 host: 10.0.0.1
 port: 5060
 resolved_ip: 10.0.0.1
 status: 2
 last_notification: 0
 local: 0
 }
 {
 host:dmq.my.domain.net
 port:
 resolved_ip: 0.0.0.0
 status: 2
 last_notification: 0
 local: 0
 }
 {
 host: 10.0.0.2
 port: 5060
 resolved_ip: 10.0.0.2
 status: 8
 last_notification: 0
 local: 1
 }


dmq.my.domain.netresolves to 10.0.0.2 which is proxy2. What I'm
 effectively trying to do with DNS is to avoid having a static entry in
 configuration file as notification address. I'd like to build a dynamic
 list of notification servers via SRV records and have DMQ honor the
 priority and weight of such list.



This will not work currently but I may have some time soon to look into it.
Unless someone else proposes a patch first.

 
 
  I will try to replicate your scenario and come back with a solution
  shortly.
 
 

I have a patch which is working in my tests but I am waiting to hear of any
possible side effects. It should be in master branch shortly.

Cheers,

Charles

  By the way, your dmq notification_address parameter looks slightly
  odd (although I don't know what you're doing with DNS) - it should
  simply point to one other node in the cluster, so that on startup it
  can request information about any other nodes from this one. Also, in
  the block which calls dmq_handle_message(), you should exit straight
  away after.


 Fixed.


 
 
  One final note, you should really listen on a separate port for dmq
  purposes (e.g. 5080) and restrict access to it at firewall level. You
  can then check in config that the message was received on this port,
  like:
 
 
  if (method == KDMQ  $Rp == 5080) {
 dmq_handle_message();
 exit;
  }


 Tnx, I will do that. Now I'm just running in a lab.

 Thanks!

 - leonardo


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Re: [SR-Users] dmq_t_replicate fails with ERROR: t_forward_nonack: no branches for forwarding

[Charles Chance]

Hi,

Can you send the output of kamcmd dmq.list_nodes (mask IP addresses if
needed)?

I will try to replicate your scenario and come back with a solution shortly.

By the way, your dmq notification_address parameter looks slightly odd
(although I don't know what you're doing with DNS) - it should simply point
to one other node in the cluster, so that on startup it can request
information about any other nodes from this one. Also, in the block which
calls dmq_handle_message(), you should exit straight away after.

One final note, you should really listen on a separate port for dmq
purposes (e.g. 5080) and restrict access to it at firewall level. You can
then check in config that the message was received on this port, like:

if (method == KDMQ  $Rp == 5080) {
   dmq_handle_message();
   exit;
}


Regards,

Charles


On 2 February 2015 at 10:52, Leonardo Arena rna...@gmail.com wrote:

 Hi all,
 I have three Registrar-only proxies DMQ interconnected [1]. I'm
 attempting to use a client to register to one of them and replicate
 REGISTRAR message to the other two.

 * What it works:
 - Clients registration to proxy1
 - Proxy1 replication of REGISTRAR message to proxy2

 * What does not work:
 - Proxy replication of REGISTRAR message to proxy3

 * Error message of proxy1:
 ERROR: tm [t_fwd.c:1770]: t_forward_nonack(): ERROR: t_forward_nonack:
 no branches for forwarding
 ERROR: dmq [dmq_funcs.c:410]: cfg_dmq_t_replicate(): error calling
 t_replicate

 append_branch() did't help me and a search on the ML either.

 What am I doing wrong?

 Thanks!

 - leonardo


 [1]

 #
 # Global Definitions
 #
 debug  =   2
 fork  =   yes
 log_stderror   =   no
 # listen  =   0.0.0.0
 port  =   5060
 children   =   4

 dns  =   no
 rev_dns  =   no
 use_dns_failover =on
 dns_srv_lb   =   yes
 #dns_try_naptr=no


 mpath  =   /usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/

 #
 # Modules
 #

 loadmodule   tm.so
 loadmodule   sl.so
 loadmodule   tmx.so
 loadmodule   maxfwd.so
 loadmodule   usrloc.so
 loadmodule   registrar.so
 loadmodule   pv.so
 loadmodule   xlog.so
 loadmodule   ctl.so
 #loadmodule   db_text.so
 loadmodule   db_sqlite.so
 loadmodule   auth.so
 loadmodule   auth_db.so
 loadmodule   memcached.so
 loadmodule   dmq.so
 loadmodule   corex.so

 #
 # Module Configuration
 #

 modparam ( usrloc, db_mode, 0 )
 modparam ( usrloc, use_domain, 1 )
 modparam ( usrloc, hash_size, 20 )
 #modparam ( ctl, binrpc_max_body_size, 1 )
 #modparam ( ctl, binrpc_struct_max_body_size, 200 )
 #default
 #modparam ( db_text, db_mode, 0 )
 modparam ( auth, use_domain, 1 )
 #modparam ( auth_db, db_url, text:///etc/kamailio/dbtext/ )
 modparam ( auth_db, db_url, sqlite:///etc/kamailio/kamailiodb )
 modparam ( auth_db, use_domain, 1 )
 #use plaintext passwords for testing
 modparam ( auth_db, calculate_ha1, 1 )
 modparam ( auth_db, version_table, 0 )
 modparam ( auth_db, password_column, password )
 modparam ( auth_db, load_credentials, $avp(s:username)=username;
 $avp(s:domain)=domain;$avp(s:password)=password )

 modparam( memcached, servers, localhost:11211 )

 modparam( dmq, server_address, sip:10.0.0.1:5060 )
 modparam( dmq, notification_address, sip:dmq.my.domain.net )
 #modparam( dmq, num_threads, 4 )
 # modparam( dmq, ping_interval, 60 )

 #
 # Main Route Block
 #

 include_file kamailio-errors.cfg

 route
   {
# Uncomment next line if you  want to see
# each incoming message
xlog( L_INFO, $rm: $fu - $ru );

if (!mf_process_maxfwd_header(20)) {
   route(ERROR_TOO_MANY_HOPS);
}

if (msg:len  4096) {
   route(ERROR_MESSAGE_OVERFLOW);
}

if ($rm == REGISTER ) {
   if ( dmq_is_from_node() ) {
  xlog( L_INFO, REGISTER from node );
  route(REGISTER);
  exit;
   }
   else {
  # if we don't have credentials in cache authenticate against DB
  if ( $mct($tU@$td) == $null ) {
 if (!www_authenticate($td, subscriber)) {
www_challenge($td, 1);
exit;
 }
 else {
route(REGISTER);
xlog( L_INFO, Authenticated user $tU for domain $td
 against DB );
# store credentials into cache
$mct($avp(s:username)@$avp(s:domain)) = $avp(s:password);
exit;
 }

  }
  # credentials are in cache, authenticate against it
  else {
 if (!pv_www_authenticate($td, 

Re: [SR-Users] dialog module shared memory

[Charles Chance]

Hello,

Currently, this is work in progress. But in the meantime, it is possible to
achieve using htable with dmq enabled.

I can post an example later if required.

Cheers,
Charles
 On 16 Jan 2015 07:54, Grant Bagdasarian g...@cm.nl wrote:

  Hello,



 I’m using the dialog module to control the maximum number of calls allowed
 to be passed through.

 Is it possible to share the dialogs stored in memory with multiple
 instances of kamailio?



 So if both kamailio instances process calls, and the maximum is set to a
 global of 10, both combined should never let more than 10 calls through.



 Regards,



 Grant

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Re: [SR-Users] Help analysing segmentation fault

[Charles Chance]

Hi Both,

In this case the system was using binaries from opensuse build service, so
gcc I believe?

Cheers,

Charles
 On 28 Aug 2014 21:25, Daniel-Constantin Mierla mico...@gmail.com wrote:

  Hello,

 On 28/08/14 20:32, Jason Penton wrote:

 Hey Daniel,

  I am puzzled by how this could make any difference? Could you explain?
 Is this dependent on the compiler used and whether or not void* arithmetic
 is allowed?


 void is incomplete type, of no defined data size, you cannot have:

 void x;

 There is nothing you can assign to x.

 An increment of a pointer jumps in memory with the size of its data type
 (e.g., char* is incremented by 1 byte, int* is incremented by 4 bytes). But
 for void is mostly unknown behaviour. Based on web, apparently gcc
 considers sizeof(void) to be 1, but it is not a C standard. Others suggests
 should be zero (because that's why is a 'void' pointer :-) ).

 Maybe Charles can say what compiler he was using.

 Cheers,
 Daniel


  Cheers
 Jason


 On Fri, Aug 22, 2014 at 1:17 PM, Daniel-Constantin Mierla 
 mico...@gmail.com wrote:

  Hello,

 can you try this small patch?

 diff --git a/modules/pua_dialoginfo/pua_dialoginfo.c
 b/modules/pua_dialoginfo/pua_dialoginfo.c
 index 1e88a04..0f02b2b 100644
 --- a/modules/pua_dialoginfo/pua_dialoginfo.c
 +++ b/modules/pua_dialoginfo/pua_dialoginfo.c
 @@ -347,7 +347,7 @@ struct str_list* get_str_list(unsigned short
 avp_flags, int_str avp_name) {

 memset( list_current, 0, len);

 -   list_current-s.s = (char*)( (void*) list_current +
 sizeof(struct str_list));
 +   list_current-s.s = (char*)list_current + sizeof(struct
 str_list);
 list_current-s.len = avp_value.s.len;
 memcpy(list_current-s.s,avp_value.s.s,avp_value.s.len);

 It is for 4.1.

 I have some ongoing work to commit soon on the master branch. if you
 confirm it is working fine, I will push this patch as well and backport to
 4.1.

 Cheers,
 Daniel


 On 22/08/14 13:03, Charles Chance wrote:

  Hi All,

  I wonder if some one could help me to diagnose a recurring issue?

  It happens at random times/intervals and under varying load. But
 always, just before the time of crash, I see the same critical error in log:

  CRITICAL: dialog [dlg_hash.c:841]: log_next_state_dlg(): bogus event 6
 in state 1 for dlg 0xb0632134 [1367:5814] with clid '
 0695dd7a346188dd24e7520e6c010...@sip.sipcentric.com' and tags
 'as77c89620' ''


  Analysing the core dump reveals:

  Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio.pid -m
 128 -M 4 -u kamailio -g kamailio'.
 Program terminated with signal 11, Segmentation fault.
 #0  0x081a737c in parse_uri (buf=0x3a70006e Address 0x3a70006e out of
 bounds, len=275, uri=0xbfa2fd2c) at parser/parse_uri.c:389
 389 scheme=buf[0]+(buf[1]8)+(buf[2]16)+(buf[3]24);

  (gdb) frame 1

  #1  0x008fe5dd in dialog_publish (state=0x903f37 Trying,
 ruri=0xb0b5fd00, entity=0xb0632188, peer=0xb0632190, callid=0xb0632180,
 initiator=1, lifetime=7200, localtag=0x0, remotetag=0x0, localtarget=0x0,
 remotetarget=0x0, do_pubruri_localcheck=1) at dialog_publish.c:275
 275 if (parse_uri(ruri-s, ruri-len, ruri_uri)  0) {

  (gdb) p *ruri

   $1 = {s = 0x3a70006e Address 0x3a70006e out of bounds, len = 275}

  (gdb) up

  #2  0x008ff277 in dialog_publish_multi (state=0x903f37 Trying,
 ruris=0xb0b5fd00, entity=0xb0632188, peer=0xb0632190, callid=0xb0632180,
 initiator=1, lifetime=7200, localtag=0x0, remotetag=0x0,
 localtarget=0x0, remotetarget=0x0, do_pubruri_localcheck=1) at
 dialog_publish.c:387
 387
 dialog_publish(state,(ruris-s),entity,peer,callid,initiator,lifetime,localtag,remotetag,localtarget,remotetarget,do_pubruri_localcheck);

  (gdb) p *ruris

  $2 = {s = {s = 0x3a70006e Address 0x3a70006e out of bounds, len =
 275}, next = 0x0}

  (gdb) up

  #3  0x0090187a in __dialog_created (dlg=0xb0632134, type=2,
 _params=0x6db064) at pua_dialoginfo.c:470
 470 dialog_publish_multi(Trying, dlginfo-pubruris_caller,
 (dlg-from_uri), (include_req_uri)?(dlg-req_uri):(dlg-to_uri),
 (dlg-callid), 1, dlginfo-lifetime, 0, 0, 0, 0,
 send_publish_flag==-1?1:0);

  (gdb) p *dlginfo-pubruris_caller

  $3 = {s = {s = 0x31590014 Address 0x31590014 out of bounds, len =
 275}, next = 0xb0b5fd00}

  (gdb) p *dlginfo-pubruris_caller-next

  $4 = {s = {s = 0x3a70006e Address 0x3a70006e out of bounds, len =
 275}, next = 0x0}


  In config, for pua_dialoginfo we are enabling the option
 use_pubruri_avps and setting pubruri_caller_avp and
 pubruri_callee_avp accordingly.

  Therefore, in pua_dialoginfo.c it is using get_str_list() function to
 set dlginfo-pubruris_caller from the avp.

  Could this be some race condition or something completely different?

  Thanks in advance,

  Charles



 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered
 office: Faraday Wharf, Innovation Birmingham Campus

Re: [SR-Users] Help analysing segmentation fault

[Charles Chance]

Hi Daniel,

The patch has tested OK so far.

Regards,

Charles



On 22 August 2014 12:37, Charles Chance charles.cha...@sipcentric.com
wrote:

 Thanks, Daniel.

 It can be hours, days or weeks between occurrences,  but I will report
 back after a day or two initially then continue to monitor.

 Cheers,

 Charles
  On 22 Aug 2014 12:18, Daniel-Constantin Mierla mico...@gmail.com
 wrote:

  Hello,

 can you try this small patch?

 diff --git a/modules/pua_dialoginfo/pua_dialoginfo.c
 b/modules/pua_dialoginfo/pua_dialoginfo.c
 index 1e88a04..0f02b2b 100644
 --- a/modules/pua_dialoginfo/pua_dialoginfo.c
 +++ b/modules/pua_dialoginfo/pua_dialoginfo.c
 @@ -347,7 +347,7 @@ struct str_list* get_str_list(unsigned short
 avp_flags, int_str avp_name) {

 memset( list_current, 0, len);

 -   list_current-s.s = (char*)( (void*) list_current +
 sizeof(struct str_list));
 +   list_current-s.s = (char*)list_current + sizeof(struct
 str_list);
 list_current-s.len = avp_value.s.len;
 memcpy(list_current-s.s,avp_value.s.s,avp_value.s.len);

 It is for 4.1.

 I have some ongoing work to commit soon on the master branch. if you
 confirm it is working fine, I will push this patch as well and backport to
 4.1.

 Cheers,
 Daniel

 On 22/08/14 13:03, Charles Chance wrote:

 Hi All,

  I wonder if some one could help me to diagnose a recurring issue?

  It happens at random times/intervals and under varying load. But
 always, just before the time of crash, I see the same critical error in log:

  CRITICAL: dialog [dlg_hash.c:841]: log_next_state_dlg(): bogus event 6
 in state 1 for dlg 0xb0632134 [1367:5814] with clid '
 0695dd7a346188dd24e7520e6c010...@sip.sipcentric.com' and tags
 'as77c89620' ''


  Analysing the core dump reveals:

  Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio.pid -m
 128 -M 4 -u kamailio -g kamailio'.
 Program terminated with signal 11, Segmentation fault.
 #0  0x081a737c in parse_uri (buf=0x3a70006e Address 0x3a70006e out of
 bounds, len=275, uri=0xbfa2fd2c) at parser/parse_uri.c:389
 389 scheme=buf[0]+(buf[1]8)+(buf[2]16)+(buf[3]24);

  (gdb) frame 1

  #1  0x008fe5dd in dialog_publish (state=0x903f37 Trying,
 ruri=0xb0b5fd00, entity=0xb0632188, peer=0xb0632190, callid=0xb0632180,
 initiator=1, lifetime=7200, localtag=0x0, remotetag=0x0, localtarget=0x0,
 remotetarget=0x0, do_pubruri_localcheck=1) at dialog_publish.c:275
 275 if (parse_uri(ruri-s, ruri-len, ruri_uri)  0) {

  (gdb) p *ruri

   $1 = {s = 0x3a70006e Address 0x3a70006e out of bounds, len = 275}

  (gdb) up

  #2  0x008ff277 in dialog_publish_multi (state=0x903f37 Trying,
 ruris=0xb0b5fd00, entity=0xb0632188, peer=0xb0632190, callid=0xb0632180,
 initiator=1, lifetime=7200, localtag=0x0, remotetag=0x0,
 localtarget=0x0, remotetarget=0x0, do_pubruri_localcheck=1) at
 dialog_publish.c:387
 387
 dialog_publish(state,(ruris-s),entity,peer,callid,initiator,lifetime,localtag,remotetag,localtarget,remotetarget,do_pubruri_localcheck);

  (gdb) p *ruris

  $2 = {s = {s = 0x3a70006e Address 0x3a70006e out of bounds, len =
 275}, next = 0x0}

  (gdb) up

  #3  0x0090187a in __dialog_created (dlg=0xb0632134, type=2,
 _params=0x6db064) at pua_dialoginfo.c:470
 470 dialog_publish_multi(Trying, dlginfo-pubruris_caller,
 (dlg-from_uri), (include_req_uri)?(dlg-req_uri):(dlg-to_uri),
 (dlg-callid), 1, dlginfo-lifetime, 0, 0, 0, 0,
 send_publish_flag==-1?1:0);

  (gdb) p *dlginfo-pubruris_caller

  $3 = {s = {s = 0x31590014 Address 0x31590014 out of bounds, len =
 275}, next = 0xb0b5fd00}

  (gdb) p *dlginfo-pubruris_caller-next

  $4 = {s = {s = 0x3a70006e Address 0x3a70006e out of bounds, len =
 275}, next = 0x0}


  In config, for pua_dialoginfo we are enabling the option
 use_pubruri_avps and setting pubruri_caller_avp and
 pubruri_callee_avp accordingly.

  Therefore, in pua_dialoginfo.c it is using get_str_list() function to
 set dlginfo-pubruris_caller from the avp.

  Could this be some race condition or something completely different?

  Thanks in advance,

  Charles



 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered
 office: Faraday Wharf, Innovation Birmingham Campus, Holt Street,
 Birmingham Science Park, Birmingham B7 4BB.

 ___
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 listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


 --
 Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/miconda
 Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
 Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA


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Re: [SR-Users] [sr-dev] rfc: distributing dialog profiles

[Charles Chance]

Alex,

Thanks for expanding in such detail. I appreciate you taking the time to
fix the things that I missed. Sometimes I think it may have been easier to
start from scratch ;)

Best regards,

Charles
On 25 Aug 2014 10:11, Alex Hermann a...@speakup.nl wrote:

 On Friday 22 August 2014, Charles Chance wrote:
  On 22 August 2014 16:46, Alex Hermann a...@speakup.nl wrote:
   Last week, i just built profile synchronisation in the dialog module,
   based on
   dmq. It took quite a bit of debugging time because of the state dmq was
   in.
 
  Can you expand a little on the state dmq was in?

 I was hoping to use it as-is, but i encountered issues which had to be
 resolved before i could even use the module:

 - As soon as i enabled the dmq module, i experienced segfaults.
 - It had bad interaction with the maxfwd module
 - Status updates between hosts were largely ignored.
 - The configured server_address wasn't used to send messages.



   It still has some rough edges, but i'll try to push a branch (shortly
   after)
   this weekend for review.
 
  Looking forward to seeing it - may save me the time :)

 I pushed my WIP to the branch alexh/dialog-sync-wip which also contains
 dialog and dmq fixes and cleanups.

 It's WIP, so it might still change. This branch is only compile-tested so
 far, because i normally develop against 3.2. I just cherry-picked most of
 my
 patches to master.

 Known issues:
  - Sync get off under load, cause unknown yet, but probably because of out-
 of-order sync messages.

 Still on the TODO list:
  - Delete 'disabled'  dmq hosts
  - Cope better with out-of-order sync messages
  - Sync initial state
  - Clean shutdown of DMQ, free all memory
  - More efficient protocol instead of JSON. Probably just write raw data in
 the packet, so the receiving side can just use a pointer into the buf
 instead of having to copy everything.

 --
 Alex



 --
 Alex Hermann


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[SR-Users] Help analysing segmentation fault

[Charles Chance]

Hi All,

I wonder if some one could help me to diagnose a recurring issue?

It happens at random times/intervals and under varying load. But always,
just before the time of crash, I see the same critical error in log:

CRITICAL: dialog [dlg_hash.c:841]: log_next_state_dlg(): bogus event 6 in
state 1 for dlg 0xb0632134 [1367:5814] with clid '
0695dd7a346188dd24e7520e6c010...@sip.sipcentric.com' and tags 'as77c89620'
''


Analysing the core dump reveals:

Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio.pid -m 128
-M 4 -u kamailio -g kamailio'.
Program terminated with signal 11, Segmentation fault.
#0  0x081a737c in parse_uri (buf=0x3a70006e Address 0x3a70006e out of
bounds, len=275, uri=0xbfa2fd2c) at parser/parse_uri.c:389
389 scheme=buf[0]+(buf[1]8)+(buf[2]16)+(buf[3]24);

(gdb) frame 1

#1  0x008fe5dd in dialog_publish (state=0x903f37 Trying, ruri=0xb0b5fd00,
entity=0xb0632188, peer=0xb0632190, callid=0xb0632180, initiator=1,
lifetime=7200, localtag=0x0, remotetag=0x0, localtarget=0x0,
remotetarget=0x0, do_pubruri_localcheck=1) at dialog_publish.c:275
275 if (parse_uri(ruri-s, ruri-len, ruri_uri)  0) {

(gdb) p *ruri

$1 = {s = 0x3a70006e Address 0x3a70006e out of bounds, len = 275}

(gdb) up

#2  0x008ff277 in dialog_publish_multi (state=0x903f37 Trying,
ruris=0xb0b5fd00, entity=0xb0632188, peer=0xb0632190, callid=0xb0632180,
initiator=1, lifetime=7200, localtag=0x0, remotetag=0x0,
localtarget=0x0, remotetarget=0x0, do_pubruri_localcheck=1) at
dialog_publish.c:387
387
dialog_publish(state,(ruris-s),entity,peer,callid,initiator,lifetime,localtag,remotetag,localtarget,remotetarget,do_pubruri_localcheck);

(gdb) p *ruris

$2 = {s = {s = 0x3a70006e Address 0x3a70006e out of bounds, len = 275},
next = 0x0}

(gdb) up

#3  0x0090187a in __dialog_created (dlg=0xb0632134, type=2,
_params=0x6db064) at pua_dialoginfo.c:470
470 dialog_publish_multi(Trying, dlginfo-pubruris_caller,
(dlg-from_uri), (include_req_uri)?(dlg-req_uri):(dlg-to_uri),
(dlg-callid), 1, dlginfo-lifetime, 0, 0, 0, 0,
send_publish_flag==-1?1:0);

(gdb) p *dlginfo-pubruris_caller

$3 = {s = {s = 0x31590014 Address 0x31590014 out of bounds, len = 275},
next = 0xb0b5fd00}

(gdb) p *dlginfo-pubruris_caller-next

$4 = {s = {s = 0x3a70006e Address 0x3a70006e out of bounds, len = 275},
next = 0x0}


In config, for pua_dialoginfo we are enabling the option use_pubruri_avps
and setting pubruri_caller_avp and pubruri_callee_avp accordingly.

Therefore, in pua_dialoginfo.c it is using get_str_list() function to set
dlginfo-pubruris_caller from the avp.

Could this be some race condition or something completely different?

Thanks in advance,

Charles

-- 
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office: Faraday Wharf, Innovation Birmingham Campus, Holt Street, 
Birmingham Science Park, Birmingham B7 4BB.
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Re: [SR-Users] Help analysing segmentation fault

[Charles Chance]

Thanks, Daniel.

It can be hours, days or weeks between occurrences,  but I will report back
after a day or two initially then continue to monitor.

Cheers,

Charles
 On 22 Aug 2014 12:18, Daniel-Constantin Mierla mico...@gmail.com wrote:

  Hello,

 can you try this small patch?

 diff --git a/modules/pua_dialoginfo/pua_dialoginfo.c
 b/modules/pua_dialoginfo/pua_dialoginfo.c
 index 1e88a04..0f02b2b 100644
 --- a/modules/pua_dialoginfo/pua_dialoginfo.c
 +++ b/modules/pua_dialoginfo/pua_dialoginfo.c
 @@ -347,7 +347,7 @@ struct str_list* get_str_list(unsigned short
 avp_flags, int_str avp_name) {

 memset( list_current, 0, len);

 -   list_current-s.s = (char*)( (void*) list_current +
 sizeof(struct str_list));
 +   list_current-s.s = (char*)list_current + sizeof(struct
 str_list);
 list_current-s.len = avp_value.s.len;
 memcpy(list_current-s.s,avp_value.s.s,avp_value.s.len);

 It is for 4.1.

 I have some ongoing work to commit soon on the master branch. if you
 confirm it is working fine, I will push this patch as well and backport to
 4.1.

 Cheers,
 Daniel

 On 22/08/14 13:03, Charles Chance wrote:

 Hi All,

  I wonder if some one could help me to diagnose a recurring issue?

  It happens at random times/intervals and under varying load. But always,
 just before the time of crash, I see the same critical error in log:

  CRITICAL: dialog [dlg_hash.c:841]: log_next_state_dlg(): bogus event 6
 in state 1 for dlg 0xb0632134 [1367:5814] with clid '
 0695dd7a346188dd24e7520e6c010...@sip.sipcentric.com' and tags
 'as77c89620' ''


  Analysing the core dump reveals:

  Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio.pid -m
 128 -M 4 -u kamailio -g kamailio'.
 Program terminated with signal 11, Segmentation fault.
 #0  0x081a737c in parse_uri (buf=0x3a70006e Address 0x3a70006e out of
 bounds, len=275, uri=0xbfa2fd2c) at parser/parse_uri.c:389
 389 scheme=buf[0]+(buf[1]8)+(buf[2]16)+(buf[3]24);

  (gdb) frame 1

  #1  0x008fe5dd in dialog_publish (state=0x903f37 Trying,
 ruri=0xb0b5fd00, entity=0xb0632188, peer=0xb0632190, callid=0xb0632180,
 initiator=1, lifetime=7200, localtag=0x0, remotetag=0x0, localtarget=0x0,
 remotetarget=0x0, do_pubruri_localcheck=1) at dialog_publish.c:275
 275 if (parse_uri(ruri-s, ruri-len, ruri_uri)  0) {

  (gdb) p *ruri

   $1 = {s = 0x3a70006e Address 0x3a70006e out of bounds, len = 275}

  (gdb) up

  #2  0x008ff277 in dialog_publish_multi (state=0x903f37 Trying,
 ruris=0xb0b5fd00, entity=0xb0632188, peer=0xb0632190, callid=0xb0632180,
 initiator=1, lifetime=7200, localtag=0x0, remotetag=0x0,
 localtarget=0x0, remotetarget=0x0, do_pubruri_localcheck=1) at
 dialog_publish.c:387
 387
 dialog_publish(state,(ruris-s),entity,peer,callid,initiator,lifetime,localtag,remotetag,localtarget,remotetarget,do_pubruri_localcheck);

  (gdb) p *ruris

  $2 = {s = {s = 0x3a70006e Address 0x3a70006e out of bounds, len =
 275}, next = 0x0}

  (gdb) up

  #3  0x0090187a in __dialog_created (dlg=0xb0632134, type=2,
 _params=0x6db064) at pua_dialoginfo.c:470
 470 dialog_publish_multi(Trying, dlginfo-pubruris_caller,
 (dlg-from_uri), (include_req_uri)?(dlg-req_uri):(dlg-to_uri),
 (dlg-callid), 1, dlginfo-lifetime, 0, 0, 0, 0,
 send_publish_flag==-1?1:0);

  (gdb) p *dlginfo-pubruris_caller

  $3 = {s = {s = 0x31590014 Address 0x31590014 out of bounds, len =
 275}, next = 0xb0b5fd00}

  (gdb) p *dlginfo-pubruris_caller-next

  $4 = {s = {s = 0x3a70006e Address 0x3a70006e out of bounds, len =
 275}, next = 0x0}


  In config, for pua_dialoginfo we are enabling the option
 use_pubruri_avps and setting pubruri_caller_avp and
 pubruri_callee_avp accordingly.

  Therefore, in pua_dialoginfo.c it is using get_str_list() function to
 set dlginfo-pubruris_caller from the avp.

  Could this be some race condition or something completely different?

  Thanks in advance,

  Charles



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Re: [SR-Users] [sr-dev] rfc: distributing dialog profiles

[Charles Chance]

Hi,

On 22 August 2014 16:46, Alex Hermann a...@speakup.nl wrote:

 On Friday 22 August 2014 16:52:59 Daniel-Constantin Mierla wrote:
  The next step is to build the notification system between kamailio
  instances. There are couple of options, as well as questions, that I
  want to discuss before going to implement the mechanism. There are two
  major aspects to care of:
 
  A) Distribution, I thought of two options for it:
 
  1) using dmq module to publish the operations done with local profiles
  (add/remove) -- this seems to be the natural choice right now, Charles
  Chance is also willing to put effort in this direction
 
  This will be coded inside the module, so config won't be affected much
  (eventually some extra parameters of dialog flags).

 Last week, i just built profile synchronisation in the dialog module,
 based on
 dmq. It took quite a bit of debugging time because of the state dmq was in.


Can you expand a little on the state dmq was in?



 It still has some rough edges, but i'll try to push a branch (shortly
 after)
 this weekend for review.


Looking forward to seeing it - may save me the time :)


 --
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Re: [SR-Users] Fwd: Early dialog

[Charles Chance]

Hi,

To my knowledge, only confirmed dialogs are ever persisted to the database.

Regards,

Charles



On 28 April 2014 15:10, Ivan Baques Garcia ivan.baq...@gmail.com wrote:

 Hi,

 I'm using modparam(dialog, db_mode, 1), but only confirmed dialogs are
 shown in the database.

 How I can enable to show unconfirmed dialogs in the database? Now are only
 in-memory.

 Best regards


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Re: [SR-Users] dialog list query

[Charles Chance]

Hi John,

For the current call, you can access dialog attributes via the $dlg(...)
pseudo-variable. But I don't believe there is a way to access other dialogs
from within the config script without querying the database (and setting
db_mode to 1).

If you can explain your use case slightly, someone may be able to suggest
another way to achieve it.

Regards,

Charles



On 2 February 2014 15:40, John Murray john.mur...@skyracktelecom.comwrote:

 Charles,

 Thanks For that.

 How do I get the callid from a call in a specific profile if it is stored
 in memory?

 Regards.

 John
 On 2 Feb 2014 15:05, Charles Chance charles.cha...@sipcentric.com
 wrote:

 Hi John,

 
  What db_mode are you using for dialog module?
 
  Mode 5. I Don't need to persist the dialog state so means table
 maintained in memory and not flushed to disk until shutdown but I can
 access it through normal avp_db_query right?

 Do you mean mode 3? Anyway, avp_db_query is for reading from the database
 - if you're not persisting dialogs to db in realtime then there'll be
 nothing to read.

  Are the entries in the db if you check manually?
 
  If I do sercmd dlg.dlg_list I see the call fine including callid etc.

 dlg.list shows all dialogs in memory, not in db.

 Regards,

 Charles

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 Follow us on twitter @sipcentric http://twitter.com/sipcentric

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 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
 B7 4EJ.



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Re: [SR-Users] dialog list query

[Charles Chance]

Hi John,


 What db_mode are you using for dialog module?

 Mode 5. I Don't need to persist the dialog state so means table
maintained in memory and not flushed to disk until shutdown but I can
access it through normal avp_db_query right?

Do you mean mode 3? Anyway, avp_db_query is for reading from the database -
if you're not persisting dialogs to db in realtime then there'll be nothing
to read.

 Are the entries in the db if you check manually?

 If I do sercmd dlg.dlg_list I see the call fine including callid etc.

dlg.list shows all dialogs in memory, not in db.

Regards,

Charles

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Re: [SR-Users] dialog list query

[Charles Chance]

John,

What db_mode are you using for dialog module? Are the entries in the db if
you check manually?

Are you able to fetch other data using avp_db_query? And do you see any
errors in the log?

Regards,

Charles
 On 1 Feb 2014 16:24, John Murray john.mur...@skyracktelecom.com wrote:

 Hello,



 I am managing calls with the dialog module in kamailio 4.0.4.



 However if is try to get the callid and from_tag of existing calls using:



 avp_db_query(select callid, from_tag from dialog, $avp(s:s_callid),
 $avp(s:s_from_tag));



 I get NULL, yet if I use:



 sercmd proxy dlg.list



 I get the calls listed correctly.



 What am I doing wrong?



 Thanks



 John

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Re: [SR-Users] Replicate registration

[Charles Chance]

Hi Volkan,

You probably should take a look at the Path module. For UACs behind NAT,
the INVITEs should be proxied via the server with which the client is
registered.

Best regards,

Charles



On 23 January 2014 21:41, Volkan Oransoy voran...@gmail.com wrote:

 First of all thank you for your replies.

 I think I am quite close to solution. But still have NAT issues in some
 cases.

 I have added a bunch of code to the standard kamailio script.

 I defined a new flag
 #!define SOCK_FLAG 8

 added these parameters to registrar module.
 modparam(registrar, sock_flag, SOCK_FLAG)
 modparam(registrar, sock_hdr_name, Sock-Info)

 set this flag in the nat test section
 if (nat_uac_test(19)) {
 if (is_method(REGISTER)) {
 fix_nated_register();
 } else {
 fix_nated_contact();
 }
 setflag(FLT_NATS);
 setflag(SOCK_FLAG);
 }

 and send it to the peer proxy to replicate client registration. I also
 added received parameter to the replicated packet.
 if ($si!=$sel(cfg_get.peer.srv_ip))
 {
 add_rcv_param();
 t_replicate(sip:10.173.187.226:5060
 ;transport=udp);
 }


 Now, the issue is, the original proxy that receives the registration works
 as expected.Replicated proxy, has the location data as expected. But if I
 sent a call over second proxy, UA does not respond. I can see the INVITE
 packet that comes from replicated proxy on my client PC, but UA does not
 respond to INVITE that comes from second proxy.

 AOR:: 1...@test.example.net
 Contact:: sip:1002@10.173.191.72:1031;transport=udp;registering_acc=test_
 example_net Q=
  Expires:: 544
 Callid:: 443bb84315de2e2b637a55ab44c0b759@0:0:0:0:0:0:0:0
  Cseq:: 256
 User-agent:: Jitsi2.4.4997Linux
  Received:: sip:10.173.191.72:1024
 State:: CS_NEW
  Flags:: 0
 Cflag:: 192
  Socket:: udp:10.192.208.194:5060
 Methods:: 4294967295
  Ruid:: uloc-52de6613-3dd4-28
 Reg-Id:: 0
  Last-Keepalive:: 1390512267
 Last-Modified:: 1390512267

 This is the ul show output on original proxy. Replicated proxy has nearly
 same values except Cflag and Socket. On replicated proxy, Cflag is 0 and
 Socket points to the original proxy. Is this output normal? Do I have
 anything missing in the setup?

 Thanks.

 /Volkan


 2014/1/21 Daniel-Constantin Mierla mico...@gmail.com

  Hello,

 run with debug=3 in config file and see if there are log messages that
 gives you more details about what happens.

 Typically, when doing replication, you may need to replicate received and
 sometime socket information -- that can be done via headers and then added
 to avps before saving to location in the second peer. Look at the readme of
 registrar module for those parameters.

 Cheers,
 Daniel


 On 19/01/14 14:05, Volkan Oransoy wrote:

 Hi all,

  I am trying to setup an active-active pair of sip proxies. I have
 connected two kamailio boxes to the same PostgreSQL database and my usrloc
 db_mode is 3. I can see the registration data in the database but ul show
 outputs of two device is not same. I think writing registration data to the
 same db is not enough.

  I have found a method to replicate the registraiton process. With
 t_replicate, I synchronize REGISTER requests to both boxes. But I had to
 disable shared database because of uniqe id validations.

  route[REGISTRAR] {
 if (is_method(REGISTER))
 {
 if(isflagset(FLT_NATS))
 {
 setbflag(FLB_NATB);
 }
 if (!save(location))
 sl_reply_error();
 t_replicate(sip:10.1.1.2:5060;transport=udp);
 exit;
 }
 }



  My media servers randomly sends calls to SIP proxies. If Media server
 send the call to the original registrar then the replicated, call initiates
 normally. But when UA registered in first proxy, and Media server sends the
 call to the send one, even if second one has UA in location table, call
 does not initiate.

  Does anyone have an idea about this kind of an active active setup?

  Thanks in advance. Best regards,

  /Volkan


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Re: [SR-Users] Link Redundancy

[Charles Chance]

Hello,

Perhaps DNS SRV is what you are looking for? See
http://en.m.wikipedia.org/wiki/SRV_record for more info.

Regards,

Charles
 On 15 Jan 2014 22:52, Owais ul Haq owaisul...@hotmail.com wrote:

 Hello,

 I have a kamailio server deployed along with an internet connection and
 database (on a seperate server than the kamailio server) obviously. Problem
 in my scenario is that my internet connection is not comfortably reliable
 and in my environment I just cant help myself in this regard.
 I want to get another internet connection with another public ip, attach
 another kamailio server (of same configurations as the first one) with it
 and then use the same database for both kamailio servers (join both
 kamailio servers to the same database server).
 So if one of my connections is down, calls can use other connection
 attached to other server (take the alternate path) and initialize through
 the second server.
 Now my questions are :

 a) How can I assign two ip's against one domain name ??
 b) How can calls be routed to second IP if kamailio server attached to
 first IP is not responding ??
 c) Is all this really possible ?? Or just a fruit less effort of my mind
 :(

 Cheers.

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Re: [SR-Users] Merry Christmas!

[Charles Chance]

And to you, Daniel.

Season's greetings to one and all!

Cheers,

Charles
 On 24 Dec 2013 20:28, Daniel-Constantin Mierla mico...@gmail.com wrote:

 Almost done with another fruitful year, counting two major releases, the
 first edition of a dedicated conference for Kamailio and presence at over a
 dozen events world wide, it's time to say thanks to everyone that helped
 along the time -- developers, users, businesses investing resources and all
 the friends around the project!

 A Merry Christmas and great winter holidays!

 Cheers,
 Daniel

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Re: [SR-Users] dialog sync via dmq

[Charles Chance]

Hi,

 In dialog module no functions to create/destroy dialogs, they described
 and used in tm module. Do you plan move dialog creation/destroying
 funcs in dialog module? I think it will more usable for replication:
 use same functions for create dialog in tm module (when real sip
 dispatch) and for create fake dialog in dmq(??) for replicate...


To which functions are you referring? There is a difference between dialogs
and transactions.

 Registrations are a different matter. Replication can be achieved
 already by sharing a database or using function such as t_replicate().

 How i may realise it?

Something like:

if (save(location)) {
t_replicate(slave);
}


 Client will register twice in such case?


Why? On slave, if register comes from master, use save_noreply(location).
The client only knows of the master.

Regards,
Charles

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Re: [SR-Users] MSRP module - Ubuntu repo

[Charles Chance]

Hi,

Is the msrp module not included already in the core kamailio package?

Regards,

Charles
On 24 Oct 2013 08:26, Wingsravi R wingsravi...@gmail.com wrote:

 Dear Kamailio'ns,

 I have Installed kamailio V 4.0.3  from Ubuntu 12.04 (precise) repository
 and running successfully.
 now i have to work around MSRP relay module of kamailio sever to achieve
 file transfer between two SIP clients. so, i just tried to install MSRP
 module, but i get logged as below:

 /$ sudo apt-get install kamailio-msrp-modules
 Reading package lists... Done
 Building dependency tree
 Reading state information... Done
 E: Unable to locate package kamailio-msrp-modules

 Is MSRP module is included in Ubuntu repository?
 how can i install it ?

 Thank you and regards,


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Re: [SR-Users] MSRP module - Ubuntu repo

[Charles Chance]

Yes, just add it as a loadmodule in your config. You must also load the
tls module -
http://kamailio.org/docs/modules/stable/modules/msrp.html#idp1653864

Regards,

Charles
 On 24 Oct 2013 10:25, Wingsravi R wingsravi...@gmail.com wrote:

 Hi,

 But i have the msrp.so file in modules library.
 Is that is enough ? with that i can load msrp module in kamailio.config
 and continue ?

 or should i have to install kamailio from sources ?

 Thank you


 On Thu, Oct 24, 2013 at 1:07 PM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hi,

 Is the msrp module not included already in the core kamailio package?

 Regards,

 Charles
 On 24 Oct 2013 08:26, Wingsravi R wingsravi...@gmail.com wrote:

 Dear Kamailio'ns,

 I have Installed kamailio V 4.0.3  from Ubuntu 12.04 (precise)
 repository and running successfully.
 now i have to work around MSRP relay module of kamailio sever to achieve
 file transfer between two SIP clients. so, i just tried to install MSRP
 module, but i get logged as below:

 /$ sudo apt-get install kamailio-msrp-modules
 Reading package lists... Done
 Building dependency tree
 Reading state information... Done
 E: Unable to locate package kamailio-msrp-modules

 Is MSRP module is included in Ubuntu repository?
 how can i install it ?

 Thank you and regards,


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Re: [SR-Users] MSRP module - Ubuntu repo

[Charles Chance]

If all your phones are on the same network, it should not be required for
local calls. However, you will still need to proxy media for external calls
to/from networks on the outside of your NAT.

Sorry, I stated before that you should load the tls module - that was taken
from the development version of the documentation but the current stable
version of the same document (the link I gave you) does not state it. I
don't know which is correct, or if there has been a change recently in the
master branch, as I don't use the msrp module personally. I guess it
depends on your own use case as to whether the module needs to be loaded or
not.

Cheers,

Charles



On 24 October 2013 10:52, Wingsravi R wingsravi...@gmail.com wrote:

 Dear Charles,

 Thank you for the reply,

 And one more doubt that I am running my Kamailio server set-up and SIP
 clients in intranet infrastructure. so, in this case also i need to
 configure Rtp proxy or mediaproxy ?
 if so, which proxy to use (either Rtpproxy or mediaproxy) with kamailio
 server ?

 Regards,
 Ravi


 On Thu, Oct 24, 2013 at 3:03 PM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Yes, just add it as a loadmodule in your config. You must also load the
 tls module -
 http://kamailio.org/docs/modules/stable/modules/msrp.html#idp1653864

 Regards,

 Charles
  On 24 Oct 2013 10:25, Wingsravi R wingsravi...@gmail.com wrote:

 Hi,

 But i have the msrp.so file in modules library.
 Is that is enough ? with that i can load msrp module in kamailio.config
 and continue ?

 or should i have to install kamailio from sources ?

 Thank you


 On Thu, Oct 24, 2013 at 1:07 PM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hi,

 Is the msrp module not included already in the core kamailio package?

 Regards,

 Charles
 On 24 Oct 2013 08:26, Wingsravi R wingsravi...@gmail.com wrote:

 Dear Kamailio'ns,

 I have Installed kamailio V 4.0.3  from Ubuntu 12.04 (precise)
 repository and running successfully.
 now i have to work around MSRP relay module of kamailio sever to
 achieve file transfer between two SIP clients. so, i just tried to install
 MSRP module, but i get logged as below:

 /$ sudo apt-get install kamailio-msrp-modules
 Reading package lists... Done
 Building dependency tree
 Reading state information... Done
 E: Unable to locate package kamailio-msrp-modules

 Is MSRP module is included in Ubuntu repository?
 how can i install it ?

 Thank you and regards,


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 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. 
 Registered
 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
 B7 4EJ.
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 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



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 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered
 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
 B7 4EJ.

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Re: [SR-Users] Communicate with Kamailio through external application

[Charles Chance]

Hi,

You are not doing anything wrong - on very quick inspection of the code it
appears you won't be able to issue the command via the xhttp_rpc module.

So other than changing DB (although I assume you are using MSSQL for a very
specific reason), it looks like you may have to stick to the original
remote SSH connection idea. Until someone has time and inclination to add
the unixodbc support that is.

Unless anyone else can offer some alternative...

Regards,

Charles



On 30 September 2013 08:01, Grant Bagdasarian g...@cm.nl wrote:

 Hello,

 ** **

 I’ve loaded the mi_rpc module and configured the rpc_url param, but I
 can’t issue the kamcmd mi ds_reload command from the web interface.
 Executing the command in the console works just fine, but only when I leave
 the binrpc modparam as default.

 I’m probably doing something wrong.

 ** **

 # - xhttp params --

 modparam(xhttp, url_match, ^/http_rpc)

 ** **

 # - xhttp_rpc params --

 modparam(xhttp_rpc, xhttp_rpc_root, http_rpc)

 ** **

 # - mi_rpc params --

 modparam(mi_rpc,rpc_url,tcp:192.168.1.10:2046)

 ** **

 The mi_rpc documentation says: “rpc_url”. This parameter is required if
 you need to run RPC commands from the MI. It must match a URL to which the
 RPC is bound.

 Is this referring to the binrpc modparam?

 ** **

 Any tips?

 ** **

 *From:* sr-users-boun...@lists.sip-router.org [mailto:
 sr-users-boun...@lists.sip-router.org] *On Behalf Of *Charles Chance
 *Sent:* Saturday, September 28, 2013 9:42 AM

 *To:* Kamailio (SER) - Users Mailing List
 *Subject:* Re: [SR-Users] Communicate with Kamailio through external
 application

 ** **

 Hi,

 ** **

 It would appear then that unixodbc support is not currently available in
 kamctl directly, so you are in fact limited to MI command ds_reload.

 ** **

 For your case, you could try loading MI_RPC module which exports an
 additional RPC command: mi.

 ** **

 Then issue your command like:

 ** **

 kamcmd mi ds_reload

 ** **

 If that works for you, then the same command passed via xhttp_rpc should
 also work.

 ** **

 Cheers,

 ** **

 Charles

 ** **

 ** **

 ** **

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Re: [SR-Users] Communicate with Kamailio through external application

[Charles Chance]

Hi,

Have you added the appropriate listen= and/or port= directive to your
config?

http://www.kamailio.org/wiki/cookbooks/4.0.x/core#listen
http://www.kamailio.org/wiki/cookbooks/4.0.x/core#port

Also make sure you have not disabled TCP.

Regards,

Charles


On 27 September 2013 12:27, Grant Bagdasarian g...@cm.nl wrote:

 What am I doing wrong here?

 ** **

 # - xhttp params --

 modparam(xhttp, url_match, null)

 ** **

 # - xhttp_rpc params --

 modparam(xhttp_rpc, xhttp_rpc_root, http_rpc)

 ** **

 ### Routing Logic 

 ** **

 event_route[xhttp:request] {

xlog(L_INFO, ** RECEIVED
 NEW HTTP REQUEST:  $hu *);

xhttp_reply(200, OK, text/html,htmlbodyOK – Request
 Received/body/html);

 }

 ** **

 It seems like Kamailio isn’t receiving the HTTP requests.

 Im doing a simple HTTP GET (http://192.168.1.10/http_rpc) using my
 browser to see if it hits the event_route.

 From what I understand, the RCP command is in the url of the HTTP request,
 and is extracted using string transformations.

 I’ve never done anything with RPC directly, so maybe I’m just wrong.

 ** **

 Also, I don’t see Kamailio binding to the HTTP port.

 The xhttp_rpc_root param only allows alphanumeric values, so I can’t
 specify a full URL.

 ** **

 ** **

 ** **

 *From:* sr-users-boun...@lists.sip-router.org [mailto:
 sr-users-boun...@lists.sip-router.org] *On Behalf Of *Grant Bagdasarian
 *Sent:* Friday, September 27, 2013 10:04 AM
 *To:* Kamailio (SER) - Users Mailing List
 *Subject:* Re: [SR-Users] Communicate with Kamailio through external
 application

 ** **

 Thanks, I’ll have a look at it. 

 Looks much easier than setting up Ssh connections programmatically. 

 ** **

 *From:* sr-users-boun...@lists.sip-router.org [
 mailto:sr-users-boun...@lists.sip-router.orgsr-users-boun...@lists.sip-router.org]
 *On Behalf Of *Charles Chance
 *Sent:* Wednesday, September 25, 2013 1:12 PM
 *To:* Kamailio (SER) - Users Mailing List
 *Subject:* Re: [SR-Users] Communicate with Kamailio through external
 application

 ** **

 Hi,

 ** **

 Take a look at
 http://www.kamailio.org/docs/modules/stable/modules/xhttp_rpc.html, which
 allows you to receive/handle RPC commands over HTTP.

 ** **

 Then use the appropriate RPC command in dispatcher module to reload table:
 http://www.kamailio.org/docs/modules/stable/modules/dispatcher.html#idp16998512
 

 ** **

 Regards,

 ** **

 Charles

 ** **

 ** **

 On 25 September 2013 11:40, Grant Bagdasarian g...@cm.nl wrote:

 Hello,

  

 When adding rows to the dispatcher table, the table in memory needs to be
 refreshed. Since we have multiple Kamailio instances running on different
 machines, I’d like to automate the process of reloading the table.

 Is there an easy way of doing this with an already existing module, which
 provides easy connectivity from the outside?

 Or do I need to write an application which basically sets up a SSH socket
 connection with the machine and sends the command “kamctl dispatcher
 reload” then exit? Not that hard either, but there might be a finer
 solution available.

  

 Regards,

  

 Grant

  


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 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

 ** **


 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered
 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
 B7 4EJ.

 ___
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 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



-- 
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Follow us on twitter @sipcentric http://twitter.com/sipcentric

Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered 
office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham 
B7 4EJ.
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Re: [SR-Users] Communicate with Kamailio through external application

[Charles Chance]

Hi,

I think you are mixing up the MI and RPC commands.

Try:

kamctl fifo ds_reload

or

kamcmd dispatcher.reload

(note the difference between kamctl and kamcmd)

Regards,

Charles



On 27 September 2013 12:57, Grant Bagdasarian g...@cm.nl wrote:

 What kind of DB handler do I need for the dispatcher module?

 ** **

 When executing the dispatcher.reload command I’m receiving the following
 error: 

 8(10201) ERROR: dispatcher [dispatch.c:725]: ds_load_db(): invalid DB
 handler

 ** **

 root@proxy01:~# kamctl dispatcher reload

 ERROR: This command requires a database engine - none was loaded

 ** **

 My Dispatcher table is loaded using db_unixodbc, because we are using
 MSSQL.

 ** **

 loadmodule mi_fifo.so

 loadmodule kex.so

 loadmodule corex.so

 loadmodule tm.so

 loadmodule tmx.so

 loadmodule sl.so

 loadmodule rr.so

 loadmodule pv.so

 loadmodule maxfwd.so

 loadmodule textops.so

 loadmodule textopsx.so

 loadmodule siputils.so

 loadmodule xlog.so

 loadmodule sanity.so

 loadmodule db_unixodbc.so

 loadmodule sqlops.so

 loadmodule dispatcher.so

 loadmodule siptrace.so

 loadmodule avp.so

 loadmodule cfgutils.so

 loadmodule xhttp.so

 loadmodule xhttp_rpc.so

 ** **

 # - db_unixodbc params -

 modparam(db_unixodbc, ping_interval, 600)

 modparam(db_unixodbc, auto_reconnect, 0)

 modparam(db_unixodbc, use_escape_common, 1)

 ** **

 modparam(dispatcher, db_url,unixodbc://domain\
 myusername:mypassword@10.0.0.1/Kamailio)

 modparam(dispatcher, table_name, Dispatcher)

 modparam(dispatcher, setid_col, SetID)

 modparam(dispatcher, destination_col, Destination)

 modparam(dispatcher, priority_col, [Priority])

 modparam(dispatcher, flags_col, Flags)

 modparam(dispatcher, attrs_col, Attributes)

 modparam(dispatcher, flags, 2)

 modparam(dispatcher, dst_avp, $avp(dsdst))

 modparam(dispatcher, grp_avp, $avp(dsgrp))

 modparam(dispatcher, cnt_avp, $avp(dscnt))

 modparam(dispatcher, attrs_avp, $avp(dsattrs))

 ** **

 ** **

 The Dispatcher table is loaded just fine on startup of Kamailio.

 kamctl fifo ds_list shows the table data. 

 ** **

 ** **

 ** **

 *From:* sr-users-boun...@lists.sip-router.org [mailto:
 sr-users-boun...@lists.sip-router.org] *On Behalf Of *Grant Bagdasarian
 *Sent:* Friday, September 27, 2013 1:39 PM

 *To:* Kamailio (SER) - Users Mailing List
 *Subject:* Re: [SR-Users] Communicate with Kamailio through external
 application

 ** **

 Right! Totally forgot about that.

 ** **

 Thanks, Kamailio is now receiving the requests!

 ** **

 *From:* sr-users-boun...@lists.sip-router.org [
 mailto:sr-users-boun...@lists.sip-router.orgsr-users-boun...@lists.sip-router.org]
 *On Behalf Of *Charles Chance
 *Sent:* Friday, September 27, 2013 1:35 PM
 *To:* Kamailio (SER) - Users Mailing List
 *Subject:* Re: [SR-Users] Communicate with Kamailio through external
 application

 ** **

 Hi,

 ** **

 Have you added the appropriate listen= and/or port= directive to your
 config?

 ** **

 http://www.kamailio.org/wiki/cookbooks/4.0.x/core#listen

 http://www.kamailio.org/wiki/cookbooks/4.0.x/core#port

 ** **

 Also make sure you have not disabled TCP.

 ** **

 Regards,

 ** **

 Charles

 ** **

 ** **

 On 27 September 2013 12:27, Grant Bagdasarian g...@cm.nl wrote:

 What am I doing wrong here?

  

 # - xhttp params --

 modparam(xhttp, url_match, null)

  

 # - xhttp_rpc params --

 modparam(xhttp_rpc, xhttp_rpc_root, http_rpc)

  

 ### Routing Logic 

  

 event_route[xhttp:request] {

xlog(L_INFO, ** RECEIVED
 NEW HTTP REQUEST:  $hu *);

xhttp_reply(200, OK, text/html,htmlbodyOK – Request
 Received/body/html);

 }

  

 It seems like Kamailio isn’t receiving the HTTP requests.

 Im doing a simple HTTP GET (http://192.168.1.10/http_rpc) using my
 browser to see if it hits the event_route.

 From what I understand, the RCP command is in the url of the HTTP request,
 and is extracted using string transformations.

 I’ve never done anything with RPC directly, so maybe I’m just wrong.

  

 Also, I don’t see Kamailio binding to the HTTP port.

 The xhttp_rpc_root param only allows alphanumeric values, so I can’t
 specify a full URL.

  

  

  

 *From:* sr-users-boun...@lists.sip-router.org [mailto:
 sr-users-boun...@lists.sip-router.org] *On Behalf Of *Grant Bagdasarian
 *Sent:* Friday, September 27, 2013 10:04 AM
 *To:* Kamailio (SER) - Users Mailing List
 *Subject:* Re: [SR-Users] Communicate with Kamailio through external
 application

  

 Thanks, I’ll have a look at it. 

 Looks much easier than setting up Ssh

Re: [SR-Users] Communicate with Kamailio through external application

[Charles Chance]

Hi,

Regarding kamcmd failure - do you have the ctl module loaded?

Regards,

Charles



On 27 September 2013 13:18, Grant Bagdasarian g...@cm.nl wrote:

 Hello,

 ** **

 The kamctl fifo ds_reload command works when I use it in putty.

 The kamcmd dispatcher.reload command does NOT work.

 ** **

 Gives me the following error:

 root@proxy01:~# kamcmd dispatcher.reload

 ERROR: connect_unix_sock: connect(/tmp/kamailio_ctl): Connection refused
 [111]

 ** **


 http://192.168.1.10/http_rpc/dispatcher/dispatcher.reload?arg=dispatcher.reload
 

 ** **

 10(10232) ERROR: dispatcher [dispatch.c:725]: ds_load_db(): invalid DB
 handler

 ….

 HTTP/1.1 500 Reload Failed

 Via: SIP/2.0/TCP 192.168.1.11:51065

 Content-Type: text/html

 Server: kamailio (4.0.2 (x86_64/linux))

 Content-Length: 2095

 ** **

 *From:* sr-users-boun...@lists.sip-router.org [mailto:
 sr-users-boun...@lists.sip-router.org] *On Behalf Of *Charles Chance
 *Sent:* Friday, September 27, 2013 2:10 PM

 *To:* Kamailio (SER) - Users Mailing List
 *Subject:* Re: [SR-Users] Communicate with Kamailio through external
 application

 ** **

 Hi,

 ** **

 I think you are mixing up the MI and RPC commands.

 ** **

 Try:

 ** **

 kamctl fifo ds_reload

 ** **

 or

 ** **

 kamcmd dispatcher.reload

 ** **

 (note the difference between kamctl and kamcmd)

 ** **

 Regards,

 ** **

 Charles

 ** **

 ** **

 On 27 September 2013 12:57, Grant Bagdasarian g...@cm.nl wrote:

 What kind of DB handler do I need for the dispatcher module?

  

 When executing the dispatcher.reload command I’m receiving the following
 error: 

 8(10201) ERROR: dispatcher [dispatch.c:725]: ds_load_db(): invalid DB
 handler

  

 root@proxy01:~# kamctl dispatcher reload

 ERROR: This command requires a database engine - none was loaded

  

 My Dispatcher table is loaded using db_unixodbc, because we are using
 MSSQL.

  

 loadmodule mi_fifo.so

 loadmodule kex.so

 loadmodule corex.so

 loadmodule tm.so

 loadmodule tmx.so

 loadmodule sl.so

 loadmodule rr.so

 loadmodule pv.so

 loadmodule maxfwd.so

 loadmodule textops.so

 loadmodule textopsx.so

 loadmodule siputils.so

 loadmodule xlog.so

 loadmodule sanity.so

 loadmodule db_unixodbc.so

 loadmodule sqlops.so

 loadmodule dispatcher.so

 loadmodule siptrace.so

 loadmodule avp.so

 loadmodule cfgutils.so

 loadmodule xhttp.so

 loadmodule xhttp_rpc.so

  

 # - db_unixodbc params -

 modparam(db_unixodbc, ping_interval, 600)

 modparam(db_unixodbc, auto_reconnect, 0)

 modparam(db_unixodbc, use_escape_common, 1)

  

 modparam(dispatcher, db_url,unixodbc://domain\
 myusername:mypassword@10.0.0.1/Kamailio)

 modparam(dispatcher, table_name, Dispatcher)

 modparam(dispatcher, setid_col, SetID)

 modparam(dispatcher, destination_col, Destination)

 modparam(dispatcher, priority_col, [Priority])

 modparam(dispatcher, flags_col, Flags)

 modparam(dispatcher, attrs_col, Attributes)

 modparam(dispatcher, flags, 2)

 modparam(dispatcher, dst_avp, $avp(dsdst))

 modparam(dispatcher, grp_avp, $avp(dsgrp))

 modparam(dispatcher, cnt_avp, $avp(dscnt))

 modparam(dispatcher, attrs_avp, $avp(dsattrs))

  

  

 The Dispatcher table is loaded just fine on startup of Kamailio.

 kamctl fifo ds_list shows the table data. 

  

  

  

 *From:* sr-users-boun...@lists.sip-router.org [mailto:
 sr-users-boun...@lists.sip-router.org] *On Behalf Of *Grant Bagdasarian
 *Sent:* Friday, September 27, 2013 1:39 PM


 *To:* Kamailio (SER) - Users Mailing List
 *Subject:* Re: [SR-Users] Communicate with Kamailio through external
 application

  

 Right! Totally forgot about that.

  

 Thanks, Kamailio is now receiving the requests!

  

 *From:* sr-users-boun...@lists.sip-router.org [
 mailto:sr-users-boun...@lists.sip-router.orgsr-users-boun...@lists.sip-router.org]
 *On Behalf Of *Charles Chance
 *Sent:* Friday, September 27, 2013 1:35 PM
 *To:* Kamailio (SER) - Users Mailing List
 *Subject:* Re: [SR-Users] Communicate with Kamailio through external
 application

  

 Hi,

  

 Have you added the appropriate listen= and/or port= directive to your
 config?

  

 http://www.kamailio.org/wiki/cookbooks/4.0.x/core#listen

 http://www.kamailio.org/wiki/cookbooks/4.0.x/core#port

  

 Also make sure you have not disabled TCP.

  

 Regards,

  

 Charles

  

  

 On 27 September 2013 12:27, Grant Bagdasarian g...@cm.nl wrote:

 What am I doing wrong here?

  

 # - xhttp params --

 modparam(xhttp, url_match, null)

  

 # - xhttp_rpc params --

 modparam(xhttp_rpc

Re: [SR-Users] Communicate with Kamailio through external application

[Charles Chance]

Hi,

So now only the one error. Which version of Kamailio are you running?

Regards,

Charles



On 27 September 2013 13:44, Grant Bagdasarian g...@cm.nl wrote:

 Hello,

 ** **

 I didn’t, but I just loaded it, but I haven’t set any modparams yet. 

 ** **

 Now it’s giving me more error output:

 ** **

 root@prox01:~# kamcmd dispatcher.reload

 8(11010) DEBUG: ctl [../../io_wait.h:390]: io_watch_add(): DBG:
 io_watch_add(0x7f9211dab440, 11, 3, 0xa02cf0), fd_no=1

 8(11010) DEBUG: ctl [io_listener.c:453]: handle_new_connect():
 handle_stream read: new connection (1) on /tmp/kamailio_ctl

 8(11010) ERROR: dispatcher [dispatch.c:725]: ds_load_db(): invalid DB
 handler

 error: 500 - Reload Failed

 8(11010) DEBUG: ctl [io_listener.c:506]: handle_stream_read():
 handle_stream read: eof on /tmp/kamailio_ctl

 8(11010) DEBUG: ctl [../../io_wait.h:617]: io_watch_del(): DBG:
 io_watch_del (0x7f9211dab440, 11, -1, 0x10) fd_no=2 called

 ** **

 *From:* sr-users-boun...@lists.sip-router.org [mailto:
 sr-users-boun...@lists.sip-router.org] *On Behalf Of *Charles Chance
 *Sent:* Friday, September 27, 2013 2:36 PM

 *To:* Kamailio (SER) - Users Mailing List
 *Subject:* Re: [SR-Users] Communicate with Kamailio through external
 application

 ** **

 Hi,

 ** **

 Regarding kamcmd failure - do you have the ctl module loaded?

 ** **

 Regards,

 ** **

 Charles

 ** **

 ** **

 On 27 September 2013 13:18, Grant Bagdasarian g...@cm.nl wrote:

 Hello,

  

 The kamctl fifo ds_reload command works when I use it in putty.

 The kamcmd dispatcher.reload command does NOT work.

  

 Gives me the following error:

 root@proxy01:~# kamcmd dispatcher.reload

 ERROR: connect_unix_sock: connect(/tmp/kamailio_ctl): Connection refused
 [111]

  


 http://192.168.1.10/http_rpc/dispatcher/dispatcher.reload?arg=dispatcher.reload
 

  

 10(10232) ERROR: dispatcher [dispatch.c:725]: ds_load_db(): invalid DB
 handler

 ….

 HTTP/1.1 500 Reload Failed

 Via: SIP/2.0/TCP 192.168.1.11:51065

 Content-Type: text/html

 Server: kamailio (4.0.2 (x86_64/linux))

 Content-Length: 2095

  

 *From:* sr-users-boun...@lists.sip-router.org [mailto:
 sr-users-boun...@lists.sip-router.org] *On Behalf Of *Charles Chance
 *Sent:* Friday, September 27, 2013 2:10 PM


 *To:* Kamailio (SER) - Users Mailing List
 *Subject:* Re: [SR-Users] Communicate with Kamailio through external
 application

  

 Hi,

  

 I think you are mixing up the MI and RPC commands.

  

 Try:

  

 kamctl fifo ds_reload

  

 or

  

 kamcmd dispatcher.reload

  

 (note the difference between kamctl and kamcmd)

  

 Regards,

  

 Charles

  

  

 On 27 September 2013 12:57, Grant Bagdasarian g...@cm.nl wrote:

 What kind of DB handler do I need for the dispatcher module?

  

 When executing the dispatcher.reload command I’m receiving the following
 error: 

 8(10201) ERROR: dispatcher [dispatch.c:725]: ds_load_db(): invalid DB
 handler

  

 root@proxy01:~# kamctl dispatcher reload

 ERROR: This command requires a database engine - none was loaded

  

 My Dispatcher table is loaded using db_unixodbc, because we are using
 MSSQL.

  

 loadmodule mi_fifo.so

 loadmodule kex.so

 loadmodule corex.so

 loadmodule tm.so

 loadmodule tmx.so

 loadmodule sl.so

 loadmodule rr.so

 loadmodule pv.so

 loadmodule maxfwd.so

 loadmodule textops.so

 loadmodule textopsx.so

 loadmodule siputils.so

 loadmodule xlog.so

 loadmodule sanity.so

 loadmodule db_unixodbc.so

 loadmodule sqlops.so

 loadmodule dispatcher.so

 loadmodule siptrace.so

 loadmodule avp.so

 loadmodule cfgutils.so

 loadmodule xhttp.so

 loadmodule xhttp_rpc.so

  

 # - db_unixodbc params -

 modparam(db_unixodbc, ping_interval, 600)

 modparam(db_unixodbc, auto_reconnect, 0)

 modparam(db_unixodbc, use_escape_common, 1)

  

 modparam(dispatcher, db_url,unixodbc://domain\
 myusername:mypassword@10.0.0.1/Kamailio)

 modparam(dispatcher, table_name, Dispatcher)

 modparam(dispatcher, setid_col, SetID)

 modparam(dispatcher, destination_col, Destination)

 modparam(dispatcher, priority_col, [Priority])

 modparam(dispatcher, flags_col, Flags)

 modparam(dispatcher, attrs_col, Attributes)

 modparam(dispatcher, flags, 2)

 modparam(dispatcher, dst_avp, $avp(dsdst))

 modparam(dispatcher, grp_avp, $avp(dsgrp))

 modparam(dispatcher, cnt_avp, $avp(dscnt))

 modparam(dispatcher, attrs_avp, $avp(dsattrs))

  

  

 The Dispatcher table is loaded just fine on startup of Kamailio.

 kamctl fifo ds_list shows the table data. 

  

  

  

 *From:* sr-users

Re: [SR-Users] Communicate with Kamailio through external application

[Charles Chance]

Hi,

Take a look at
http://www.kamailio.org/docs/modules/stable/modules/xhttp_rpc.html, which
allows you to receive/handle RPC commands over HTTP.

Then use the appropriate RPC command in dispatcher module to reload table:
http://www.kamailio.org/docs/modules/stable/modules/dispatcher.html#idp16998512

Regards,

Charles



On 25 September 2013 11:40, Grant Bagdasarian g...@cm.nl wrote:

 Hello,

 ** **

 When adding rows to the dispatcher table, the table in memory needs to be
 refreshed. Since we have multiple Kamailio instances running on different
 machines, I’d like to automate the process of reloading the table.

 Is there an easy way of doing this with an already existing module, which
 provides easy connectivity from the outside?

 Or do I need to write an application which basically sets up a SSH socket
 connection with the machine and sends the command “kamctl dispatcher
 reload” then exit? Not that hard either, but there might be a finer
 solution available.

 ** **

 Regards,

 ** **

 Grant

 ** **

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 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



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Follow us on twitter @sipcentric http://twitter.com/sipcentric

Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered 
office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham 
B7 4EJ.
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Re: [SR-Users] Calling users on another kamailio server

[Charles Chance]

Hi,

When you perform lookup and t_relay, the invite will be sent via the proxy
which added the path header, not forwarded to the registrar that performed
the save. The To header will contain the callee uri, but the destination
($du) will be set to the appropriate proxy so it can forward it on.

This is the same whether the registrar and proxy functions are on separate
servers or not. Logically, there is no difference. If on the same server,
you just have to separate the routing logic according to where the message
came from.

Regards,

Charles
On 16 Sep 2013 18:33, Brian Wallen bwal...@gmail.com wrote:

 I'm thinking about changing the architecture of the system I'm building.
 As it stands now, my registrars are on the edge of the network and they
 share a DB. If a user calls another user on another server, the callers
 registrar sends an INVITE to the callee and the caller's rtpproxy is used
 by both the caller and callee. So my first question is, should it be set up
 like this? Or should the caller's registrar hand the INVITE to the callee's
 registrar instead of the callee directly?

 That leads me to my change. I'm thinking about adding a proxy at the edge
 in front of each registrar. On the proxies I'd use add_path() on each
 REGISTER before sending them to the registrar. Do you guys think this would
 be a better way of setting it up? Correct me if I'm wrong but I think if I
 do it this way, when an INVITE reaches a proxy it'll do a lookup() and then
 t_relay() to the appropriate registrar, right?


 On Mon, Sep 16, 2013 at 11:06 AM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Peter - calling msg_apply_changes() after add_path() should indeed work,
 I'm sure, so thanks for pointing it out. However, when testing I'm getting
 parse errors in debug output, so am unable to confirm at this time.

 Brian - which version are you using? Anything in log? After calling
 save(), path column in DB is empty?



 On 16 September 2013 15:18, Brian Wallen bwal...@gmail.com wrote:

 In my test setup the registrar is at the edge and add_path() does in
 fact do nothing. I also tried Charles' suggestion which didn't seem to add
 the Path header either.


 On Mon, Sep 16, 2013 at 10:04 AM, Peter Dunkley 
 peter.dunk...@crocodilertc.net wrote:

 I thought append_hf() didn't take affect (unless you use
 msg_apply_changes()) until the message left Kamailio too?

 If that is the case, and msg_apply_changes() is called, doesn't that
 mean the Path: header from add_path() would be added in that scenario?


 On 16 September 2013 14:49, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 It is possible if the edge proxy and registrar are separate, as you
 say. But if the registrar is at the edge with no separate proxy, 
 add_path()
 does nothing (because the message never leaves Kamailio for the header to
 be added).


 On 16 September 2013 14:42, Peter Dunkley 
 peter.dunk...@crocodilertc.net wrote:

 Is there any reason it isn't possible?


 On 16 September 2013 14:21, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hi,

 Yes, you are right - and I agree, it would be better if this was
 possible :)

 Charles




 On 16 September 2013 14:15, Peter Dunkley 
 peter.dunk...@crocodilertc.net wrote:

 Hello,

 It'd be better if the add_path() function could be used here.
  That way, if using outbound (RFC5626), the flow-token (the userinfo 
 part
 of the Path-URI) would be present and there would be no need to add the
 ;received parameter.

 This would address the one issue remaining for SIP outbound on
 Kamailio, which is its use without an edge proxy that is separate from 
 the
 registrar.

 Regards,

 Peter


 On 16 September 2013 14:05, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hi,

 This sounds like a case for sharing same database, and adding Path
 before saving incoming register. That way, no need to replicate 
 register
 message to other servers and all subscribers use the same domain.

 Add path something like this before calling save():

 append_hf(Path: sip:$Ri:$Rp;
 received=sip:$si:$sp;lr\r\n);
 msg_apply_changes();

 Whichever server receives the incoming invite, will perform lookup
 and automatically route to the server which received the register. On 
 the
 proxying server set $du according to received param of route header, 
 add
 record-route, and then t_relay(). As Daniel said, no need to
 re-authenticate or perform lookup again.

 Regards,

 Charles



 On Mon, Sep 16, 2013 at 7:34 AM, Daniel-Constantin Mierla 
 mico...@gmail.com wrote:

 Hello,


 On 9/12/13 10:08 PM, Brian Wallen wrote:

 I currently have two independent kamailio servers. I'd like to
 set them up in a way that user1 on server1 can make a call to user2 
 on
 server2. After searching I've come up with two ways that this might 
 be able
 to be done. Can someone please sanity check these or point me in 
 the right
 direction?

 1. Have one registrar server

Re: [SR-Users] Calling users on another kamailio server

[Charles Chance]

Hi,

Yes, you are right - and I agree, it would be better if this was possible :)

Charles




On 16 September 2013 14:15, Peter Dunkley peter.dunk...@crocodilertc.netwrote:

 Hello,

 It'd be better if the add_path() function could be used here.  That way,
 if using outbound (RFC5626), the flow-token (the userinfo part of the
 Path-URI) would be present and there would be no need to add the
 ;received parameter.

 This would address the one issue remaining for SIP outbound on Kamailio,
 which is its use without an edge proxy that is separate from the registrar.

 Regards,

 Peter


 On 16 September 2013 14:05, Charles Chance 
 charles.cha...@sipcentric.comwrote:

 Hi,

 This sounds like a case for sharing same database, and adding Path before
 saving incoming register. That way, no need to replicate register message
 to other servers and all subscribers use the same domain.

 Add path something like this before calling save():

 append_hf(Path: sip:$Ri:$Rp;
 received=sip:$si:$sp;lr\r\n);
 msg_apply_changes();

 Whichever server receives the incoming invite, will perform lookup and
 automatically route to the server which received the register. On the
 proxying server set $du according to received param of route header, add
 record-route, and then t_relay(). As Daniel said, no need to
 re-authenticate or perform lookup again.

 Regards,

 Charles



 On Mon, Sep 16, 2013 at 7:34 AM, Daniel-Constantin Mierla 
 mico...@gmail.com wrote:

 Hello,


 On 9/12/13 10:08 PM, Brian Wallen wrote:

 I currently have two independent kamailio servers. I'd like to set them
 up in a way that user1 on server1 can make a call to user2 on server2.
 After searching I've come up with two ways that this might be able to be
 done. Can someone please sanity check these or point me in the right
 direction?

 1. Have one registrar server and convert the other server to a proxy
 2. Keep them both as registrars and somehow make them each aware of the
 users on the other server

 I like 2 better because if one server went down users on the other
 server would still be up. The only thing is I don't know how to set the
 servers up to communicate with each other.

 the nat can create problems when a server is down - if the nat is
 symmetric, only the server that received the registrar can send back calls
 to the phone.

 Communication between users on two servers is as simple as using
 t_relay_to(proto:serverip:**port) after you do lookup(location) and
 no record is found.


 Suppose I have three or more kamailio servers. If a call comes in and
 lookup() returns that no record was found, how do I know which server to
 forward to? Is that a case in which I should replicate the database?


 You have to add an extra check for the case the call was coming from the
 other server, not to forward back to it in case of no found again.


 Thanks for the tip, I hadn't thought of that.


 Also, you should skip user authentication for calls from the other
 server (not do authenticate twice). Another aspect to take care is chaining
 rtpproxy, you have to use the flat for trusting the other server (r, iirc).

 You can also replicate the registration, but again, it can add troubles
 to the nat. Look at t_replicate() (in tm module).

 Cheers,
 Daniel

 --
 Daniel-Constantin Mierla - http://www.asipto.com
 http://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/**micondahttp://www.linkedin.com/in/miconda
 Kamailio Advanced Trainings - Berlin, Oct 21-24; Miami, Nov 11-13, 2013
   - more details about Kamailio trainings at http://www.asipto.com -


 __**_
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**usershttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered
 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
 B7 4EJ.

 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




 --
 Peter Dunkley
 Technical Director
 Crocodile RCS Ltd

 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




-- 
*Charles Chance*
Managing Director

t. 0121 285 4400m. 07932 063 891

-- 
www.sipcentric.com

Follow us on twitter

Re: [SR-Users] Calling users on another kamailio server

[Charles Chance]

Hi Brian,

Another tip, if you do share the same database, make sure the clocks are in
sync ;)

Cheers,

Charles



On 16 September 2013 14:22, Brian Wallen bwal...@gmail.com wrote:

 Thanks for the tips guys. I'll try them out today and report back.


 On Mon, Sep 16, 2013 at 9:21 AM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hi,

 Yes, you are right - and I agree, it would be better if this was possible
 :)

 Charles




 On 16 September 2013 14:15, Peter Dunkley peter.dunk...@crocodilertc.net
  wrote:

 Hello,

 It'd be better if the add_path() function could be used here.  That
 way, if using outbound (RFC5626), the flow-token (the userinfo part of the
 Path-URI) would be present and there would be no need to add the
 ;received parameter.

 This would address the one issue remaining for SIP outbound on Kamailio,
 which is its use without an edge proxy that is separate from the registrar.

 Regards,

 Peter


 On 16 September 2013 14:05, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hi,

 This sounds like a case for sharing same database, and adding Path
 before saving incoming register. That way, no need to replicate register
 message to other servers and all subscribers use the same domain.

 Add path something like this before calling save():

 append_hf(Path: sip:$Ri:$Rp;
 received=sip:$si:$sp;lr\r\n);
 msg_apply_changes();

 Whichever server receives the incoming invite, will perform lookup and
 automatically route to the server which received the register. On the
 proxying server set $du according to received param of route header, add
 record-route, and then t_relay(). As Daniel said, no need to
 re-authenticate or perform lookup again.

 Regards,

 Charles



 On Mon, Sep 16, 2013 at 7:34 AM, Daniel-Constantin Mierla 
 mico...@gmail.com wrote:

 Hello,


 On 9/12/13 10:08 PM, Brian Wallen wrote:

 I currently have two independent kamailio servers. I'd like to set
 them up in a way that user1 on server1 can make a call to user2 on 
 server2.
 After searching I've come up with two ways that this might be able to be
 done. Can someone please sanity check these or point me in the right
 direction?

 1. Have one registrar server and convert the other server to a proxy
 2. Keep them both as registrars and somehow make them each aware of
 the users on the other server

 I like 2 better because if one server went down users on the other
 server would still be up. The only thing is I don't know how to set the
 servers up to communicate with each other.

 the nat can create problems when a server is down - if the nat is
 symmetric, only the server that received the registrar can send back calls
 to the phone.

 Communication between users on two servers is as simple as using
 t_relay_to(proto:serverip:**port) after you do lookup(location)
 and no record is found.


 Suppose I have three or more kamailio servers. If a call comes in and
 lookup() returns that no record was found, how do I know which server to
 forward to? Is that a case in which I should replicate the database?


 You have to add an extra check for the case the call was coming from
 the other server, not to forward back to it in case of no found again.


 Thanks for the tip, I hadn't thought of that.


 Also, you should skip user authentication for calls from the other
 server (not do authenticate twice). Another aspect to take care is 
 chaining
 rtpproxy, you have to use the flat for trusting the other server (r, 
 iirc).

 You can also replicate the registration, but again, it can add
 troubles to the nat. Look at t_replicate() (in tm module).

 Cheers,
 Daniel

 --
 Daniel-Constantin Mierla - http://www.asipto.com
 http://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/**micondahttp://www.linkedin.com/in/miconda
 Kamailio Advanced Trainings - Berlin, Oct 21-24; Miami, Nov 11-13, 2013
   - more details about Kamailio trainings at http://www.asipto.com -


 __**_
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**usershttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. 
 Registered
 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
 B7 4EJ.

 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




 --
 Peter Dunkley
 Technical Director

Re: [SR-Users] Calling users on another kamailio server

[Charles Chance]

It is possible if the edge proxy and registrar are separate, as you say.
But if the registrar is at the edge with no separate proxy, add_path() does
nothing (because the message never leaves Kamailio for the header to be
added).


On 16 September 2013 14:42, Peter Dunkley peter.dunk...@crocodilertc.netwrote:

 Is there any reason it isn't possible?


 On 16 September 2013 14:21, Charles Chance 
 charles.cha...@sipcentric.comwrote:

 Hi,

 Yes, you are right - and I agree, it would be better if this was possible
 :)

 Charles




 On 16 September 2013 14:15, Peter Dunkley peter.dunk...@crocodilertc.net
  wrote:

 Hello,

 It'd be better if the add_path() function could be used here.  That
 way, if using outbound (RFC5626), the flow-token (the userinfo part of the
 Path-URI) would be present and there would be no need to add the
 ;received parameter.

 This would address the one issue remaining for SIP outbound on Kamailio,
 which is its use without an edge proxy that is separate from the registrar.

 Regards,

 Peter


 On 16 September 2013 14:05, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hi,

 This sounds like a case for sharing same database, and adding Path
 before saving incoming register. That way, no need to replicate register
 message to other servers and all subscribers use the same domain.

 Add path something like this before calling save():

 append_hf(Path: sip:$Ri:$Rp;
 received=sip:$si:$sp;lr\r\n);
 msg_apply_changes();

 Whichever server receives the incoming invite, will perform lookup and
 automatically route to the server which received the register. On the
 proxying server set $du according to received param of route header, add
 record-route, and then t_relay(). As Daniel said, no need to
 re-authenticate or perform lookup again.

 Regards,

 Charles



 On Mon, Sep 16, 2013 at 7:34 AM, Daniel-Constantin Mierla 
 mico...@gmail.com wrote:

 Hello,


 On 9/12/13 10:08 PM, Brian Wallen wrote:

 I currently have two independent kamailio servers. I'd like to set
 them up in a way that user1 on server1 can make a call to user2 on 
 server2.
 After searching I've come up with two ways that this might be able to be
 done. Can someone please sanity check these or point me in the right
 direction?

 1. Have one registrar server and convert the other server to a proxy
 2. Keep them both as registrars and somehow make them each aware of
 the users on the other server

 I like 2 better because if one server went down users on the other
 server would still be up. The only thing is I don't know how to set the
 servers up to communicate with each other.

 the nat can create problems when a server is down - if the nat is
 symmetric, only the server that received the registrar can send back calls
 to the phone.

 Communication between users on two servers is as simple as using
 t_relay_to(proto:serverip:**port) after you do lookup(location)
 and no record is found.


 Suppose I have three or more kamailio servers. If a call comes in and
 lookup() returns that no record was found, how do I know which server to
 forward to? Is that a case in which I should replicate the database?


 You have to add an extra check for the case the call was coming from
 the other server, not to forward back to it in case of no found again.


 Thanks for the tip, I hadn't thought of that.


 Also, you should skip user authentication for calls from the other
 server (not do authenticate twice). Another aspect to take care is 
 chaining
 rtpproxy, you have to use the flat for trusting the other server (r, 
 iirc).

 You can also replicate the registration, but again, it can add
 troubles to the nat. Look at t_replicate() (in tm module).

 Cheers,
 Daniel

 --
 Daniel-Constantin Mierla - http://www.asipto.com
 http://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/**micondahttp://www.linkedin.com/in/miconda
 Kamailio Advanced Trainings - Berlin, Oct 21-24; Miami, Nov 11-13, 2013
   - more details about Kamailio trainings at http://www.asipto.com -


 __**_
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**usershttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. 
 Registered
 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
 B7 4EJ.

 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip

Re: [SR-Users] Calling users on another kamailio server

[Charles Chance]

Peter - calling msg_apply_changes() after add_path() should indeed work,
I'm sure, so thanks for pointing it out. However, when testing I'm getting
parse errors in debug output, so am unable to confirm at this time.

Brian - which version are you using? Anything in log? After calling save(),
path column in DB is empty?



On 16 September 2013 15:18, Brian Wallen bwal...@gmail.com wrote:

 In my test setup the registrar is at the edge and add_path() does in fact
 do nothing. I also tried Charles' suggestion which didn't seem to add the
 Path header either.


 On Mon, Sep 16, 2013 at 10:04 AM, Peter Dunkley 
 peter.dunk...@crocodilertc.net wrote:

 I thought append_hf() didn't take affect (unless you use
 msg_apply_changes()) until the message left Kamailio too?

 If that is the case, and msg_apply_changes() is called, doesn't that mean
 the Path: header from add_path() would be added in that scenario?


 On 16 September 2013 14:49, Charles Chance charles.cha...@sipcentric.com
  wrote:

 It is possible if the edge proxy and registrar are separate, as you say.
 But if the registrar is at the edge with no separate proxy, add_path() does
 nothing (because the message never leaves Kamailio for the header to be
 added).


 On 16 September 2013 14:42, Peter Dunkley 
 peter.dunk...@crocodilertc.net wrote:

 Is there any reason it isn't possible?


 On 16 September 2013 14:21, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hi,

 Yes, you are right - and I agree, it would be better if this was
 possible :)

 Charles




 On 16 September 2013 14:15, Peter Dunkley 
 peter.dunk...@crocodilertc.net wrote:

 Hello,

 It'd be better if the add_path() function could be used here.  That
 way, if using outbound (RFC5626), the flow-token (the userinfo part of 
 the
 Path-URI) would be present and there would be no need to add the
 ;received parameter.

 This would address the one issue remaining for SIP outbound on
 Kamailio, which is its use without an edge proxy that is separate from 
 the
 registrar.

 Regards,

 Peter


 On 16 September 2013 14:05, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hi,

 This sounds like a case for sharing same database, and adding Path
 before saving incoming register. That way, no need to replicate register
 message to other servers and all subscribers use the same domain.

 Add path something like this before calling save():

 append_hf(Path: sip:$Ri:$Rp;
 received=sip:$si:$sp;lr\r\n);
 msg_apply_changes();

 Whichever server receives the incoming invite, will perform lookup
 and automatically route to the server which received the register. On 
 the
 proxying server set $du according to received param of route header, add
 record-route, and then t_relay(). As Daniel said, no need to
 re-authenticate or perform lookup again.

 Regards,

 Charles



 On Mon, Sep 16, 2013 at 7:34 AM, Daniel-Constantin Mierla 
 mico...@gmail.com wrote:

 Hello,


 On 9/12/13 10:08 PM, Brian Wallen wrote:

 I currently have two independent kamailio servers. I'd like to set
 them up in a way that user1 on server1 can make a call to user2 on 
 server2.
 After searching I've come up with two ways that this might be able to 
 be
 done. Can someone please sanity check these or point me in the right
 direction?

 1. Have one registrar server and convert the other server to a
 proxy
 2. Keep them both as registrars and somehow make them each aware
 of the users on the other server

 I like 2 better because if one server went down users on the other
 server would still be up. The only thing is I don't know how to set 
 the
 servers up to communicate with each other.

 the nat can create problems when a server is down - if the nat is
 symmetric, only the server that received the registrar can send back 
 calls
 to the phone.

 Communication between users on two servers is as simple as using
 t_relay_to(proto:serverip:**port) after you do
 lookup(location) and no record is found.


 Suppose I have three or more kamailio servers. If a call comes in
 and lookup() returns that no record was found, how do I know which 
 server
 to forward to? Is that a case in which I should replicate the database?


 You have to add an extra check for the case the call was coming
 from the other server, not to forward back to it in case of no found 
 again.


 Thanks for the tip, I hadn't thought of that.


 Also, you should skip user authentication for calls from the other
 server (not do authenticate twice). Another aspect to take care is 
 chaining
 rtpproxy, you have to use the flat for trusting the other server (r, 
 iirc).

 You can also replicate the registration, but again, it can add
 troubles to the nat. Look at t_replicate() (in tm module).

 Cheers,
 Daniel

 --
 Daniel-Constantin Mierla - http://www.asipto.com
 http://twitter.com/#!/miconda - http://www.linkedin.com/in/**
 miconda http://www.linkedin.com/in/miconda
 Kamailio Advanced Trainings - Berlin, Oct 21

Re: [SR-Users] Calling users on another kamailio server

[Charles Chance]

Hi,

Just managed to check code. Correct me if I'm wrong, but I think it won't
work with add_path() because the function uses substitution SUBST_SND_ALL
which of course refers to the source ip:port;transport=proto of the
outgoing message.

Regards,

Charles



On 16 September 2013 16:06, Charles Chance charles.cha...@sipcentric.comwrote:

 Peter - calling msg_apply_changes() after add_path() should indeed work,
 I'm sure, so thanks for pointing it out. However, when testing I'm getting
 parse errors in debug output, so am unable to confirm at this time.

 Brian - which version are you using? Anything in log? After calling
 save(), path column in DB is empty?



 On 16 September 2013 15:18, Brian Wallen bwal...@gmail.com wrote:

 In my test setup the registrar is at the edge and add_path() does in fact
 do nothing. I also tried Charles' suggestion which didn't seem to add the
 Path header either.


 On Mon, Sep 16, 2013 at 10:04 AM, Peter Dunkley 
 peter.dunk...@crocodilertc.net wrote:

 I thought append_hf() didn't take affect (unless you use
 msg_apply_changes()) until the message left Kamailio too?

 If that is the case, and msg_apply_changes() is called, doesn't that
 mean the Path: header from add_path() would be added in that scenario?


 On 16 September 2013 14:49, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 It is possible if the edge proxy and registrar are separate, as you
 say. But if the registrar is at the edge with no separate proxy, add_path()
 does nothing (because the message never leaves Kamailio for the header to
 be added).


 On 16 September 2013 14:42, Peter Dunkley 
 peter.dunk...@crocodilertc.net wrote:

 Is there any reason it isn't possible?


 On 16 September 2013 14:21, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hi,

 Yes, you are right - and I agree, it would be better if this was
 possible :)

 Charles




 On 16 September 2013 14:15, Peter Dunkley 
 peter.dunk...@crocodilertc.net wrote:

 Hello,

 It'd be better if the add_path() function could be used here.
  That way, if using outbound (RFC5626), the flow-token (the userinfo 
 part
 of the Path-URI) would be present and there would be no need to add the
 ;received parameter.

 This would address the one issue remaining for SIP outbound on
 Kamailio, which is its use without an edge proxy that is separate from 
 the
 registrar.

 Regards,

 Peter


 On 16 September 2013 14:05, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hi,

 This sounds like a case for sharing same database, and adding Path
 before saving incoming register. That way, no need to replicate 
 register
 message to other servers and all subscribers use the same domain.

 Add path something like this before calling save():

 append_hf(Path: sip:$Ri:$Rp;
 received=sip:$si:$sp;lr\r\n);
 msg_apply_changes();

 Whichever server receives the incoming invite, will perform lookup
 and automatically route to the server which received the register. On 
 the
 proxying server set $du according to received param of route header, 
 add
 record-route, and then t_relay(). As Daniel said, no need to
 re-authenticate or perform lookup again.

 Regards,

 Charles



 On Mon, Sep 16, 2013 at 7:34 AM, Daniel-Constantin Mierla 
 mico...@gmail.com wrote:

 Hello,


 On 9/12/13 10:08 PM, Brian Wallen wrote:

 I currently have two independent kamailio servers. I'd like to
 set them up in a way that user1 on server1 can make a call to user2 
 on
 server2. After searching I've come up with two ways that this might 
 be able
 to be done. Can someone please sanity check these or point me in the 
 right
 direction?

 1. Have one registrar server and convert the other server to a
 proxy
 2. Keep them both as registrars and somehow make them each aware
 of the users on the other server

 I like 2 better because if one server went down users on the
 other server would still be up. The only thing is I don't know how 
 to set
 the servers up to communicate with each other.

 the nat can create problems when a server is down - if the nat is
 symmetric, only the server that received the registrar can send back 
 calls
 to the phone.

 Communication between users on two servers is as simple as using
 t_relay_to(proto:serverip:**port) after you do
 lookup(location) and no record is found.


 Suppose I have three or more kamailio servers. If a call comes in
 and lookup() returns that no record was found, how do I know which 
 server
 to forward to? Is that a case in which I should replicate the database?


 You have to add an extra check for the case the call was coming
 from the other server, not to forward back to it in case of no found 
 again.


 Thanks for the tip, I hadn't thought of that.


 Also, you should skip user authentication for calls from the other
 server (not do authenticate twice). Another aspect to take care is 
 chaining
 rtpproxy, you have to use the flat for trusting the other server (r, 
 iirc

Re: [SR-Users] Kamailio realtime fails to forward register with Asterisk.

[Charles Chance]

Hi,

Could be something network related, or firewall? I know you stated phones
could register directly to Asterisk but if nothing is showing in Asterisk
SIP log, then the requests are never getting to it.

Was your trace performed on Kamailio server or Asterisk?

Cheers,

Charles



On 13 September 2013 17:12, imperium broadcast imperium.broadc...@gmail.com
 wrote:

 Hi all I have been trying to figure out where Im going wrong with
 following this guide:
 http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb

 The only thing that is different is I am using Asterisk 1.8.22.0

 Kamailio works fine and I am able to register a phone to it but the
 forward register fails to register with Asterisk.


 Its like Asterisk doesn't see the request even running sip debug on the
 Asterisk Console it doesn't show the register attempts. I have to use ngrep
 to see any thing.

 Realtime works fine when I register the phone on Asterisk.

 Although I can get round the register using AGI I get the same issue when
 passing a call to asterisk, Asterisk doesn't see the invite.

 Am I missing something so obvious :-/


 This is what I have in sip.conf
 IP change to protect the innocent!

 [kamailio-2]
 type=peer
 host=1.1.1.1
 fromdomain=1.1.1.1
 context=outgoing
 insecure=invite,port
 directmedia=nonat
 qualify=yes
 disallow=all
 allow=all

 and this is the sip trace  for the register


 U 2013/09/13 16:54:50.697484 1.1.1.1:5060 - 172.16.0.110:5080
 REGISTER sip:172.16.0.110:5080 SIP/2.0.
 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0.
 To: sip:102@172.16.0.110.
 From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5.
 CSeq: 10 REGISTER.
 Call-ID: 4670378d1185f5b6-25000@127.0.0.1.
 Content-Length: 0.
 User-Agent: kamailio (3.3.1 (x86_64/linux)).
 Contact: sip:102@172.16.0.112:5060.
 Expires: 120.
 .


 U 2013/09/13 16:54:54.697484 1.1.1.1:5060 - 172.16.0.110:5080
 REGISTER sip:172.16.0.110:5080 SIP/2.0.
 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0.
 To: sip:102@172.16.0.110.
 From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5.
 CSeq: 10 REGISTER.
 Call-ID: 4670378d1185f5b6-25000@127.0.0.1.
 Content-Length: 0.
 User-Agent: kamailio (3.3.1 (x86_64/linux)).
 Contact: sip:102@172.16.0.112:5060.
 Expires: 120.

 and I have this for my
 Asterisk binds in kamamilio.cfg

 #!ifdef WITH_ASTERISK
 asterisk.bindip = 172.16.0.110 desc Asterisk IP Address
 asterisk.bindport = 5080 desc Asterisk Port
 kamailio.bindip = 172.16.0.112 desc Kamailio IP Address
 kamailio.bindport = 5060 desc Kamailio Port
 #!endif


 --
 Regards
 Mick


 ___
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 sr-users@lists.sip-router.org
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Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered 
office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham 
B7 4EJ.
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Re: [SR-Users] issues with dispatcher rerouting to next server when getting a 500 from freeswitch

[Charles Chance]

Hi,

You describe what you are trying to achieve, but not the result you are
seeing.

One thing I noticed is that in your failure route you set
t_on_failure(DISPATCH), but should it not be t_on_failure(RTF_DISPATCH)?

Regards,

Charles
 On 14 Sep 2013 08:33, Joseph Watson joseph.wat...@veranetworks.com
wrote:

 Hello Everyone,

 I am really having an issue getting kamailio to try another dispatch route
 when I get a 500 back from one of my freeswitch servers.

 Currently I have a list of freeswitch servers located in my dispatch.list
 file. If one of then is to busy it will respond with a 500 Maximum Call in
 Progress. I am trying to get Kamailio to go to the next route in the
 dispatch.list when it get one of these 503 and if it should exhaust the
 whole list respond with a 503. Here is my kamailio config


 ### Global Parameters #

 #!ifdef WITH_DEBUG
 debug=4
 log_stderror=yes
 #!else
 debug=4
 log_stderror=yes
 #!endif

 memdbg=5
 memlog=5

 log_facility=LOG_LOCAL0

 fork=yes
 children=10

 /* comment the next line to enable TCP */
 disable_tcp=yes

 /* uncomment the next line to disable the auto discovery of local aliases
based on revers DNS on IPs (default on) */
 auto_aliases=no

 /* uncomment and configure the following line if you want Kamailio to
bind on a specific interface/port/proto (default bind on all available)
 */
 listen=udp:eth0:5060

 sip_warning=no

 ### Modules Section 

 #set module path

 mpath=/usr/local/lib64/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/

 loadmodule mi_fifo.so
 loadmodule kex.so
 loadmodule tm.so
 loadmodule tmx.so
 loadmodule sl.so
 loadmodule rr.so
 loadmodule pv.so
 loadmodule maxfwd.so
 loadmodule textops.so
 loadmodule siputils.so
 loadmodule xlog.so
 loadmodule sanity.so
 loadmodule ctl.so
 loadmodule mi_rpc.so
 loadmodule acc.so
 loadmodule dispatcher.so


 # - setting module-specific parameters ---

 # - mi_fifo params -
 modparam(mi_fifo, fifo_name, /tmp/kamailio_fifo)

 modparam(xlog, buf_size, 8192)

 # - rr params -
 # add value to ;lr param to cope with most of the UAs
 modparam(rr, enable_full_lr, 1)
 # do not append from tag to the RR (no need for this script)
 modparam(rr, append_fromtag, 0)


 # - acc params -
 modparam(acc, log_flag, 1)
 modparam(acc, failed_transaction_flag, 3)
 modparam(acc, log_extra,
 src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si)

 # - tm params -
 modparam(tm, fr_timer, 2000)
 modparam(tm, fr_inv_timer, 4)
 modparam(tm, remap_503_500, 0)

 # - dispatcher params -
 modparam(dispatcher, list_file,
 /usr/local/etc/kamailio/dispatcher.list)
 modparam(dispatcher, flags, 2)
 modparam(dispatcher, dst_avp, $avp(i:271))
 modparam(dispatcher, grp_avp, $avp(i:272))
 modparam(dispatcher, cnt_avp, $avp(i:273))

 #modparam(dispatcher, dst_avp, $avp(AVP_DST))
 #modparam(dispatcher, grp_avp, $avp(AVP_GRP))
 #modparam(dispatcher, cnt_avp, $avp(AVP_CNT))
 #modparam(dispatcher, ds_append_branch, 1)
 ### Routing Logic 


 # main request routing logic

 route {
 append_hf(X-VSIP-CUST-IP: $si\r\n);
 route(DISPATCH);

 route(RELAY);
 }

 # Dispatch requests
 route[DISPATCH] {
 # round robin dispatching on gateways group '1'
 if(!ds_select_dst(1, 4))
 {
 send_reply(404, No destination);
 exit;
 }
 #   xlog(L_DBG, --- SCRIPT: going to $ru via $du\n);
 t_on_failure(RTF_DISPATCH);
 return;
 }

 # Sample failure route
 failure_route[RTF_DISPATCH] {
 if (t_is_canceled()) {
 exit;
 }
 # next DST - only for 500 or local timeout
 if (t_check_status(500) or (t_branch_timeout() and
 !t_branch_replied()))
 {
 if(ds_next_dst())
 {
 t_on_failure(DISPATCH);
 route(RELAY);
 exit;
 }
 }
 }

 route[RELAY] {
 if (!t_relay()) {
 send_reply(503, LB01 - All Circuits Are Busy);
 #   sl_reply_error();
 }
 exit;
 }

 I am very new to kamailio and still learning my way around. Any help you
 can provide with be fantastic!

 Thank you,
 Joe Watson


 ___
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 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



-- 
www.sipcentric.com

Follow us on twitter @sipcentric http://twitter.com/sipcentric

Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered 
office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham 
B7 4EJ.
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Re: [SR-Users] Kamailio realtime fails to forward register with Asterisk.

[Charles Chance]

Hi Mick,

What is the output of the following commands on Asterisk machine?

netstat -lnp | grep asterisk
iptables -L
sestatus

Regards,

Charles



On 14 September 2013 12:43, imperium broadcast imperium.broadc...@gmail.com
 wrote:

 Hi Charles,  the sip trace was carried out on the asterisk server. I have
 also disable the fw on both servers. I suspect  it might be network issue
 but I have no idea what.
 I have also set up another server running asterisk 11 same problem so I
 know its not asterisk 1.8. I get no errors on either sever and logs don't
 show any issues. Its just an odd issue thats got me baffled.

 Regards
 Mick
 On 14 Sep 2013 09:08, Charles Chance charles.cha...@sipcentric.com
 wrote:

 Hi,

 Could be something network related, or firewall? I know you stated phones
 could register directly to Asterisk but if nothing is showing in Asterisk
 SIP log, then the requests are never getting to it.

 Was your trace performed on Kamailio server or Asterisk?

 Cheers,

 Charles



 On 13 September 2013 17:12, imperium broadcast 
 imperium.broadc...@gmail.com wrote:

 Hi all I have been trying to figure out where Im going wrong with
 following this guide:

 http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb

 The only thing that is different is I am using Asterisk 1.8.22.0

 Kamailio works fine and I am able to register a phone to it but the
 forward register fails to register with Asterisk.


 Its like Asterisk doesn't see the request even running sip debug on the
 Asterisk Console it doesn't show the register attempts. I have to use ngrep
 to see any thing.

 Realtime works fine when I register the phone on Asterisk.

 Although I can get round the register using AGI I get the same issue
 when passing a call to asterisk, Asterisk doesn't see the invite.

 Am I missing something so obvious :-/


 This is what I have in sip.conf
 IP change to protect the innocent!

 [kamailio-2]
 type=peer
 host=1.1.1.1
 fromdomain=1.1.1.1
 context=outgoing
 insecure=invite,port
 directmedia=nonat
 qualify=yes
 disallow=all
 allow=all

 and this is the sip trace  for the register


 U 2013/09/13 16:54:50.697484 1.1.1.1:5060 - 172.16.0.110:5080
 REGISTER sip:172.16.0.110:5080 SIP/2.0.
 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0.
 To: sip:102@172.16.0.110.
 From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5.
 CSeq: 10 REGISTER.
 Call-ID: 4670378d1185f5b6-25000@127.0.0.1.
 Content-Length: 0.
 User-Agent: kamailio (3.3.1 (x86_64/linux)).
 Contact: sip:102@172.16.0.112:5060.
 Expires: 120.
 .


 U 2013/09/13 16:54:54.697484 1.1.1.1:5060 - 172.16.0.110:5080
 REGISTER sip:172.16.0.110:5080 SIP/2.0.
 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0.
 To: sip:102@172.16.0.110.
 From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5.
 CSeq: 10 REGISTER.
 Call-ID: 4670378d1185f5b6-25000@127.0.0.1.
 Content-Length: 0.
 User-Agent: kamailio (3.3.1 (x86_64/linux)).
 Contact: sip:102@172.16.0.112:5060.
 Expires: 120.

 and I have this for my
 Asterisk binds in kamamilio.cfg

 #!ifdef WITH_ASTERISK
 asterisk.bindip = 172.16.0.110 desc Asterisk IP Address
 asterisk.bindport = 5080 desc Asterisk Port
 kamailio.bindip = 172.16.0.112 desc Kamailio IP Address
 kamailio.bindport = 5060 desc Kamailio Port
 #!endif


 --
 Regards
 Mick


 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered
 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
 B7 4EJ.

 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


 ___
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 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




-- 
*Charles Chance*
Managing Director

t. 0121 285 4400m. 07932 063 891

-- 
www.sipcentric.com

Follow us on twitter @sipcentric http://twitter.com/sipcentric

Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered 
office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham 
B7 4EJ.
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Re: [SR-Users] Kamailio realtime fails to forward register with Asterisk.

[Charles Chance]

Another thing, you do have bindport=5080 in your sip.conf, don't you?

Regards,

Charles



On 14 September 2013 13:42, Charles Chance charles.cha...@sipcentric.comwrote:

 Hi Mick,

 What is the output of the following commands on Asterisk machine?

 netstat -lnp | grep asterisk
 iptables -L
 sestatus

 Regards,

 Charles



 On 14 September 2013 12:43, imperium broadcast 
 imperium.broadc...@gmail.com wrote:

 Hi Charles,  the sip trace was carried out on the asterisk server. I have
 also disable the fw on both servers. I suspect  it might be network issue
 but I have no idea what.
 I have also set up another server running asterisk 11 same problem so I
 know its not asterisk 1.8. I get no errors on either sever and logs don't
 show any issues. Its just an odd issue thats got me baffled.

 Regards
 Mick
 On 14 Sep 2013 09:08, Charles Chance charles.cha...@sipcentric.com
 wrote:

 Hi,

 Could be something network related, or firewall? I know you stated
 phones could register directly to Asterisk but if nothing is showing in
 Asterisk SIP log, then the requests are never getting to it.

 Was your trace performed on Kamailio server or Asterisk?

 Cheers,

 Charles



 On 13 September 2013 17:12, imperium broadcast 
 imperium.broadc...@gmail.com wrote:

 Hi all I have been trying to figure out where Im going wrong with
 following this guide:

 http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb

 The only thing that is different is I am using Asterisk 1.8.22.0

 Kamailio works fine and I am able to register a phone to it but the
 forward register fails to register with Asterisk.


 Its like Asterisk doesn't see the request even running sip debug on the
 Asterisk Console it doesn't show the register attempts. I have to use ngrep
 to see any thing.

 Realtime works fine when I register the phone on Asterisk.

 Although I can get round the register using AGI I get the same issue
 when passing a call to asterisk, Asterisk doesn't see the invite.

 Am I missing something so obvious :-/


 This is what I have in sip.conf
 IP change to protect the innocent!

 [kamailio-2]
 type=peer
 host=1.1.1.1
 fromdomain=1.1.1.1
 context=outgoing
 insecure=invite,port
 directmedia=nonat
 qualify=yes
 disallow=all
 allow=all

 and this is the sip trace  for the register


 U 2013/09/13 16:54:50.697484 1.1.1.1:5060 - 172.16.0.110:5080
 REGISTER sip:172.16.0.110:5080 SIP/2.0.
 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0.
 To: sip:102@172.16.0.110.
 From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5.
 CSeq: 10 REGISTER.
 Call-ID: 4670378d1185f5b6-25000@127.0.0.1.
 Content-Length: 0.
 User-Agent: kamailio (3.3.1 (x86_64/linux)).
 Contact: sip:102@172.16.0.112:5060.
 Expires: 120.
 .


 U 2013/09/13 16:54:54.697484 1.1.1.1:5060 - 172.16.0.110:5080
 REGISTER sip:172.16.0.110:5080 SIP/2.0.
 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0.
 To: sip:102@172.16.0.110.
 From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5.
 CSeq: 10 REGISTER.
 Call-ID: 4670378d1185f5b6-25000@127.0.0.1.
 Content-Length: 0.
 User-Agent: kamailio (3.3.1 (x86_64/linux)).
 Contact: sip:102@172.16.0.112:5060.
 Expires: 120.

 and I have this for my
 Asterisk binds in kamamilio.cfg

 #!ifdef WITH_ASTERISK
 asterisk.bindip = 172.16.0.110 desc Asterisk IP Address
 asterisk.bindport = 5080 desc Asterisk Port
 kamailio.bindip = 172.16.0.112 desc Kamailio IP Address
 kamailio.bindport = 5060 desc Kamailio Port
 #!endif


 --
 Regards
 Mick


 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. 
 Registered
 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
 B7 4EJ.

 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


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Re: [SR-Users] Nat_traversal module not sending keepalives

[Charles Chance]

Hi,

Many thanks for your replies. Turns out we were calling t_newtran()
elsewhere in the config before handling the register, and I guess the
callback was being attached to the wrong transaction. Now it's removed, all
is working fine :)

Best,

Charles



On 9 September 2013 19:09, Carsten Bock cars...@ng-voice.com wrote:

 Hi,

 you find a working example actually in the examples/pcscf/ Folder of
 Kamailio.
 Since we don't have a full compatible usrloc-API for IMS (yet), i used
 nat_traversal as replacement for nathelper; as it's not relying on the
 usrloc-database.

 In the config we have:
 # - nat_traversal params -
 # If another keepalive is wanted, this is the place
 modparam(nat_traversal, keepalive_interval, 10)
 # If another method than NOTIFY is wanted:
 modparam(nat_traversal, keepalive_method, OPTIONS)
 # From?
 modparam(nat_traversal, keepalive_from, sip:keepalive@HOSTNAME)
 # Where we store information about keep-alives:
 modparam(nat_traversal, keepalive_state_file,
 /var/run/kamailio/keepalive_state)

 With these parameters, we just call nat_keepalive(); before
 forwarding the request to the IMS-Core.  This works for us... (with
 latest 4.0 or trunk versions, i haven't tried it before).

 Kind regards,
 Carsten

 2013/9/9 Daniel-Constantin Mierla mico...@gmail.com:
  Hello,
 
  not using this module, however, after you start, can you see a dedicated
  timer module 'NT' if you do kamctl ps?
 
  Cheers,
  Daniel
 
 
  On Fri, Sep 6, 2013 at 11:00 AM, Charles Chance
  charles.cha...@sipcentric.com wrote:
 
  Hi all,
 
  Currently evaluating the nat_traversal module in some tests, and trying
 to
  configure keepalives for registers. We currently use nathelper, but due
 to
  the distributed nature of our setup, nat_traversal seems more
 appropriate
  for the keepalive functionality (nathelper is sending keepalives from
 every
  node).
 
  I am calling nat_keepalive() before save(), but no keepalives are being
  sent. The documentation suggests that is all there is to it, but maybe
 I'm
  missing something. There are no errors in the log that I can see.
 
  Any ideas?
 
  Many thanks,
 
  Charles
 
 
  www.sipcentric.com
 
  Follow us on twitter @sipcentric
 
  Sipcentric Ltd. Company registered in England  Wales no. 7365592.
  Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court
 South,
  Birmingham B7 4EJ.
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 ng-voice GmbH
 Schomburgstr. 80
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 mailto:cars...@ng-voice.com

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 Fax +49 40 34927220

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 Registergericht: Amtsgericht Hamburg, HRB 120189
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Re: [SR-Users] Kamailio's XMPP issue

[Charles Chance]

Hi,

Not familiar with the specific module but it looks like a more general DNS
issue - do you have the correct DNS server(s) configured in resolv.conf?
Are you able to resolve other hostnames?

Regards,

Charles
On 7 Sep 2013 11:56, Nandini madhu sermj2...@gmail.com wrote:

 Dear All,

 I am working on XMPP module of kamailio server, but i stucked with some
 Integration error.
 Firstly my Kamailio and Jabberd servers are on the same machine. I
 managed to separately make them work. Call setup between SIP clients
 is working and so with messaging/presence, File transfer functions for
 Jabberd
 clients.
 But the problem is when trying to integrate this jabberd2 server with my
 kamailio server by setting XMPP module in kamailio's configuration file,
 SIP server is starting as usually but syslog gives the following error :

 Aug 28 19:45:14 kamailio /usr/local/sbin/kamailio[
 19648]: ERROR: xmpp [network.c:110]: net_connect(): resolving
 xmpp.example.com failed (Unknown host).

 what could be the problem ? please help me in resolving this issue ?

 PS: My task is to achieve 'File transfer' feature between SIP and XMPP
 clients (also between two SIP clients). so with this regards can you give
 me any hints or links to accomplish this.

 Any help will greatly appreciate.

 Regards,
 Nandini

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[SR-Users] Nat_traversal module not sending keepalives

[Charles Chance]

Hi all,

Currently evaluating the nat_traversal module in some tests, and trying to
configure keepalives for registers. We currently use nathelper, but due to
the distributed nature of our setup, nat_traversal seems more appropriate
for the keepalive functionality (nathelper is sending keepalives from every
node).

I am calling nat_keepalive() before save(), but no keepalives are being
sent. The documentation suggests that is all there is to it, but maybe I'm
missing something. There are no errors in the log that I can see.

Any ideas?

Many thanks,

Charles

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Re: [SR-Users] Presence: Get number of active_watchers

[Charles Chance]

Hi Jan,

Why not use sqlops to query the database from within your config script?

Best regards,

Charles



On 21 August 2013 13:36, Jan Gaida jan.ga...@grupoamper.com wrote:

 Hello,

 I need to obtain the number of subscribed watchers for a given
 presentity-uri.

 I can get it with a SQL Query like select count(*) from
 kamailio.active_watchers where presentity_uri = 'test@domain';. But I
 think there should be a way to get it from within the configuration script.

 Regards

 --
 *Jan **Gaida*
 Ingeniero Desarrollo Software C/ Marconi 3 (PTM)
 28760 Tres Cantos
 Spain
 jan.ga...@grupoamper.com | www.grupoamper.com


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[SR-Users] DB_CLUSTER - Cluster Definition

[Charles Chance]

Hi All,

I'm having difficulties with the above module and can't see what I'm doing
wrong.

Config as follows:

modparam(db_cluster, connection, db01=mysql://
 readwrite:passw...@db01.domain.com/kamailio)
 modparam(db_cluster, connection, db02=mysql://
 readonly:passw...@db02.domain.com/kamailio)
 modparam(db_cluster, cluster, cls1=db01=0s9s;db02=9s0s)



As you can see, there are two database servers - a master and a slave. I am
trying to force all read operations to the slave (db02) and all write
operations to the master (db01).

Unless I am missing the point, the above priorities should achieve the
desired effect. However, it seems to be going wrong somewhere as write
operations are being sent to the slave:

ERROR: db_mysql [km_dbase.c:122]: driver error on query: DELETE command
 denied to user 'readonly'@'x.x.x.x' for table 'location'

ERROR: db_mysql [km_dbase.c:122]: driver error on query: UPDATE command
 denied to user 'readonly'@'x.x.x.x' for table 'location'
 ERROR: db_mysql [km_dbase.c:122]: driver error on query: INSERT command
 denied to user 'readonly'@'x.x.x.x' for table 'location'


It is the same for all modules, by the way, not just usrloc. For each
module, I have db_url set to cluster://cls1.

If I change the cluster definition to cls1=db01=9s9s;db02=0s0s, all
operations (read and write) are sent to db01 as expected.

Am I missing something here? Can anyone see what I may be doing wrong?

Many thanks,

Charles

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Re: [SR-Users] releasing v3.3.5

[Charles Chance]

Hi Daniel,

Sorry if I missed it already, but did the fix to clear early state dialogs
get backported in the end?

Cheers,

Charles
 On 2 Aug 2013 09:17, Daniel-Constantin Mierla mico...@gmail.com wrote:

 Hello,

 I am considering to package soon a new version out of branch 3.3. If
 anyone has spotted fixes that should be backported to that branch, reply
 here. I'm not sure yet of an exact date, but should be rather soon, being
 just a patch update.

 A new release out of branch 4.0 will be done short after this one.

 Cheers,
 Daniel

 --
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 http://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/**micondahttp://www.linkedin.com/in/miconda


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Re: [SR-Users] releasing v3.3.5

[Charles Chance]

Hi Daniel,

 ...it might be better just to upgrade to 4.0 instead of making 3.3
unstable.

Good point, thanks :)

Cheers,

Charles
On 2 Aug 2013 09:35, Daniel-Constantin Mierla mico...@gmail.com wrote:

  Hello,

 iirc, not yet in 4.0, but it will. Doing it for 3.3 has to be analyzed,
 not sure if would be just simple backport and then it might be better just
 to upgrade to 4.0 instead of making 3.3 unstable.

 Cheers,
 Daniel

 On 8/2/13 10:27 AM, Charles Chance wrote:

 Hi Daniel,

 Sorry if I missed it already, but did the fix to clear early state dialogs
 get backported in the end?

 Cheers,

 Charles
  On 2 Aug 2013 09:17, Daniel-Constantin Mierla mico...@gmail.com
 wrote:

 Hello,

 I am considering to package soon a new version out of branch 3.3. If
 anyone has spotted fixes that should be backported to that branch, reply
 here. I'm not sure yet of an exact date, but should be rather soon, being
 just a patch update.

 A new release out of branch 4.0 will be done short after this one.

 Cheers,
 Daniel

 --
 Daniel-Constantin Mierla - http://www.asipto.com
 http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda


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 - http://www.linkedin.com/in/miconda



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Re: [SR-Users] Asterisk cluster with Kamailio for load balancing.

[Charles Chance]

I'm not aware of anything, but others may be.

However, before diving straight into the Asterisk/Kamailio integration, may
I suggest you familiarise yourself first with Kamailio on its own. In my
opinion, this will give you a far better understanding of how it all fits
together, and the learning curve will be much more manageable.

Any specific questions you have as you go along can be asked here.

Cheers,

Charles



On 11 July 2013 16:50, Володимир Іванець volodyaivan...@gmail.com wrote:

 Hello,

 Yes I have. But I was hoping find some tutorial with experience sharing.
 Because I'm new to Kamailio, official tutorial did not cower all my
 questions.

 Thanks.

 Regards, Volodya Ivanets.


 2013/7/11 Charles Chance charles.cha...@sipcentric.com

 Hello,

 Have you looked at
 http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
 ?

 Charles


 On 11 July 2013 16:01, Володимир Іванець volodyaivan...@gmail.comwrote:

 Hello everyone!

 Recently I became interested in concept of putting couple of Asterisks
 servers in cluster. So I tried to replicate this scenario in testing
 environment. I was using database back-end for Asterisks settings and
 statistics and Kamailio as a load balancer.

 Obviously it did not worked [image: :cry:] . There is possible to make
 so many mistakes. Now I'm trying to find some step by step tutorial, but
 nothing yet.

 If someone experienced this, please advice. And if someone have useful
 links, please share.

 Thanks a lot.

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Re: [SR-Users] How to cleanup unconfirmed dialog in memory?

[Charles Chance]

Hi Daniel,

If someone can try it and confirm is working fine for sl replied dialogs as
 well as for those forwarded, I will backport.


Seems to work fine for us in our test environment. So far, no unconfirmed
dialogs stuck in memory for statelessly replied transactions.

Cheers,

Charles



On 3 July 2013 17:25, Charles Chance charles.cha...@sipcentric.com wrote:

 Hi Daniel,

 Sounds perfect! I will try to test here tomorrow and let you know.

 Cheers,

 Charles


 On 2 July 2013 22:44, Daniel-Constantin Mierla mico...@gmail.com wrote:

  Hello,

 I pushed a patch in master that detects when the dialog is created by not
 getting to transaction due to a stateless reply.


 http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=fa0339b1906690f009786fc9ed92c73a8c9e6520;hp=b4682cac2e2f151288a411018da077b6d1526eca

 If someone can try it and confirm is working fine for sl replied dialogs
 as well as for those forwarded, I will backport.

 Cheers,
 Daniel


 On 7/2/13 4:22 PM, Carlos Ruiz Díaz wrote:

  Now that you mention it, it makes perfect sense since this function
 sends BYE to both legs which only makes sense with confirmed dialogs.

  I run out of alternatives. Maybe patching the module is the only
 solution.

  Regards,
 Carlos


 On Tue, Jul 2, 2013 at 10:16 AM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 That doesn't work I'm afraid. Also from
 http://www.kamailio.org/docs/modules/3.1.x/modules_k/dialog.html#id2524783:


  *Note: Works only for confirmed dialogs.*


  It's something I've been meaning to look further into for a while, but
 haven't yet had chance. I suspect a small patch will be required though.

  Regards,

  Charles



 On 2 July 2013 15:11, Carlos Ruiz Díaz carlos.ruizd...@gmail.comwrote:

 Ok, I haven't noticed that, although I can't tell for sure whether it
 will work or not.

  If you can dump the dialogs using xmlrpc or rpc interface, maybe you
 could parse the info and tear down those unconfirmed dialogs using
 dlg_end_dlg:


 http://www.kamailio.org/docs/modules/3.1.x/modules_k/dialog.html#id2524783


 On Tue, Jul 2, 2013 at 10:02 AM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 The parameter does not work for us. We have modparam(dialog,
 default_timeout,7200), but it only has an effect on confirmed dialogs. 
 As
 you can see from my previous example, there is not even a timestart 
 value
 on the unconfirmed dialog, so how can Kamailio know when timeout is
 reached?



 On 2 July 2013 14:51, Carlos Ruiz Díaz carlos.ruizd...@gmail.comwrote:

 There is, but for all dialogs, not only the unconfirmed ones.


 http://www.kamailio.org/docs/modules/3.3.x/modules_k/dialog.html#default-timeout-id


 On Tue, Jul 2, 2013 at 9:49 AM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hmm, I don't think there is even a timeout value set on unconfirmed
 dialogs in memory.

  Example (Kamailio 3.3.3):

  dialog::  hash=1791:10106
  state:: 1
  ref_count:: 1
  timestart:: 0
  timeout:: 0
  ...

  Whereas:

  dialog::  hash=2963:2808
  state:: 4
  ref_count:: 2
  timestart:: 1372772302
  timeout:: 114829207
  ...

  Therefore, the unconfirmed dialogs never get cleared
 automatically, in my experience at least. I hope I'm wrong though :)

  Cheers,

  Charles



 On 2 July 2013 14:31, Henning Westerholt h...@kamailio.org wrote:

 Am Dienstag, 2. Juli 2013, 14:23:25 schrieb Charles Chance:
  I don't think this will help at all, as regardless of DB mode,
 unconfirmed
  dialogs are not stored in DB anyway.
 
  The important thing to remember is that if you are calling
 dialog_manage()
  in your config, to only do it once you are ready to forward the
 request. If
  you call it but then exit for some reason without actually
 forwarding, you
  will probably end up with a stuck dialog.
 
  Maybe someone else can suggest other possible causes?
 
  To my knowledge, there is no existing way to clear these without
 restarting.

  Hello,

 AFAIK these stale dialogs are cleaned up after the dialog timeout.
 There are
 module parameter and also dialog specific parameter to control this
 variable.
 This stale dialogs needs a bit of memory, but are otherwise
 harmless.

 Best regards,

 Henning





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 Registered
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  *Charles Chance

Re: [SR-Users] How to cleanup unconfirmed dialog in memory?

[Charles Chance]

Hi Gary,

Are you looking at the correct dialog? Before the BYE message, an ongoing
dialog will have already moved from state 1 through 2, 3, 4 and finally
state 5 after the BYE, before being cleared automatically after a few
seconds.

Perhaps there are two dialogs being created for the same initial request?
Maybe due to retransmissions or an error in config? If you can supply your
config we may be able to get a better idea.

Regards,

Charles



On 4 July 2013 04:09, Gary Chen gchen3...@gmail.com wrote:

 We need also have a way to timeout all the dead unconfirmed dialog. I have
 a situation that the SIP message was perfect fine with Bye signal, yet the
 dialog is still hang with state 1.

 Gary


 On Wed, Jul 3, 2013 at 12:25 PM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hi Daniel,

 Sounds perfect! I will try to test here tomorrow and let you know.

 Cheers,

 Charles


 On 2 July 2013 22:44, Daniel-Constantin Mierla mico...@gmail.com wrote:

  Hello,

 I pushed a patch in master that detects when the dialog is created by
 not getting to transaction due to a stateless reply.


 http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=fa0339b1906690f009786fc9ed92c73a8c9e6520;hp=b4682cac2e2f151288a411018da077b6d1526eca

 If someone can try it and confirm is working fine for sl replied dialogs
 as well as for those forwarded, I will backport.

 Cheers,
 Daniel


 On 7/2/13 4:22 PM, Carlos Ruiz Díaz wrote:

  Now that you mention it, it makes perfect sense since this function
 sends BYE to both legs which only makes sense with confirmed dialogs.

  I run out of alternatives. Maybe patching the module is the only
 solution.

  Regards,
 Carlos


 On Tue, Jul 2, 2013 at 10:16 AM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 That doesn't work I'm afraid. Also from
 http://www.kamailio.org/docs/modules/3.1.x/modules_k/dialog.html#id2524783:


  *Note: Works only for confirmed dialogs.*


  It's something I've been meaning to look further into for a while,
 but haven't yet had chance. I suspect a small patch will be required 
 though.

  Regards,

  Charles



 On 2 July 2013 15:11, Carlos Ruiz Díaz carlos.ruizd...@gmail.comwrote:

 Ok, I haven't noticed that, although I can't tell for sure whether it
 will work or not.

  If you can dump the dialogs using xmlrpc or rpc interface, maybe you
 could parse the info and tear down those unconfirmed dialogs using
 dlg_end_dlg:


 http://www.kamailio.org/docs/modules/3.1.x/modules_k/dialog.html#id2524783


 On Tue, Jul 2, 2013 at 10:02 AM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 The parameter does not work for us. We have modparam(dialog,
 default_timeout,7200), but it only has an effect on confirmed dialogs. 
 As
 you can see from my previous example, there is not even a timestart 
 value
 on the unconfirmed dialog, so how can Kamailio know when timeout is
 reached?



 On 2 July 2013 14:51, Carlos Ruiz Díaz carlos.ruizd...@gmail.comwrote:

 There is, but for all dialogs, not only the unconfirmed ones.


 http://www.kamailio.org/docs/modules/3.3.x/modules_k/dialog.html#default-timeout-id


 On Tue, Jul 2, 2013 at 9:49 AM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hmm, I don't think there is even a timeout value set on unconfirmed
 dialogs in memory.

  Example (Kamailio 3.3.3):

  dialog::  hash=1791:10106
  state:: 1
  ref_count:: 1
  timestart:: 0
  timeout:: 0
  ...

  Whereas:

  dialog::  hash=2963:2808
  state:: 4
  ref_count:: 2
  timestart:: 1372772302
  timeout:: 114829207
  ...

  Therefore, the unconfirmed dialogs never get cleared
 automatically, in my experience at least. I hope I'm wrong though :)

  Cheers,

  Charles



 On 2 July 2013 14:31, Henning Westerholt h...@kamailio.org wrote:

 Am Dienstag, 2. Juli 2013, 14:23:25 schrieb Charles Chance:
  I don't think this will help at all, as regardless of DB mode,
 unconfirmed
  dialogs are not stored in DB anyway.
 
  The important thing to remember is that if you are calling
 dialog_manage()
  in your config, to only do it once you are ready to forward the
 request. If
  you call it but then exit for some reason without actually
 forwarding, you
  will probably end up with a stuck dialog.
 
  Maybe someone else can suggest other possible causes?
 
  To my knowledge, there is no existing way to clear these without
 restarting.

  Hello,

 AFAIK these stale dialogs are cleaned up after the dialog timeout.
 There are
 module parameter and also dialog specific parameter to control
 this variable.
 This stale dialogs needs a bit of memory, but are otherwise
 harmless.

 Best regards,

 Henning





 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. 
 Registered
 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, 
 Birmingham
 B7 4EJ.

 ___
 SIP Express Router (SER) and Kamailio

Re: [SR-Users] How to cleanup unconfirmed dialog in memory?

[Charles Chance]

Hi Daniel,

Sounds perfect! I will try to test here tomorrow and let you know.

Cheers,

Charles


On 2 July 2013 22:44, Daniel-Constantin Mierla mico...@gmail.com wrote:

  Hello,

 I pushed a patch in master that detects when the dialog is created by not
 getting to transaction due to a stateless reply.


 http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=fa0339b1906690f009786fc9ed92c73a8c9e6520;hp=b4682cac2e2f151288a411018da077b6d1526eca

 If someone can try it and confirm is working fine for sl replied dialogs
 as well as for those forwarded, I will backport.

 Cheers,
 Daniel


 On 7/2/13 4:22 PM, Carlos Ruiz Díaz wrote:

  Now that you mention it, it makes perfect sense since this function
 sends BYE to both legs which only makes sense with confirmed dialogs.

  I run out of alternatives. Maybe patching the module is the only
 solution.

  Regards,
 Carlos


 On Tue, Jul 2, 2013 at 10:16 AM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 That doesn't work I'm afraid. Also from
 http://www.kamailio.org/docs/modules/3.1.x/modules_k/dialog.html#id2524783:


  *Note: Works only for confirmed dialogs.*


  It's something I've been meaning to look further into for a while, but
 haven't yet had chance. I suspect a small patch will be required though.

  Regards,

  Charles



 On 2 July 2013 15:11, Carlos Ruiz Díaz carlos.ruizd...@gmail.com wrote:

 Ok, I haven't noticed that, although I can't tell for sure whether it
 will work or not.

  If you can dump the dialogs using xmlrpc or rpc interface, maybe you
 could parse the info and tear down those unconfirmed dialogs using
 dlg_end_dlg:


 http://www.kamailio.org/docs/modules/3.1.x/modules_k/dialog.html#id2524783


 On Tue, Jul 2, 2013 at 10:02 AM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 The parameter does not work for us. We have modparam(dialog,
 default_timeout,7200), but it only has an effect on confirmed dialogs. As
 you can see from my previous example, there is not even a timestart value
 on the unconfirmed dialog, so how can Kamailio know when timeout is
 reached?



 On 2 July 2013 14:51, Carlos Ruiz Díaz carlos.ruizd...@gmail.comwrote:

 There is, but for all dialogs, not only the unconfirmed ones.


 http://www.kamailio.org/docs/modules/3.3.x/modules_k/dialog.html#default-timeout-id


 On Tue, Jul 2, 2013 at 9:49 AM, Charles Chance 
 charles.cha...@sipcentric.com wrote:

 Hmm, I don't think there is even a timeout value set on unconfirmed
 dialogs in memory.

  Example (Kamailio 3.3.3):

  dialog::  hash=1791:10106
  state:: 1
  ref_count:: 1
  timestart:: 0
  timeout:: 0
  ...

  Whereas:

  dialog::  hash=2963:2808
  state:: 4
  ref_count:: 2
  timestart:: 1372772302
  timeout:: 114829207
  ...

  Therefore, the unconfirmed dialogs never get cleared automatically,
 in my experience at least. I hope I'm wrong though :)

  Cheers,

  Charles



 On 2 July 2013 14:31, Henning Westerholt h...@kamailio.org wrote:

 Am Dienstag, 2. Juli 2013, 14:23:25 schrieb Charles Chance:
  I don't think this will help at all, as regardless of DB mode,
 unconfirmed
  dialogs are not stored in DB anyway.
 
  The important thing to remember is that if you are calling
 dialog_manage()
  in your config, to only do it once you are ready to forward the
 request. If
  you call it but then exit for some reason without actually
 forwarding, you
  will probably end up with a stuck dialog.
 
  Maybe someone else can suggest other possible causes?
 
  To my knowledge, there is no existing way to clear these without
 restarting.

  Hello,

 AFAIK these stale dialogs are cleaned up after the dialog timeout.
 There are
 module parameter and also dialog specific parameter to control this
 variable.
 This stale dialogs needs a bit of memory, but are otherwise harmless.

 Best regards,

 Henning





 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. 
 Registered
 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, 
 Birmingham
 B7 4EJ.

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 list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




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 http://caruizdiaz.com
 +595981146623

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  --
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 Managing Director

  t. 0121 285 4400m. 07932 063 891

 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. 
 Registered
 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
 B7 4EJ.

 ___
 SIP Express Router

Re: [SR-Users] memcache problem - reporting protocol error

[Charles Chance]

Hi Dragos,

Yes, that should work fine.

Cheers,

Charles
On 7 Jun 2013 12:00, Dragos Oancea droan...@yahoo.com wrote:

 Hi Charles,

 Thank you for the information. These details are good to know.
 Do you think it's safe to use the new module from git with kamailio-4.0.0 ?
 I mean just copying it in the kamailio-4.0.0 source tree, compiling it and
 then putting it in the modules directory to be loaded at runtime.


 Regards,
 Dragos

   --
  *From:* Charles Chance charles.cha...@sipcentric.com
 *To:* Dragos Oancea droan...@yahoo.com
 *Cc:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
 Users Mailing List sr-users@lists.sip-router.org
 *Sent:* Friday, June 7, 2013 11:06 AM
 *Subject:* Re: [SR-Users] memcache problem - reporting protocol error

 The main difference, other than the new library, is the ability to specify
 an expiry for each stored key/value at the time of setting it:

 So now there are two ways to set a value:

 $mct(key) = value; (default expiry will be used as per module param)

 or the new way, $mct(key=expiry) = value;


 Currently, the module still only supports a single memcache server,
 although if you are looking for a distributed solution, we have tested
 Couchbase/Moxi and it works really well.

 All the best,

 Charles


 On 6 June 2013 20:40, Dragos Oancea droan...@yahoo.com wrote:

 Hi Charles,

 Thank you for your answer.
 I've notice that your name shows in the headers of the files of the new
 (patched) memcached module.
 Can you please detail a little bit on what does your patch bring new apart
 from libmemcached support ?
 Or what does it still miss, since you said it's under heavy development.

 Many thanks!

 Regards,
 Dragos

   --
  *From:* Charles Chance charles.cha...@sipcentric.com
 *To:* Dragos Oancea droan...@yahoo.com
 *Cc:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
 Users Mailing List sr-users@lists.sip-router.org
 *Sent:* Thursday, June 6, 2013 7:37 PM

 *Subject:* Re: [SR-Users] memcache problem - reporting protocol error

 Although it's difficult to know the exact cause of your protocol error,
 it does appear from your tests that the graceful handling of errors and
 subsequent reconnects is better in libmemcached than in libmemcache. So if
 you do encounter the same conditions again, I would imagine that the
 library would handle it and attempt to reconnect transparently.
 If not, however, then at least you are more likely to find support as the
 library is still under active development :)
 Cheers,
 Charles
  On 6 Jun 2013 18:12, Dragos Oancea droan...@yahoo.com wrote:

 Hi Charles,

 I tried first to trigger again the bug using some iptables rules (to
 simulate a network problem) , so Kamailio could not communicate with the
 memcache anymore:

  WARNING: memcached [memcached.c:189]: WARNING: memcached:
 mcm_server_readable():2582: timeout: select(2) call timed out for
 read(2)able fds
 WARNING: memcached [memcached.c:189]: WARNING: memcached:
 mcm_server_writable():3173: timeout: write select(2) call timed out
 WARNING: memcached [memcached.c:189]: WARNING: memcached:
 mcm_server_connect():2290: select(2) failed: select(2) timed out on
 establishing conn
 ection
 NOTICE: memcached [memcached.c:194]: NOTICE: memcached:
 mcm_server_connect():2297: connect(2) failed
 NOTICE: memcached [memcached.c:194]: NOTICE: memcached:
 mcm_server_connect_next_avail():2323: unable to find a server to connect to

 This looks very much like what i've seen in the logs just before it
 started to report the protocol error problem.

 But in the simulated case , after I remove the iptables rules that block
 the access to memcache, Kamailo does not recover from the situation, it
 just keeps on printing messages like the above, when it should
 connect to memcache instead. It looks like another libmemcache-related bug
 to me.

 Then I tried the same thing with  the latest memcached module from git:
 With the memcached firewalled after starting Kamailio it prints this to
 the logs:

 ERROR: memcached [mcd_var.c:154]: pv_get_mcd_value_helper(): could not get
 result for
 key 29515_HA1 - error was 'SERVER HAS FAILED AND IS DISABLED UNTIL TIMED
 RETRY'

 When I flush the firewall rules, it reconnects to memcache, which makes it
 better than the old one.
 But how can I be sure that the new one will not have the same protocol
 error issue, since I could not reproduce exactly the problem so far ?


 Regards,
 Dragos



   --
  *From:* Charles Chance charles.cha...@sipcentric.com
 *To:* Dragos Oancea droan...@yahoo.com; Kamailio (SER) - Users Mailing
 List sr-users@lists.sip-router.org
 *Sent:* Thursday, June 6, 2013 6:27 PM
 *Subject:* Re: [SR-Users] memcache problem - reporting protocol error

 Hi Dragos,

 The memcached module has indeed been updated in the master to use
 libmemcached, as the old libmemcache library is no longer under active
 development. Have you tried

Re: [SR-Users] memcache problem - reporting protocol error

[Charles Chance]

Hi Dragos,

The memcached module has indeed been updated in the master to use
libmemcached, as the old libmemcache library is no longer under active
development. Have you tried using the latest version of the module from
git? Do you still get the same errors?

Cheers,

Charles


On 6 June 2013 16:18, Dragos Oancea droan...@yahoo.com wrote:

 Hello


 We are having issues with the memcached module (Kamailio 4.0.0) .


 All of a sudden we are getting this kind of messages in the logs, and the
 values for the requested keys are not retrieved.


 ALERT: memcached [memcached.c:189]: ALERT: memcached:
 mcm_fetch_cmd():1305: memcache(4) protocol error: STORED#015#012VALUE
 29515_HA1 0
 32#015#012b52299247f16c6adfa4c8cc53b34e74a#015#012END#015#012STORED#015#012
 [...]
 ALERT: memcached [memcached.c:189]: ALERT: memcached:
 mcm_fetch_cmd():1176: memcache(4) protocol error: server sent data for key
 not in request


 I suspect that at some point there was a network problem between the
 memcached server and Kamailio that triggered this abnormal behavior and I
 also suspect that it is
 connected with the (old) libmemcache library (libmemcache-1.4.0.rc2) .


 This is fixed only with a Kamailio restart.


 Has anyone experience this ?  Is there a fix ?
 I noticed that the Kamailio from the trunk compiles the memcached module
 with the more new libmemcached library from here:
 https://launchpad.net/libmemcached/+download .
 Maybe this is one of the reasons libmemcache is going to be dropped ?

 Thank you very much.

 Regards,
 Dragos




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[SR-Users] Fwd:

[Charles Chance]

http://www.toyosoken.co.jp/ivtduw.php


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Re: [SR-Users] Fwd:

[Charles Chance]

Hi,

Sincere apologies, not sure how this happened - please ignore, it was not
me who sent it :/

Regards,

Charles


On 2 April 2013 08:21, Charles Chance charles.cha...@sipcentric.com wrote:

 http://www.toyosoken.co.jp/ivtduw.php


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t. 0121 285 4400m. 07932 063 891

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[SR-Users] Dialog module - shared state

[Charles Chance]

Hi all,

I was wondering, how are others sharing dialog module (particularly
profile) state between multiple servers?

Thanks in advance,

Charles

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office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham 
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Re: [SR-Users] Dialog module - shared state

[Charles Chance]

Hi Daniel,

Yes, dmq is the clear way forward once it's working. Is it a lot of work to
get going again? For it to be really useful, I would imagine other modules
will need to make use of it from within their code, rather than the user
trying to do stuff in the config. I may take a look and see how far along
it is.

The others all seem to hinge around dialog:start and dialog:end event
routes, but we really need to replicate info about early dialogs too. So I
think we'll have to send some notifications from the other routes as well.

Anyway, thanks again,

Charles



On 26 March 2013 15:05, Daniel-Constantin Mierla mico...@gmail.com wrote:

 Hello,


 On 3/26/13 1:51 PM, Charles Chance wrote:

 Hi all,

 I was wondering, how are others sharing dialog module (particularly
 profile) state between multiple servers?


 if it is just about counting the active dialog calls per user/ip/etc ...
 so far I am mainly using htable, with notifications from each node to the
 other using uac_req_send() (see next for ways for when to send
 notifications). I index by key (userid,ip/etc) plus the call id, then I use
 $shtcn(...) to get the number of items for an user/ip/etc...

 An alternative is to use a database table and replicate its content. This
 comes in addition to the dialog table, using the event_routes exported by
 the dialog module to update records:

 - http://kamailio.org/docs/**modules/stable/modules/dialog.**
 html#id2556376http://kamailio.org/docs/modules/stable/modules/dialog.html#id2556376

 dmq module is supposed to offer the framework for inter-kamailio
 communication, iirc, the developer added support for distributing htable
 keys. But then he changed job and didn't work further on this part, it is
 in my list to get it going again before next major release.

 Cheers,
 Daniel

 --
 Daniel-Constantin Mierla - http://www.asipto.com
 http://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/**micondahttp://www.linkedin.com/in/miconda
 Kamailio World Conference, April 16-17, 2013, Berlin
  - http://conference.kamailio.com -


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Re: [SR-Users] Dialog module - shared state

[Charles Chance]

Thanks Daniel, that's what I'm thinking (dmq).

I'll let you know how I get on!

Cheers,

Charles
 On Mar 26, 2013 4:40 PM, Daniel-Constantin Mierla mico...@gmail.com
wrote:

 Hello,

 On 3/26/13 5:30 PM, Charles Chance wrote:

 Hi Daniel,

 Yes, dmq is the clear way forward once it's working. Is it a lot of work
 to get going again? For it to be really useful, I would imagine other
 modules will need to make use of it from within their code, rather than the
 user trying to do stuff in the config. I may take a look and see how far
 along it is.

 The others all seem to hinge around dialog:start and dialog:end event
 routes, but we really need to replicate info about early dialogs too. So I
 think we'll have to send some notifications from the other routes as well.


 Initial dialog event_route was not added because that is practically the
 moment when dlg_manage() is executed for initial INVITE -- if there is no
 error executing dlg_manage(), do the actions for initialing the state on
 the other nodes as well.

 Could be easier to add event_routes for add to/remove from profiles -- the
 information is really small there, replicating it should be easy. Or even
 skip event routes and replicate directly from inside dialog module using
 dmq.

 Cheers,
 Daniel

 --
 Daniel-Constantin Mierla - http://www.asipto.com
 http://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/**micondahttp://www.linkedin.com/in/miconda
 Kamailio World Conference, April 16-17, 2013, Berlin
  - http://conference.kamailio.com -



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Re: [SR-Users] Message Waiting

[Charles Chance]

From past experience you may need to add another entry to the dispatcher
list with a different setid because no doubt sipsak is sending the notify
from a different port to that which Asterisk is listening on.

Eg. setid - 10, destination - sip:asterisk_ip:5065

Then - if(ds_is_from_list(10))...

Cheers,

Charles
Thanks to all for the input. I wrote a script to call sipsak to generate
the notify and all seems to be working, however am having issues getting
kamailio to forward to the endpoint.

The notify is sent unsolicited so I figured I could just add a statement
such as:

if(is_method(NOTIFY)  ds_is_from_list(1)){
route(LOCATION);
.etc

(asterisk server sending notify is in the ds list.)

What would be the proper way for kamailio to handle this?

-Original Message-
From: sr-users-boun...@lists.sip-router.org [mailto:
sr-users-boun...@lists.sip-router.org] On Behalf Of Juha Heinanen
Sent: Thursday, March 14, 2013 3:44 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Message Waiting

Jon Farmer writes:

  I have also seen a couple of script methods that look at the
  voicemail directory structure for file changes and trigger an
  application like sipsak to generate the notifications and let
  kamailio relay them. From what I see, the consensus is not to go
  this route. In my case the script would need to look at the database
  level, and maybe some sort of database trigger could be used.
...
 
 I have been using sipsak to generate the NOTIFY for the last 7 years.
 It never has caused an issue. I have a setup similar to you but I
 wrote a customer voicemail application in AGI instead of using the
 built in Asterisk app.

my sems voicemail application is using xmlrpc to make pua_publish call to
presence server when status of user's message box changes.

-- juha

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[SR-Users] Malformed PUBLISH from pua_dialoginfo (4.0.0)

[Charles Chance]

Hi all,

We are testing 4.0.0 in the lab and since upgrading, PUBLISHes generated by
pua_dialoginfo appear to be malformed (content-length mismatch).

Please see attached output from Wireshark for both the PUBLISH and the 400
response from Kamailio. PUBLISH is being sent from one Kamailio to another
- nothing has changed other than upgrading from 3.3.3 to 4.0.0.

Has anyone else noticed the same?

Regards,

Charles

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B7 4EJ.
No. TimeSourceDestination   Protocol Info
   6123 2660.332884 10.28.0.2010.28.0.21SIP  
Request: PUBLISH sip:001...@sip.sipcentric.com

Frame 6123 (791 bytes on wire, 791 bytes captured)
Arrival Time: Mar 12, 2013 12:46:48.231722000
[Time delta from previous captured frame: 0.000536000 seconds]
[Time delta from previous displayed frame: 0.000536000 seconds]
[Time since reference or first frame: 2660.332884000 seconds]
Frame Number: 6123
Frame Length: 791 bytes
Capture Length: 791 bytes
[Frame is marked: True]
[Protocols in frame: eth:ip:udp:sip]
[Coloring Rule Name: UDP]
[Coloring Rule String: udp]
Ethernet II, Src: RealtekU_7e:bb:71 (52:54:00:7e:bb:71), Dst: RealtekU_3c:a2:4f 
(52:54:00:3c:a2:4f)
Destination: RealtekU_3c:a2:4f (52:54:00:3c:a2:4f)
Address: RealtekU_3c:a2:4f (52:54:00:3c:a2:4f)
 ...0     = IG bit: Individual address (unicast)
 ..1.     = LG bit: Locally administered address 
(this is NOT the factory default)
Source: RealtekU_7e:bb:71 (52:54:00:7e:bb:71)
Address: RealtekU_7e:bb:71 (52:54:00:7e:bb:71)
 ...0     = IG bit: Individual address (unicast)
 ..1.     = LG bit: Locally administered address 
(this is NOT the factory default)
Type: IP (0x0800)
Internet Protocol, Src: 10.28.0.20 (10.28.0.20), Dst: 10.28.0.21 (10.28.0.21)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0x10 (DSCP 0x04: Unknown DSCP; ECN: 0x00)
0001 00.. = Differentiated Services Codepoint: Unknown (0x04)
 ..0. = ECN-Capable Transport (ECT): 0
 ...0 = ECN-CE: 0
Total Length: 777
Identification: 0x7641 (30273)
Flags: 0x00
0.. = Reserved bit: Not Set
.0. = Don't fragment: Not Set
..0 = More fragments: Not Set
Fragment offset: 0
Time to live: 64
Protocol: UDP (0x11)
Header checksum: 0xed32 [correct]
[Good: True]
[Bad : False]
Source: 10.28.0.20 (10.28.0.20)
Destination: 10.28.0.21 (10.28.0.21)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Source port: sip (5060)
Destination port: sip (5060)
Length: 757
Checksum: 0x6656 [validation disabled]
[Good Checksum: False]
[Bad Checksum: False]
Session Initiation Protocol
Request-Line: PUBLISH sip:001...@sip.sipcentric.com SIP/2.0
Method: PUBLISH
Request-URI: sip:001...@sip.sipcentric.com
Request-URI User Part: 001462
Request-URI Host Part: sip.sipcentric.com
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 10.28.0.20;branch=z9hG4bK8023.45ffe121.0
Transport: UDP
Sent-by Address: 10.28.0.20
Branch: z9hG4bK8023.45ffe121.0
To: sip:001...@sip.sipcentric.com
SIP to address: sip:001...@sip.sipcentric.com
SIP to address User Part: 001462
SIP to address Host Part: sip.sipcentric.com
From: 
sip:001...@sip.sipcentric.com;tag=2283942c43d5e413d656674cb322bb6f-3954
SIP from address: sip:001...@sip.sipcentric.com
SIP from address User Part: 001462
SIP from address Host Part: sip.sipcentric.com
SIP tag: 2283942c43d5e413d656674cb322bb6f-3954
CSeq: 10 PUBLISH
Sequence Number: 10
Method: PUBLISH
Call-ID: 1acf33f3068b19b8-27919@10.28.0.20
Content-Length: 290
User-Agent: kamailio (4.0.0 (x86_64/linux))
Max-Forwards: 70
Event: dialog
Expires: 7201
Content-Type: application/dialog-info+xml
Message Body
?xml version=1.0?\n
dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=0 
state=full entity=sip:001...@sip.sipcentric.com\n
  dialog id=30a5ea2d34d254b56e7efd6a15512...@sip.sipcentric.com 
direction=recipient\n
stateterminated/state\n
  /dialog\n
/dialog-info\n

No. TimeSourceDestination   Protocol Info
   6125 2660.333430 10.28.0.2110.28.0.20SIP  
Status: 400 Content-Length mis-match

Re: [SR-Users] Malformed PUBLISH from pua_dialoginfo (4.0.0)

[Charles Chance]

Hi Daniel,

It is from release (rpm).

Thanks,

Charles



On 12 March 2013 13:18, Daniel-Constantin Mierla mico...@gmail.com wrote:

  Hello,

 is 4.0 from release of a bit older taken from the branch 4.0?

 Cheers,
 Daniel


 On 3/12/13 2:04 PM, Charles Chance wrote:

 Hi all,

  We are testing 4.0.0 in the lab and since upgrading, PUBLISHes generated
 by pua_dialoginfo appear to be malformed (content-length mismatch).

  Please see attached output from Wireshark for both the PUBLISH and the
 400 response from Kamailio. PUBLISH is being sent from one Kamailio to
 another - nothing has changed other than upgrading from 3.3.3 to 4.0.0.

  Has anyone else noticed the same?

  Regards,

  Charles


 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered
 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
 B7 4EJ.

 ___
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 --
 Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda 
 - http://www.linkedin.com/in/miconda
 Kamailio World Conference, April 16-17, 2013, Berlin
  - http://conference.kamailio.com -


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Re: [SR-Users] Malformed PUBLISH from pua_dialoginfo (4.0.0)

[Charles Chance]

Hi Daniel,

They were downloaded yesterday afternoon.

We have downloaded again just now and the problem seems to have gone away.
Sorry, we should have spotted the newer packages today and tried them first
:/

Many thanks,

Charles



On 12 March 2013 13:30, Daniel-Constantin Mierla mico...@gmail.com wrote:

  When did you download it? I had some builds done days before the release
 just to test packaging is ok. Those ones after release were built last
 evening and in this morning.

 Cheers,
 Daniel


 On 3/12/13 2:25 PM, Charles Chance wrote:

 Hi Daniel,

  It is from release (rpm).

  Thanks,

  Charles



  On 12 March 2013 13:18, Daniel-Constantin Mierla mico...@gmail.comwrote:

  Hello,

 is 4.0 from release of a bit older taken from the branch 4.0?

 Cheers,
 Daniel


 On 3/12/13 2:04 PM, Charles Chance wrote:

  Hi all,

  We are testing 4.0.0 in the lab and since upgrading, PUBLISHes
 generated by pua_dialoginfo appear to be malformed (content-length
 mismatch).

  Please see attached output from Wireshark for both the PUBLISH and the
 400 response from Kamailio. PUBLISH is being sent from one Kamailio to
 another - nothing has changed other than upgrading from 3.3.3 to 4.0.0.

  Has anyone else noticed the same?

  Regards,

  Charles


  www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered
 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
 B7 4EJ.

 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
 listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


 --
 Daniel-Constantin Mierla - 
 http://www.asipto.comhttp://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/miconda
 Kamailio World Conference, April 16-17, 2013, Berlin
  - http://conference.kamailio.com -


 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered
 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
 B7 4EJ.


 --
 Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda 
 - http://www.linkedin.com/in/miconda
 Kamailio World Conference, April 16-17, 2013, Berlin
  - http://conference.kamailio.com -



-- 
www.sipcentric.com

Follow us on twitter @sipcentric http://twitter.com/sipcentric

Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered 
office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham 
B7 4EJ.
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Re: [SR-Users] Message Waiting

[Charles Chance]

The script method works fine for us - but we use the externnotify option in
voicemail.conf instead of scanning for changes at directory/database level.

Regards,

Charles


On 12 March 2013 19:32, Daniel W. Graham d...@cmsinter.net wrote:

 While I have seen other posts I have not found anything that describes my
 scenario.

 ** **

 Kamailio sets in front of a group of asterisk servers which are used in a
 round robin dispatch group. All asterisk servers share a common database in
 which voicemail boxes are defined and messages are stored.

 ** **

 I’m looking for a way to make message waiting work in this example. I have
 tried creating a view with basic peer information for asterisk to use for
 message notification, however asterisk does not ‘see’ these peers until a
 call is made to the specific user and the peer info is retrieved from the
 database, which can delay notifications severely rendering them useless. I
 also want to avoid sending a copy of the registrations to asterisk servers,
 as the purpose of the distributed system is to eliminate the large number
 of sip registrations asterisk needs to manage.

 ** **

 I have also seen a couple of script methods that look at the voicemail
 directory structure for file changes and trigger an application like sipsak
 to generate the notifications and let kamailio relay them. From what I see,
 the consensus is not to go this route. In my case the script would need to
 look at the database level, and maybe some sort of database trigger could
 be used.

 ** **

 What are others doing / what do others recommend as the best way to handle
 message waiting with a situation where kamailio sets in front of a
 voicemail server? 

 ** **

 Any input is appreciated.

 Dan-

 ** **

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 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



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Re: [SR-Users] Message Waiting

[Charles Chance]

The script you specify (e.g. externnotify=myscript.sh) runs every time a
new message is left for a user or when a user exits the VoiceMailMain()
application (for instance, if they have dialled in to clean up their
mailbox).

The voicemail application passes certain things as parameters to your
script such as number of new/old messages etc - for more info and example
script take a look here:
http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf

Cheers,

Charles



On 12 March 2013 19:45, Daniel W. Graham d...@cmsinter.net wrote:

 Can you describe how your implementation works without having to scan
 directory/database for changes?

 ** **

 -Dan

 ** **

 *From:* sr-users-boun...@lists.sip-router.org [mailto:
 sr-users-boun...@lists.sip-router.org] *On Behalf Of *Charles Chance
 *Sent:* Tuesday, March 12, 2013 3:42 PM
 *To:* Kamailio (SER) - Users Mailing List
 *Subject:* Re: [SR-Users] Message Waiting

 ** **

 The script method works fine for us - but we use the externnotify option
 in voicemail.conf instead of scanning for changes at directory/database
 level.

 ** **

 Regards,

 ** **

 Charles

 ** **

 On 12 March 2013 19:32, Daniel W. Graham d...@cmsinter.net wrote:

 While I have seen other posts I have not found anything that describes my
 scenario.

  

 Kamailio sets in front of a group of asterisk servers which are used in a
 round robin dispatch group. All asterisk servers share a common database in
 which voicemail boxes are defined and messages are stored.

  

 I’m looking for a way to make message waiting work in this example. I have
 tried creating a view with basic peer information for asterisk to use for
 message notification, however asterisk does not ‘see’ these peers until a
 call is made to the specific user and the peer info is retrieved from the
 database, which can delay notifications severely rendering them useless. I
 also want to avoid sending a copy of the registrations to asterisk servers,
 as the purpose of the distributed system is to eliminate the large number
 of sip registrations asterisk needs to manage.

  

 I have also seen a couple of script methods that look at the voicemail
 directory structure for file changes and trigger an application like sipsak
 to generate the notifications and let kamailio relay them. From what I see,
 the consensus is not to go this route. In my case the script would need to
 look at the database level, and maybe some sort of database trigger could
 be used.

  

 What are others doing / what do others recommend as the best way to handle
 message waiting with a situation where kamailio sets in front of a
 voicemail server? 

  

 Any input is appreciated.

 Dan-

  


 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

 ** **


 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered
 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
 B7 4EJ.

 ___
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 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



-- 
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Follow us on twitter @sipcentric http://twitter.com/sipcentric

Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered 
office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham 
B7 4EJ.
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Re: [SR-Users] Message Waiting

[Charles Chance]

Sipsak is generally what we use, although any similar tool should do it.
Just watch the port numbers!

Cheers,

Charles



On 12 March 2013 19:58, Daniel W. Graham d...@cmsinter.net wrote:

 I read that right after I posted my next question, sorry for jumping the
 gun without doing some research. Do you mind me asking what method you use
 for generating the notifications, asterisk call file / sipsak / etc ?

 ** **

 *From:* sr-users-boun...@lists.sip-router.org [mailto:
 sr-users-boun...@lists.sip-router.org] *On Behalf Of *Charles Chance
 *Sent:* Tuesday, March 12, 2013 3:54 PM
 *To:* Kamailio (SER) - Users Mailing List
 *Subject:* Re: [SR-Users] Message Waiting

 ** **

 The script you specify (e.g. externnotify=myscript.sh) runs every time a
 new message is left for a user or when a user exits the VoiceMailMain()
 application (for instance, if they have dialled in to clean up their
 mailbox).

 ** **

 The voicemail application passes certain things as parameters to your
 script such as number of new/old messages etc - for more info and example
 script take a look here:
 http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf

 ** **

 Cheers,

 ** **

 Charles

 ** **

 ** **

 On 12 March 2013 19:45, Daniel W. Graham d...@cmsinter.net wrote:

 Can you describe how your implementation works without having to scan
 directory/database for changes?

  

 -Dan

  

 *From:* sr-users-boun...@lists.sip-router.org [mailto:
 sr-users-boun...@lists.sip-router.org] *On Behalf Of *Charles Chance
 *Sent:* Tuesday, March 12, 2013 3:42 PM
 *To:* Kamailio (SER) - Users Mailing List
 *Subject:* Re: [SR-Users] Message Waiting

  

 The script method works fine for us - but we use the externnotify option
 in voicemail.conf instead of scanning for changes at directory/database
 level.

  

 Regards,

  

 Charles

  

 On 12 March 2013 19:32, Daniel W. Graham d...@cmsinter.net wrote:

 While I have seen other posts I have not found anything that describes my
 scenario.

  

 Kamailio sets in front of a group of asterisk servers which are used in a
 round robin dispatch group. All asterisk servers share a common database in
 which voicemail boxes are defined and messages are stored.

  

 I’m looking for a way to make message waiting work in this example. I have
 tried creating a view with basic peer information for asterisk to use for
 message notification, however asterisk does not ‘see’ these peers until a
 call is made to the specific user and the peer info is retrieved from the
 database, which can delay notifications severely rendering them useless. I
 also want to avoid sending a copy of the registrations to asterisk servers,
 as the purpose of the distributed system is to eliminate the large number
 of sip registrations asterisk needs to manage.

  

 I have also seen a couple of script methods that look at the voicemail
 directory structure for file changes and trigger an application like sipsak
 to generate the notifications and let kamailio relay them. From what I see,
 the consensus is not to go this route. In my case the script would need to
 look at the database level, and maybe some sort of database trigger could
 be used.

  

 What are others doing / what do others recommend as the best way to handle
 message waiting with a situation where kamailio sets in front of a
 voicemail server? 

  

 Any input is appreciated.

 Dan-

  


 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

  


 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered
 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
 B7 4EJ.


 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



 

 ** **


 www.sipcentric.com

 Follow us on twitter @sipcentric http://twitter.com/sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered
 office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
 B7 4EJ.

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office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham 
B7 4EJ

Re: [SR-Users] Update existing module or create new?

[Charles Chance]

Hi Henning,


On 22 February 2013 13:19, Henning Westerholt h...@kamailio.org wrote:

 Am Donnerstag, 21. Februar 2013, 23:37:07 schrieb Charles Chance:
  [..]
  We now have an updated memcached module, working with libmemcached and
 also
  with the added ability to (optionally) specify expiry in the format
  $mct(key=expiry).
 
  How do we get these changes pushed back into the master?

 Hello Charles,

 sounds great! If this is a smaller patch, it could be simply applied to the
 git master branch from a developer, after the branching of the 4.0 version
 next week. Ovidiu sended you some informations about that already.

 But I assume its a more intrusive change which maybe also bring some new
 bugs
 etc...? Do you plan to do more work on this module?

 Best regards,

 Henning Westerholt



The structure is very much the same as before, but I think it is more than
a small patch. There is certainly potential for new bugs! I do plan to add
multi-server support at some point and I am happy to maintain/bug fix where
necessary. Also happy to update the documentation. Other than that, there's
not much more to add at this time.

Regards,

Charles

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Follow us on twitter @sipcentric http://twitter.com/sipcentric

Sipcentric Ltd. Company registered in England  Wales no. 7365592. Registered 
office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham 
B7 4EJ.
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Re: [SR-Users] Path and DNS SRV

[Charles Chance]

Hi Klaus,

That worked perfectly, thanks!

One thing I've noticed - nathelper is sending OPTIONS pings to all
contacts, regardless of the incoming socket. So both servers are sending
pings to all clients. I am assuming this is because in our tests the
clients and servers are on the same subnet - do you think this is the case?
Obviously, in a production environment where clients are behind NAT, it
only makes sense to send keep-alives from the server a client is registered
on.

Thanks again,

Charles


On 22 February 2013 07:43, Klaus Darilion klaus.mailingli...@pernau.atwrote:

 If the add the Path header manually, then it is not seen by the
 registrar/usrloc module, asthe header will only be added when generating
 the to be forwarded message - and this never happens as the message
 terminates on the registrar. You could try msg_apply_changes() after adding
 the Path header.

 regards
 Klaus

 http://kamailio.org/docs/**modules/3.3.x/modules/**
 textopsx.html#textopsx.msg_**apply_changeshttp://kamailio.org/docs/modules/3.3.x/modules/textopsx.html#textopsx.msg_apply_changes




 On 22.02.2013 00:33, Charles Chance wrote:

 Hi,

 Thanks for your reply.

 I did try that, but the path is still not stored in location table on
 save. I have use_path parameter of registrar module enabled.


 On 21 February 2013 23:17, Carsten Bock cars...@ng-voice.com
 mailto:cars...@ng-voice.com wrote:

 Hi,

 you cannot use add_path(), but by creating your own Path header this
 works, e.g.:

 append_hf(Path: sip:term@HOSTNAME:+PORT+;**lr\r\n);
 (from our P-CSCF config)

 Then your Proxy should also evaluate the DNS-SRV records.

 Kind regards,
 Carsten
 --
 Carsten Bock
 CEO (Geschäftsführer)

 ==**==**= Die
 ng-voice GmbH ist auf der Kamailio World 2013, 16./17. April 2013
 =
 http://conference.kamailio.**com/k01/==**
 ==**=http://conference.kamailio.com/k01/=

 ng-voice GmbH
 Schomburgstr. 80
 D-22767 Hamburg / Germany

 http://www.ng-voice.com
 mailto:cars...@ng-voice.com mailto:cars...@ng-voice.com


 Office +49 40 34927219
 Fax +49 40 34927220

 Sitz der Gesellschaft: Hamburg
 Registergericht: Amtsgericht Hamburg, HRB 120189
 Geschäftsführer: Carsten Bock
 Ust-ID: DE279344284

 Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
 http://www.ng-voice.com/**imprint/ http://www.ng-voice.com/imprint/

 Hi All,

 Given a situation where there are multiple registrars, behind a
 single load balancer, sharing the same location table in DB, I
 understand that Path can be used to ensure everything works nicely
 with clients behind NAT. So, if UA1 is registered on P1, UA2 on P2,
 and UA1 calls UA2... P1 performs lookup, sees path and sends via P2.

 What about situations where the distribution between registrars is
 being handled client-side through DNS SRV, not a load balancer? Path
 cannot be used then, right? As I understand it, in this instance
 add_path() will do nothing, is this correct?

 Thanks in advance,

 Charles


 www.sipcentric.com http://www.sipcentric.com/

 Follow us on twitter @sipcentric http://twitter.com/sipcentric**


 Sipcentric Ltd. Company registered in England  Wales no. 7365592.
 Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court
 South, Birmingham B7 4EJ.
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