Re: [SR-Users] replace From header username for AUTH

[Daniel Tryba]

On Tuesday 21 July 2015 14:30:51 Al S wrote:
> I was wondering how can I replace From header username to perform the
> following authentication check with the new value:
> 
> route[AUTH]..if (!auth_check("$fd", "subscriber", "1"))
> {auth_challenge("$fd", "1");exit;}...

http://www.kamailio.org/wiki/cookbooks/4.2.x/pseudovariables#fu_-_from_uri_username

If changing the value doesn't work you need to use
http://kamailio.org/docs/modules/stable/modules/textopsx.html#textopsx.f.msg_apply_changes

But even that will not fix your problem, since it is not the From that is used 
for identification/authentication but the Authorization header set by the 
endpoint. And since the hash is calculated with this username by the endpoint, 
changing the username will break the process.


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[SR-Users] About Acc_radius rad_sttaus Function

[Yasin CANER]

  
  

    Hello;
    There is a problem in acc_radius module that radius message has
2 time sends "Acct-Status-Type = Start" when call is established as
example below. it should be ALIVE as i see in rad_status.if
req_METHOD is ACK , it should be ALIVE. And There is a problem
Sip-Method Problem that even if it is ACK , it sends Invite. How can
i fix these problems could you help me about it?
i already defined Acct-Status-Ttpe Values like Below
VALUE Acct-Status-Type   Failed   15   # RFC2866, acc
VALUE Acct-Status-Type   Start    1   # RFC2866, acc
VALUE Acct-Status-Type   Stop 2   # RFC2866, acc
VALUE Acct-Status-Type   Alive   3   # RFC2866, acc

When i debug tag.len in Invite and ack transaction i get 0 value but
in Bye it has bigger than 0 .So that it never be Alive

static inline uint32_t rad_status( struct sip_msg *req, int code )
{
    str tag;
    unsigned int in_dialog_req = 0;

    tag = get_to(req)->tag_value;
    if(tag.s!=0 && tag.len!=0)
    in_dialog_req = 1;

    if (req->REQ_METHOD==METHOD_INVITE &&
in_dialog_req == 0
    && code>=200 && code<300)
    return rd_vals[RV_STATUS_START].v;
    if ((req->REQ_METHOD==METHOD_BYE ||
req->REQ_METHOD==METHOD_CANCEL))
    return rd_vals[RV_STATUS_STOP].v;
    if (in_dialog_req != 0)
    return rd_vals[RV_STATUS_ALIVE].v;
    return rd_vals[RV_STATUS_FAILED].v;
 }

-- RADIUS PACKET
Mon Jun 29 12:47:33 2015
    Acct-Status-Type = Start
    Service-Type = Sip-Session
    Sip-Response-Code = 200
    Sip-Method = Invite
Mon Jun 29 12:47:33 2015
    Acct-Status-Type = Start
    Service-Type = Sip-Session
    Sip-Response-Code = 200
    Sip-Method = Invite
Mon Jun 29 12:48:07 2015
    Acct-Status-Type = Stop
    Service-Type = Sip-Session
    Sip-Response-Code = 200
    Sip-Method = Bye

  


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Re: [SR-Users] About Acc_radius rad_sttaus Function

[ycaner]

Hello ;
When i debug req->REQ_METHOD for METHOD_ACK , it never give that state. it
is always Invite and Invite
Thanks



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Re: [SR-Users] Kazoo like load balancing

[Jurijs Ivolga]

Hi,

Please find my config in attachment. I'm using default config, with minor
changes.

Please let me know if you need any additional info for debugging this issue.

With kind regards,

Jurijs

2015-07-21 19:18 GMT+03:00 Daniel Tryba :

> On Tuesday 21 July 2015 17:59:53 Jurijs Ivolga wrote:
> > So basically extension 100 sends invite to LB, LB then send invite to
> > Kamailio server and Kamailio server replies with 407 error.
> >
> > Do you have any ideas why?
>
> No, you'll have to debug this the usual way. Run kamailio at a highe debug
> level to see what it is doing and log queries to the database. You'll have
> to
> make captures to see how the flow op sip messages is and who is requiring
> authentication.
>
> > Maybe you can give me more details about config on Kamailio server behind
> > LB?
>
> The config is equal to standard in this regard. Show us your config for the
> backend regarding authentication and registration.
>
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kamailio.cfg
Description: Binary data
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Re: [SR-Users] Kazoo like load balancing

[Jurijs Ivolga]

Hi,

And attachment. :)

With kind regards,

Jurijs

2015-07-21 19:18 GMT+03:00 Daniel Tryba :

> On Tuesday 21 July 2015 17:59:53 Jurijs Ivolga wrote:
> > So basically extension 100 sends invite to LB, LB then send invite to
> > Kamailio server and Kamailio server replies with 407 error.
> >
> > Do you have any ideas why?
>
> No, you'll have to debug this the usual way. Run kamailio at a highe debug
> level to see what it is doing and log queries to the database. You'll have
> to
> make captures to see how the flow op sip messages is and who is requiring
> authentication.
>
> > Maybe you can give me more details about config on Kamailio server behind
> > LB?
>
> The config is equal to standard in this regard. Show us your config for the
> backend regarding authentication and registration.
>
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>
#!KAMAILIO
#
# Kamailio (OpenSER) SIP Server v4.3 - default configuration script
# - web: http://www.kamailio.org
# - git: http://sip-router.org
#
# Direct your questions about this file to: 
#
# Refer to the Core CookBook at http://www.kamailio.org/wiki/
# for an explanation of possible statements, functions and parameters.
#
# Several features can be enabled using '#!define WITH_FEATURE' directives:
#
# *** To run in debug mode: 
# - define WITH_DEBUG
#
# *** To enable mysql: 
# - define WITH_MYSQL
#
# *** To enable authentication execute:
# - enable mysql
# - define WITH_AUTH
# - add users using 'kamctl'
#
# *** To enable IP authentication execute:
# - enable mysql
# - enable authentication
# - define WITH_IPAUTH
# - add IP addresses with group id '1' to 'address' table
#
# *** To enable persistent user location execute:
# - enable mysql
# - define WITH_USRLOCDB
#
# *** To enable presence server execute:
# - enable mysql
# - define WITH_PRESENCE
#
# *** To enable nat traversal execute:
# - define WITH_NAT
# - install RTPProxy: http://www.rtpproxy.org
# - start RTPProxy:
#rtpproxy -l _your_public_ip_ -s udp:localhost:7722
# - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING
#
# *** To enable PSTN gateway routing execute:
# - define WITH_PSTN
# - set the value of pstn.gw_ip
# - check route[PSTN] for regexp routing condition
#
# *** To enable database aliases lookup execute:
# - enable mysql
# - define WITH_ALIASDB
#
# *** To enable speed dial lookup execute:
# - enable mysql
# - define WITH_SPEEDDIAL
#
# *** To enable multi-domain support execute:
# - enable mysql
# - define WITH_MULTIDOMAIN
#
# *** To enable TLS support execute:
# - adjust CFGDIR/tls.cfg as needed
# - define WITH_TLS
#
# *** To enable XMLRPC support execute:
# - define WITH_XMLRPC
# - adjust route[XMLRPC] for access policy
#
# *** To enable anti-flood detection execute:
# - adjust pike and htable=>ipban settings as needed (default is
#   block if more than 16 requests in 2 seconds and ban for 300 seconds)
# - define WITH_ANTIFLOOD
#
# *** To block 3XX redirect replies execute:
# - define WITH_BLOCK3XX
#
# *** To enable VoiceMail routing execute:
# - define WITH_VOICEMAIL
# - set the value of voicemail.srv_ip
# - adjust the value of voicemail.srv_port
#
# *** To enhance accounting execute:
# - enable mysql
# - define WITH_ACCDB
# - add following columns to database
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_USRLOCDB
#!define WITH_MULTIDOMAIN
#!define WITH_NAT
#define WITH_NATSIPPING
#!ifdef ACCDB_COMMENT
  ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default '';
  ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT 
'';
  ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default '';
  ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT 
'';
#!endif

### Include Local Config If Exists #
import_file "kamailio-local.cfg"

### Defined Values #

# *** Value defines - IDs used later in config
#!ifdef WITH_MYSQL
# - database URL - used to connect to database server by modules such
#   as: auth_db, acc, usrloc, a.s.o.
#!ifndef DBURL
#!define DBURL "mysql://kamai

[SR-Users] Kamailio mod_destroy() ERROR

[Ali Taher]

Hello,

 

I'm getting below error when starting Kamailio.

 

ERROR: ctl [ctl.c:379]: mod_destroy(): ERROR: ctl: could not delete unix
socket /tmp/kamailio_ctl: Operation not permitted (1)

 

Any advice ?

 

Regards,

Ali

 



 

 

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Re: [SR-Users] dmq_usrloc replication and max_contacts

[Asgaroth]

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] "kamctl ul show"/xmlrpc ul.lookup AOR not found for valid registration

[Asgaroth]

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 15:11, Asgaroth wrote:

Hi All,

I have a situation where a device has successfully registered and its 
location saved/stored in memory.


I can see all the stored information when I issue a "kamctl ul show"

However, if I perform a kamctl ul show subscriber@domain, where the 
'subscriber@domain' is copied/pasted from the output of the "kamctl ul 
show" command, it results in an AOR not found error. However, with 
other AOR's on the same registrar, the "kamctl ul show 
subscriber@domain" works.


I checked the usernames for the failed commands, and I dont see 
anything obvious, they are a combination of lower/upper-case leters 
and digits. String length is around 24 characters for the username 
(the username table for the subscriber is 64 characters). I'm not sure 
what elese to check to see why the AOR lookup for these particular 
clients are failing when they are indeed registered. All other 
functions around the lookup seem to work, for example, a location 
lookup for these AOR's are working.


I wonder if I have come accross an issue with the kamctl function 
and/or the equivilient xmlrpc ul.lookup method.


Any thoughts on this would be greatly appreciated.

Thanks



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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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*** We're rebranding! Learn more here 
 ***

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Follow us on twitter @sipcentric 

Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered 
office: Faraday Wharf, Innovation Birmingham Campus, Holt Street, 
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Re: [SR-Users] dmq_usrloc replication and max_contacts

[Asgaroth]

Thanks Charles, I'll send some examples of what I am on about to you in 
the next 30 minutes or so.


On 22/07/2015 11:48, Charles Chance wrote:

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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*** We're rebranding! Learn more here 
 ***


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Follow us on twitter @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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Re: [SR-Users] About Acc_radius rad_sttaus Function

[ycaner]

Hello all;
i found problems. int acc_logic.c file at *acc_onack* function sends
"request structer" instead of "ack structer". When i changed  from
"acc_run_engines(req, 0, NULL);" to "acc_run_engines(ack, 0, NULL);" , it
sends right parameter to acc api engine. But  there is a problem in acc.c at
acc_run_engines function.


in the this row "if((type==0) && (msg->flags&(e->acc_flag))) {"   i deleted
msg->flags parameter and all work is good and radius send ACK (ALIVE)
packets.

Could you check it why we need msg->flags?

METHOD 1 INVITE
METHOD 2 ACK
METHOD 3 BYE

acc [acc.c:1020]: acc_run_engines(): acc_run_engine METHOD 1  type  0 flags
1 acc_flag 2 msg_flags 134217774
acc [acc.c:1020]: acc_run_engines(): acc_run_engine METHOD 2  type  0 flags
1 acc_flag 2 msg_flags 32
acc [acc.c:1020]: acc_run_engines(): acc_run_engine METHOD 3  type  0 flags
1 acc_flag 2 msg_flags 10



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Re: [SR-Users] New SIP INVITE from UAS (new dialogue)

[Joao Alves]

Hi again,

Following up on this and having done some further tests, I was finally able to 
generate a SIP INVITE request using the following command sequence on the 
command file.

$uac_req(method)="INVITE";
$uac_req(ruri)=$shv(userAcontact);
$uac_req(turi)="sip:1000@decasa.website";
$uac_req(furi)="sip:1001@decasa.website";
$uac_req(callid)=$var(call_id);

$uac_req(hdrs)="Contact: 
mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Alex 
Balashov
Sent: sexta-feira, 19 de Junho de 2015 22:11
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] New SIP INVITE from UAS (new dialogue)

On 06/19/2015 04:30 PM, Joao Alves wrote:

> Tell me one thing. If was not a SIP INVITE but just a SIP INFO or SIP
> MESSAGE (since are stateless), would it be easier to do?

Yes, those would be more viable.

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generated SIP INVITE.pcap
Description: generated SIP INVITE.pcap
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Re: [SR-Users] New SIP INVITE from UAS (new dialogue)

[Daniel Tryba]

On Wednesday 22 July 2015 12:37:58 Joao Alves wrote:
> The first, it seems we cannot send more than one header on the SIP INVITE.
> That is I need to add the "Contact", "Allow" and "Content-Type" fields, but
> only the last one is inserted, which is the "Content-Type". If I marked it
> as a comment (with #), them the "Allow" is included instead.

A quick solution could be:
$uac_req(hdrs)="Contact:..\r\nAllow:..\r\nContent-type: application/sdp\r\n";

> The second issue is related with the inclusion of SDP offer on the body of
> the message. Since this SDP is quite a long one(1266 bytes) it is being
> truncated. 

Where? How did you check this?

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Re: [SR-Users] New SIP INVITE from UAS (new dialogue)

[Joao Alves]

Hi Daniel,

For the first issue, your solution did work!

In relation with the SDP size, I originally just compared with the source one 
(see attached).  What I just did was to double check using an online tool and 
confirmed that the original has 1266 bytes (as also indicated by the Content's 
length) while the one sent has only 1077 bytes.

Many Thanks,
Joao

-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of 
Daniel Tryba
Sent: quarta-feira, 22 de Julho de 2015 13:58
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] New SIP INVITE from UAS (new dialogue)

On Wednesday 22 July 2015 12:37:58 Joao Alves wrote:
> The first, it seems we cannot send more than one header on the SIP INVITE.
> That is I need to add the "Contact", "Allow" and "Content-Type"
> fields, but only the last one is inserted, which is the
> "Content-Type". If I marked it as a comment (with #), them the "Allow" is 
> included instead.

A quick solution could be:
$uac_req(hdrs)="Contact:..\r\nAllow:..\r\nContent-type: application/sdp\r\n";

> The second issue is related with the inclusion of SDP offer on the
> body of the message. Since this SDP is quite a long one(1266 bytes) it
> is being truncated.

Where? How did you check this?

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v=0
o=- 3118595415729848979 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS tF3pDUWjeeQeaAe1WP5wdjheYDmKoCRvOAK0
m=audio 55888 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 10.233.194.135
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:4230081851 1 udp 2122260223 10.233.194.135 55888 typ host 
generation 0
a=ice-ufrag:O/ZcxBXjBxRUV2YZ
a=ice-pwd:k3YkJrfA9Vc+fflHmlYx4Owk
a=fingerprint:sha-256 
1A:96:4D:78:4C:AF:38:94:1A:F8:97:E5:D6:2C:B8:28:1C:99:9B:9B:32:86:95:37:E3:B8:22:60:FA:BC:DF:1D
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:4190739506 cname:/LGo3KIXN/RE8uKn
a=ssrc:4190739506 msid:tF3pDUWjeeQeaAe1WP5wdjheYDmKoCRvOAK0 
feeaa28b-8608-4468-a1e7-e95739017058
a=ssrc:4190739506 mslabel:tF3pDUWjeeQeaAe1WP5wdjheYDmKoCRvOAK0
a=ssrc:4190739506 label:feeaa28b-8608-4468-a1e7-e95739017058___
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[SR-Users] RTP Proxy to WAV Script conversion

[Alberto Sagredo]

Using RTPBreak and sox im able to convert rtpproxy rtp files to wav. Later
you could convert to mp3 if liked

Just to share with list

https://github.com/albersag/rtpproxy-utils

Any comment is well appreciated

Best Regards
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[SR-Users] dmq_is_from_node() issue

[Asgaroth]

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: 

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

[José Seabra]

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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Re: [SR-Users] dmq_is_from_node() issue

[Asgaroth]

sorry, I forgot to mention, this is with Kamailio v4.3.1:

# kamailio -V
version: kamailio 4.3.1 (x86_64/linux) f38e67
flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, 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: f38e67
compiled on 18:15:23 Jul 20 2015 with gcc 4.4.7

Thanks

On 22/07/2015 15:22, Asgaroth 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: 

Re: [SR-Users] [sr-dev] RFC: about the self signed tls certificates

[Alberto Sagredo]

+1 also here :)

2015-07-21 12:25 GMT+02:00 Alekzander Spiridonov :

> Agree
>
> 2015-07-20 22:18 GMT+03:00 Victor Seva  >:
>
>> On 07/20/2015 08:58 PM, Daniel-Constantin Mierla wrote:
>> > My proposal is to move generation of self signed certificates to kamctl.
>> > There can be a kamctl.tls file to be deployed by the tls package (same
>> > is done by kamctl.mysql, being part of mysql package), which should add
>> > a new group of commands, among them something like:
>> >
>> > kamctl tls generate-certificate
>> >
>> > Anyone with comments, pros/cons?
>>
>> +1 kamctl tls approach
>>
>>
>> ___
>> sr-dev mailing list
>> sr-...@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
>>
>>
>
>
> --
> Best regards,
> Alekzander Spiridonov
>
> ___
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>
>
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Re: [SR-Users] dmq_is_from_node() issue

[Robert Boisvert]

You might want to consider this function:
http://kamailio.org/docs/modules/4.3.x/modules/dmq.html#dmq.p.multi_notify

On Wed, Jul 22, 2015 at 10:25 AM, Asgaroth <00asgarot...@gmail.com> wrote:

> sorry, I forgot to mention, this is with Kamailio v4.3.1:
>
> # kamailio -V
> version: kamailio 4.3.1 (x86_64/linux) f38e67
> flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, 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: f38e67
> compiled on 18:15:23 Jul 20 2015 with gcc 4.4.7
>
> Thanks
>
>
> On 22/07/2015 15:22, Asgaroth 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: 

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: 

Re: [SR-Users] dmq_is_from_node() issue

[Asgaroth]

Hi Charles,

Thanks for the information, I was intending to use dmq_is_from_node as a 
security check to ensure that the dmq message arrived from a node that 
it saw as part of the notification address list (excluding itself of 
course).


I see that it is not intended to be used this way. I'll build on your 
snippet along with ipops to check these messages.


Thanks again.

On 22/07/2015 15:46, Charles Chance wrote:

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: 

Re: [SR-Users] dmq_is_from_node() issue

[Asgaroth]

Hi Robert,

Thanks for the suggestion, however, I already have that module parameter 
enabled as described in the original post :-) :


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)

Thanks

On 22/07/2015 15:49, Robert Boisvert wrote:
You might want to consider this function: 
http://kamailio.org/docs/modules/4.3.x/modules/dmq.html#dmq.p.multi_notify


On Wed, Jul 22, 2015 at 10:25 AM, Asgaroth <00asgarot...@gmail.com 
> wrote:


sorry, I forgot to mention, this is with Kamailio v4.3.1:

# kamailio -V
version: kamailio 4.3.1 (x86_64/linux) f38e67
flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, 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: f38e67
compiled on 18:15:23 Jul 20 2015 with gcc 4.4.7

Thanks


On 22/07/2015 15:22, Asgaroth 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: 

Re: [SR-Users] RTP Proxy to WAV Script conversion

[Daniel-Constantin Mierla]

Hello,

thanks for sharing! Shouldn't the end of the loop (the 'done') be more
at the end of the script in order to handle all the saves streams, or is
it intended to run for a single saved rtp session?

Cheers,
Daniel

On 22/07/15 16:18, Alberto Sagredo wrote:
> Using RTPBreak and sox im able to convert rtpproxy rtp files to wav.
> Later you could convert to mp3 if liked
>
> Just to share with list
>
> https://github.com/albersag/rtpproxy-utils
>
> Any comment is well appreciated
>
> Best Regards
>
>
>
>
> ___
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com

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Re: [SR-Users] New SIP INVITE from UAS (new dialogue)

[Daniel Tryba]

On Wednesday 22 July 2015 14:01:02 Joao Alves wrote:
> In relation with the SDP size, I originally just compared with the source
> one (see attached).  What I just did was to double check using an online
> tool and confirmed that the original has 1266 bytes (as also indicated by
> the Content's length) while the one sent has only 1077 bytes.

I saw the capture after my message. The message is fragmented and not shown 
correctly in the capture. There shouldn't be any problem (unless there are 
network issues between you and destination).

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Re: [SR-Users] Kazoo like load balancing

[Jurijs Ivolga]

Hi Daniel,

First of all, thank you for your help!

Below you can find more detailed info:

I have following infrastructure:

22@10.50.210.9:6574 - UAC
11@10.50.210.9:50365 - UAC

10.101.237.133 - Loadbalancer Kamailio
10.101.241.55 - Kamailio Proxy, where UACs are registered

So this is how Invite travels

Invite (1)22@10.50.210.9:6574 => 10.101.237.133 - Loadbalancer
Invite (2)10.101.237.133 - Loadbalancer => 10.101.241.55 - Kamailio Proxy
Invite (3)10.101.241.55 - Kamailio Proxy => 10.101.237.133 - Loadbalancer
Invite (4)10.101.237.133 - Loadbalancer => 10.101.241.55 - Kamailio Proxy
407 Error (5)10.101.241.55 - Kamailio Proxy => 10.101.237.133 -
Loadbalancer

So invite get stuck at LB 10.101.237.133, it should send invite to
11@10.50.210.9:50365 at step (4) but it sends to Proxy back instead. After
this proxy replies with 407 error.

Please find full SIP trace in attachment.

Thank you!

With kind regards,

Jurijs

2015-07-21 19:18 GMT+03:00 Daniel Tryba :

> On Tuesday 21 July 2015 17:59:53 Jurijs Ivolga wrote:
> > So basically extension 100 sends invite to LB, LB then send invite to
> > Kamailio server and Kamailio server replies with 407 error.
> >
> > Do you have any ideas why?
>
> No, you'll have to debug this the usual way. Run kamailio at a highe debug
> level to see what it is doing and log queries to the database. You'll have
> to
> make captures to see how the flow op sip messages is and who is requiring
> authentication.
>
> > Maybe you can give me more details about config on Kamailio server behind
> > LB?
>
> The config is equal to standard in this regard. Show us your config for the
> backend regarding authentication and registration.
>
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>
#
U 2015/07/22 10:46:19.362379 10.50.210.9:6574 -> 10.101.237.133:5060
INVITE sip:1...@voipdev.mydomain.net SIP/2.0.
Via: SIP/2.0/UDP 192.168.5.148:55529;branch=z9hG4bK.Lp01Vj8qs;rport.
From: ;tag=AZWk8e87s.
To: sip:1...@voipdev.mydomain.net.
CSeq: 20 INVITE.
Call-ID: LJoKbYM0YY.
Max-Forwards: 70.
Supported: outbound.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO, UPDATE.
Content-Type: application/sdp.
Content-Length: 557.
Contact: 
;+sip.instance="".
User-Agent: LinphoneAndroid/2.4.1 (belle-sip/1.4.1).
.
v=0.
o=22 4026 1869 IN IP4 192.168.5.148.
s=Talk.
c=IN IP4 192.168.5.148.
b=AS:380.
t=0 0.
a=rtcp-xr:rcvr-rtt=all:1 stat-summary=loss,dup,jitt,TTL voip-metrics.
m=audio 7076 RTP/AVP 96 97 98 99 0 8 101 100 102.
a=rtpmap:96 opus/48000/2.
a=fmtp:96 useinbandfec=1; stereo=0; sprop-stereo=0.
a=rtpmap:97 SILK/16000.
a=rtpmap:98 speex/16000.
a=fmtp:98 vbr=on.
a=rtpmap:99 speex/8000.
a=fmtp:99 vbr=on.
a=rtpmap:101 telephone-event/48000.
a=rtpmap:100 telephone-event/16000.
a=rtpmap:102 telephone-event/8000.
m=video 9078 RTP/AVP 96.
a=rtpmap:96 VP8/9.

#
U 2015/07/22 10:46:19.364240 10.101.237.133:5060 -> 10.50.210.9:6574
SIP/2.0 100 trying -- your call is important to us.
Via: SIP/2.0/UDP 
192.168.5.148:55529;branch=z9hG4bK.Lp01Vj8qs;rport=6574;received=10.50.210.9.
From: ;tag=AZWk8e87s.
To: sip:1...@voipdev.mydomain.net.
CSeq: 20 INVITE.
Call-ID: LJoKbYM0YY.
Server: kamailio (4.3.1 (x86_64/linux)).
Content-Length: 0.
.

#
U 2015/07/22 10:46:19.366730 10.101.237.133:5060 -> 10.101.241.55:5060
INVITE sip:1...@voipdev.mydomain.net SIP/2.0.
Record-Route: .
Via: SIP/2.0/UDP 
10.101.237.133;branch=z9hG4bK8f82.23ebd7beee7641516f01572434199e7f.0.
Via: SIP/2.0/UDP 
192.168.5.148:55529;received=10.50.210.9;branch=z9hG4bK.Lp01Vj8qs;rport=6574.
From: ;tag=AZWk8e87s.
To: sip:1...@voipdev.mydomain.net.
CSeq: 20 INVITE.
Call-ID: LJoKbYM0YY.
Max-Forwards: 70.
Supported: outbound.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO, UPDATE.
Content-Type: application/sdp.
Content-Length: 557.
Contact: 
;+sip.instance="".
User-Agent: LinphoneAndroid/2.4.1 (belle-sip/1.4.1).
Path: .
.
v=0.
o=22 4026 1869 IN IP4 192.168.5.148.
s=Talk.
c=IN IP4 192.168.5.148.
b=AS:380.
t=0 0.
a=rtcp-xr:rcvr-rtt=all:1 stat-summary=loss,dup,jitt,TTL voip-metrics.
m=audio 7076 RTP/AVP 96 97 98 99 0 8 101 100 102.
a=rtpmap:96 opus/48000/2.
a=fmtp:96 useinbandfec=1; stereo=0; sprop-stereo=0.
a=rtpmap:97 SILK/16000.
a=rtpmap:98 speex/16000.
a=fmtp:98 vbr=on.
a=rtpmap:99 speex/8000.
a=fmtp:99 vbr=on.
a=rtpmap:101 telephone-event/48000.
a=rtpmap:100 telephone-event/16000.
a=rtpmap:102 telephone-event/8000.
m=video 9078 RTP/AVP 96.
a=rtpmap:96 VP8/9.

#
U 2015/07/22 10:46:19.370363 10.101.241.55:5060 -> 10.101.237.133:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 
10.101.237.133;branch=z9hG4bK8f82.23ebd7beee7641516f01572434199e7f.0;rport=5060.
Via: SIP/2.0/UDP 
192.168.5.148:55529;received=10.50.210.9;branch=z9hG4bK.Lp01Vj8qs;rport=6574.
From: ;tag=AZWk8e87s.
To: sip:1...@voipdev.mydomain.n

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


-- 
*Charles Chance*
Managing Director

t. 0121 285 4400m. 07932 063 891

-- 
*** We're rebranding! Learn more here 
 ***

www.sipcentric.com

Follow us on twitter @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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Re: [SR-Users] New SIP INVITE from UAS (new dialogue)

[Joao Alves]

Hi Daniel,

Yes, you're right. It was fragmented at UDP level. I've repeated with TCP as 
transport and the SDP is complete. 

Thanks again,
Joao 

-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of 
Daniel Tryba
Sent: quarta-feira, 22 de Julho de 2015 16:39
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] New SIP INVITE from UAS (new dialogue)

On Wednesday 22 July 2015 14:01:02 Joao Alves wrote:
> In relation with the SDP size, I originally just compared with the 
> source one (see attached).  What I just did was to double check using 
> an online tool and confirmed that the original has 1266 bytes (as also 
> indicated by the Content's length) while the one sent has only 1077 bytes.

I saw the capture after my message. The message is fragmented and not shown 
correctly in the capture. There shouldn't be any problem (unless there are 
network issues between you and destination).

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This message and the information contained herein is proprietary and 
confidential and subject to the Amdocs policy statement,
you may review at http://www.amdocs.com/email_disclaimer.asp

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Re: [SR-Users] Port 5061 Issue

[Daniel-Constantin Mierla]

Hello,

run with debug=3 and look at the syslog to see if there are some
messages giving you a clue about what happens.

Cheers,
Daniel

On 19/06/15 20:45, Ankhit Vivekananda wrote:
> Hello,
>
> I have configured kamailio 4.0.4 long back. Now, I wanted to enable
> the usage of port 5061 with TLS. By doing so, the client(video camera)
> can place outgoing calls fine. But while receiving an incoming call,
> the message does not get sent out to the other client. Hence, the call
> fails. I can place and receive calls on port 5060 successfully without
> any issues. Can anyone please tell me why port 5061 is giving me a
> problem.
>
> Also, as a test, I set up kamailio on another machine(version 4.1) and
> 5061 works fine in all directions. It's only on the other machine that
> I have problems with. 
>
> Appreciate any help!
>
> Thank you,
> Ankhit
>
>
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-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com

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Re: [SR-Users] Kamailio mod_destroy() ERROR

[Ali Taher]

Any help regarding the below would be appreciated.

 

BR,

Ali

From: Ali Taher [mailto:ata...@vanrise.com] 
Sent: Wednesday, July 22, 2015 1:39 PM
To: 'Kamailio (SER) - Users Mailing List' 
Cc: 'Ali Taher~Vanrise Technical Support' 
Subject: Kamailio mod_destroy() ERROR

 

Hello,

 

I'm getting below error when starting Kamailio.

 

ERROR: ctl [ctl.c:379]: mod_destroy(): ERROR: ctl: could not delete unix
socket /tmp/kamailio_ctl: Operation not permitted (1)

 

Any advice ?

 

Regards,

Ali

 



 

 

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