Re: [OpenSIPS-Users] rest_append_hf

[Pasan Meemaduma via Users]

Hi Liviu,
Apologies for the delayed reply, But your fix works, I have tested and its 
working as expected now. Thank you.
 

On Friday, 16 February 2018, 7:25, Pasan Meemaduma via Users 
 wrote:
 

 HI Liviu,
Thanks for the prompt reply as you guys always do :). I'll rebuild my opensips 
and test again. I'll get back to you with results.

 

On Thursday, 15 February 2018, 22:38, Liviu Chircu  
wrote:
 

  Hi Pasan, Thank you for the report - I have spotted and fixed [1] a bug in 
rest_append_hf() that could definitely lead to a world of problems. Can you try 
to pull the latest version of 2.3 and try again? If you're running off 
packages, the only option is to wait until tomorrow and deploy the nightly 
built ones.  Cheers,
  [1]: https://github.com/OpenSIPS/opensips/commit/888d5e9b3f9
  Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com On 15.02.2018 11:53, Pasan Meemaduma via 
Users wrote:
  
  Hi List, 
  I'm trying to use rest_append_hf function to append an extra header for my 
rest call. But it doesn't set it unless I call the same function twice. 
  How do I get around this ? I tried async() statement as well as per docs but 
it didn't work either. 
  I'm using opensips 2.3.3 
  route();
  
  route[] {
     rest_append_hf("Accept: application/vnd.xxx.vapi+json; version=2");
     $var(rc) = rest_post("http://{{ API_USER }}:{{ API_SECRET }}@{{ API_HOST 
}}/xx/xx/xx", "{ \"xx\": { \"xx\": [ \"$var(auth_user)\" ] }}", 
"application/json", "$var(body)", "$var(ct)", "$var(rcode)");
     $json(res) := $var(body);
     if($json(res/success) != 1 ) {
     xlog("L_ERR", "API: Failed to Blocked call request - M=$rm RURI=$ru 
F=$fu T=$tu IP=$si ID=$ci\n");
     } else {
     xlog("L_WARN", "API: Blocked call request - M=$rm RURI=$ru F=$fu T=$tu 
IP=$si ID=$ci\n");
     }
 }
 
  Accept header will only send if I call route() twice :/. 
  
   
  
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[OpenSIPS-Users] Opensips module like htable in kamailio

[Kirill Galinurov]

Hi. we need some module to store some keys like  %ip -> { port => 15643,

  username => 'test',

  domain => 'test',

  auth_count => '2'}
to do  protect against dictionary attacks. Like in Kamaillio.
What is the best module to do this in   Opensips ?
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Re: [OpenSIPS-Users] dialog replication

[Pasan Meemaduma via Users]

Hi Vlad, 
Thanks for your tip. I'll give it try. 

On Tuesday, 6 March 2018, 19:57, Vlad Patrascu  wrote:
 

  Hi Pasan, You could have another OpenSIPS instance that exposes the service 
IP, in front of the 2 clustered HA nodes. Calls would first land on this box 
and then would be dispatched to one of the 2 floating IPs. You could for 
example hash over the From tag or Callid in order to send to the same IP.
  Regards,
  Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com
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Re: [OpenSIPS-Users] SNMPstats not working

[Alexey K . via Users]

upd:
created /etc/snmp/snmp.conf like this:


mibs +OPENSER-MIB
mibs +OPENSER-REG-MIB
mibs +OPENSER-SIP-COMMON-MIB
mibs +OPENSER-SIP-SERVER-MIB
mibs +OPENSER-TC As for now, at least this works:

sip01 ~ # snmptranslate -IR -On openserSIPRegStats
.1.3.6.1.4.1.27483.3.1.2.1.6

sip01 ~ # snmpwalk -v 2c -c public localhost openserSIPRegCfg
OPENSER-SIP-SERVER-MIB::openserSIPRegCfg = No more variables left in this MIB 
View (It is past the end of the MIB tree)

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[OpenSIPS-Users] topology_hinding & load_balancer

[Mirko Csiky]

Hi community,
since we activated the topology hiding module in our load_balance test
scenario (sipp uac -> opensips & load_balance ->  5x - sipp uas), load
balance functionality always choose the same destination (from 5 possible).
I can not see the link between topology_hiding &
load_balance_destination_choice, why is it so ?
Does anyone have experience with topology_hiding&load_balance  /
workaround-idea ?
best regards
Mirko
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Re: [OpenSIPS-Users] opensips 2.3.3 rtpproxy behind the NAT

[Callum Guy]

Apologies, got a little carried away there...

*CALLER *- client--NAT--|
*PROXY  *-  |--NAT--Opensips&Rtpproxy
*CALLED *- client--NAT--|

A little more detail required for full assistance however given your
diagram if the caller hopes to get in touch with the called party you will
need to get rtpproxy working in bridging mode such that it opens two
sessions, one for each leg, each traversing NAT. This article may help:
https://saevolgo.blogspot.co.uk/2013/08/rtpproxy-revisited-kamailio-40.html

On Tue, Mar 6, 2018 at 2:29 PM Callum Guy  wrote:

> Ah, zee NAT, sacre bleu!
>
>
>
> On Tue, Mar 6, 2018 at 2:27 PM 2289959...@qq.com <2289959...@qq.com>
> wrote:
>
>> Dears
>>  I have a environment
>>
>> client--NAT--|
>>   |NAT--Opensips&Rtpproxy
>> client--NAT--|
>>
>> so rtpproxy and opensips in same server , and behind zhe NAT.
>> How can i do?   now just config listen.
>> Can do sip protocol, but rtp no sound.
>> No Error Message. Please help me Thanks.
>>
>> --
>> BrightLin
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>>
> --
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> Head of Information Security
> X-on
>
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Re: [OpenSIPS-Users] opensips 2.3.3 rtpproxy behind the NAT

[Callum Guy]

Ah, zee NAT, sacre bleu!



On Tue, Mar 6, 2018 at 2:27 PM 2289959...@qq.com <2289959...@qq.com> wrote:

> Dears
>  I have a environment
>
> client--NAT--|
>   |NAT--Opensips&Rtpproxy
> client--NAT--|
>
> so rtpproxy and opensips in same server , and behind zhe NAT.
> How can i do?   now just config listen.
> Can do sip protocol, but rtp no sound.
> No Error Message. Please help me Thanks.
>
> --
> BrightLin
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The information in this e-mail is confidential and for use by the 
addressee(s) only. If you are not the intended recipient, please notify 
X-on immediately on +44(0)333 332  and delete the
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Views 
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Re: [OpenSIPS-Users] mid_registrar

[Slava Bendersky]

Sorry, I was pulled away to another work. I will provide info soon, but 
I liked more first approach with rid (just make it short), because it 
not modify Caller ID or contact header. Right now I noticed when  
contact overwritten it use long...@domain.com;params  that causing show 
on the phone wrong caller and record it into history log in client as 
well to CDR. Example if  I call from ext 4300 to external number 
12345678901 record will be long_id not 4300. The only way I was be able 
set original extension is in mode 2 AOR throttling which not use 
contactid.


 REGISTRAR module
loadmodule "mid_registrar.so"
modparam("mid_registrar", "mode", 2)
modparam("mid_registrar", "insertion_mode", 0)
modparam("mid_registrar", "received_avp", "$avp(RECEIVED)")
modparam("mid_registrar", "max_contacts", 10)
modparam("mid_registrar", "tcp_persistent_flag", 
"TCP_PERSIST_REGISTRATIONS")

modparam("mid_registrar", "outgoing_expires", 900)
# Old way
#modparam("mid_registrar", "contact_match_param", "rid")


On Tue, Feb 20, 2018 at 5:54 AM, Liviu Chircu  
wrote:

Hi, volga!

Can you send a SIP trace of this failed call to my mailbox? It's one 
of those "it's working for me" situations. The mid_reg_lookup() 
should be smart enough to locate a contact based on its unique 
ContactID, when the call comes in from FS.


Regarding regid: it's completely gone now (even in 2.3), thanks for 
the heads up: I'll clean up the docs asap!


Cheers,

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 19.02.2018 22:51, volga...@networklab.ca wrote:

Hello Everyone,
I am back to test mid registrar and having issue with


modparam("mid_registrar", "contact_match_param", "regid")
I so comment this parmater is obsolote what correct way to set it, 
because opensips reject calls with 404 on location lookup. I see 
that contact header is replaced with CallerID and when freeswitch 
send reply it can't locate the cantact.

Original contact
root@casbc00 ~> [/etc/opensips]# opensipsctl ul show
Domain:: location table=512 records=2
AOR:: 4300
Contact:: sip:4300@192.168.89.227:34846;transport=tcp Q=
ContactID:: 1205768431484279123
Expires:: 1478
Callid:: 567846761-2141...@bjc.bgi.ij.cch
Cseq:: 2027
User-agent:: Grandstream GXP2140 1.0.8.47
State:: CS_NEW
Flags:: 0
Cflags::
Socket:: tcp:158.69.151.95:5082
Methods:: 6015

Feb 19 15:45:20 casbc00 /usr/sbin/opensips[2049]: looking up 
[sip:1205768431484279123@10.18.130.27:5060]
Feb 19 15:45:20 casbc00 /usr/sbin/opensips[2049]: 
ERROR:mid_registrar:mid_reg_lookup: no record found for 
sip:1205768431484279123@10.18.130.27:5060, ci: 
a413e68c-9058-1236-7e9c-5254003e39bb

Github report https://github.com/OpenSIPS/opensips/issues/


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Re: [OpenSIPS-Users] dialog replication

[Vlad Patrascu]

Hi Pasan,

You could have another OpenSIPS instance that exposes the service IP, in 
front of the 2 clustered HA nodes. Calls would first land on this box 
and then would be dispatched to one of the 2 floating IPs. You could for 
example hash over the From tag or Callid in order to send to the same IP.


Regards,

Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com

On 01.03.2018 03:21, Pasan Meemaduma via Users wrote:

Hi Vlad,

Thanks for your response. I think I'll have to wait until 2.4 
transaction replication come in to action if I am to go with anycast 
ip. I didn't understand how it would solve with 2 moveable ips. how do 
i make sure dialog will tie to a particular ip ?



On Wednesday, 28 February 2018, 17:04, Vlad Patrascu 
 wrote:



Hi,
Yes, you are correct, this is currently a limitation. But it could 
also work in an "active/active" sort of setup if you have 2 movable 
IPs and each dialog is tied to one of the IPs. Full anycast support 
though is coming up in OpenSIPS 2.4.

Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com 
On 28.02.2018 04:46, Pasan Meemaduma via Users wrote:

Hi Vlad,

On a second thought its how anycast should work, packets could get to 
the closest node so basically with 2.3.3 we can't replicate 
transactions across mulitple opensips servers ? dialog replication  
would only work for an active/passive setup doesn't it ?




On Wednesday, 28 February 2018, 7:36, Pasan Meemaduma via Users 
  wrote:



Hi Vlad,

That explains the issue then, my sip listener ip is an anycast one, 
and for some weired reason replies for INVITE that leaves node2 
receive by node1 hence it can't create the dialog. I checked with 
node2 being down and recovered and calls init via node1 always 
received final reply hence dialog replication works. So it means my 
anycast configuration is broken right ? Thanks for you time to look 
in to it.



On Tuesday, 27 February 2018, 21:42, Vlad Patrascu 
  wrote:



Hi Pasan,
I don't see anything in the logs or your cluster configuration which 
could indicate that the dialogs don't replicate. Are you sure that 
the problem is consistently reproducible and that the INVITE for the 
call in question receives a final reply? Dialogs are replicated when 
they get to the confirmed state, and judging from the logs, this 
doesn't appear to be happening for that call.

Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com 
On 27.02.2018 04:15, Pasan Meemaduma via Users wrote:

Hi Vlad,

I have sent you the full debug logs as requested. Also 
clusterer_list mi command gives following output when dialog 
replication stopped working.


node1>>opensipsctl fifo clusterer_list
Cluster:: 1
    Node:: 2 DB_ID=5 URL=bin:10.3.1.137:5566 Enabled=1 
Link_state=Up Next_hop=2 Description=Node 2



 node2>> opensipsctl fifo clusterer_list
Cluster:: 1
    Node:: 1 DB_ID=4 URL=bin:10.3.1.136:5566 Enabled=1 
Link_state=Up Next_hop=1 Description=Node 1

1



On Monday, 26 February 2018, 18:02, Vlad Patrascu 
  wrote:



Hi,
Can you send the full logs for both nodes from the time that node1 
restarts onwards? Also, what is the output of 'clusterer_list' mi 
command on the instances?

Regards,
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com 
On 26.02.2018 08:38, Pasan Meemaduma via Users wrote:

Hi Guys,

Its me again :). I'm using dialog replication in opensips 2.3.3 and 
it appears it doesn't work after a node goes down and come back 
online. The recovered node

doesn't seems to receiving dialog info via binary interface.

I have node1 and node2 with dialog replication on. everything works 
fine and if I shutdown node1 and bring it back online after a while

node2 doesn't send new call dialog info via binary interface.

I have attach the debug trace from both nodes, if you need anything 
else let me know. I'm also using an anycast ip as the service ip.



on node2 for new call

Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: 
DBG:dialog:build_new_dlg: new dialog 0x7f188afd3bf8 
(c=ktQ0Pkdwz50qGYjED6Brpw..,f=sip:X@somedomain;transport=UDP,t=sip:+YYY@somedomain;transport=UDP,ft=2b161508) 
on hash 2317
Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: 
DBG:dialog:init_leg_info: route_set , contact 
sip:X@192.168.27.11:56419;transport=UDP 
, cseq 2 
and bind_addr udp:10.3.3.1:5060
Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: 
DBG:dialog:dlg_add_leg_info: set leg 0 for 0x7f188afd3bf8: 
tag=<2b161508> rcseq=<0>
Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: 
DBG:dialog:link_dlg: ref dlg 0x7f188afd3bf8 with 3 -> 3 in h_entry 
0x7f188afc3828 - 2317
Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: 
D

[OpenSIPS-Users] rl_check in failure route

[William Luke]

Hi there,

Hoping for some guidance.

We have multiple carriers we place calls out to the PSTN over, and some have 
CPS limits which we check using rl_check and then choose a different carrier if 
we’re over the limit.

This all works great.

Sometimes calls fail with reasons that would mean we want to retry using a 
different carrier, so we need to handle this in the failure route.

However the documentation states that rl_check can only be used in the request 
route, so I’m not sure how we can check for limits when a call has failed and a 
new carrier is being selected.

We’ve a route that handles carrier selection and does the checking, and I’ve 
tried routing to that inside the failure route and it all appears to work, but 
I’m guessing something is going to break so wanted to know if there was a way 
to do what I’m looking to do?

Many thanks,

William

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[OpenSIPS-Users] opensips 2.3.3 rtpproxy behind the NAT

[2289959...@qq.com]

Dears
 I have a environment 

client--NAT--|
  |NAT--Opensips&Rtpproxy
client--NAT--|

so rtpproxy and opensips in same server , and behind zhe NAT.
How can i do?   now just config listen.
Can do sip protocol, but rtp no sound.
No Error Message. Please help me Thanks.



BrightLin
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[OpenSIPS-Users] SNMPstats not working

[Alexey K . via Users]

Hi all.

- CentOS 7.4.1708
- OpenSIPS 2.2.5
- configured as written here  
http://www.opensips.org/html/docs/modules/2.2.x/snmpstats.html  and 
http://saevolgo.blogspot.ru/2012/09/opensips-monitoring-using-snmp-part-i.html
-  OPENSER-MIB, OPENSER-REG-MIB, OPENSER-SIP-COMMON-MIB, 
OPENSER-SIP-SERVER-MIB, OPENSER-TC - put from opensips-2.2.5.gz to all MIBs 
location (/usr/share/snmp/mibs/)

opensips.cfg
***
loadmodule "snmpstats.so"
modparam("snmpstats", "snmpgetPath", "/usr/bin/")
modparam("snmpstats", "sipEntityType", "registrarServer")

netstat OK:
***
sip01 ~ # netstat -tulpn | grep -e opens -e snm
tcp 0 0 127.0.0.1:705 0.0.0.0:* LISTEN 26539/snmpd 
tcp 0 0 127.0.0.1:199 0.0.0.0:* LISTEN 26539/snmpd 
udp 0 0 10.5.10.21:5060 0.0.0.0:* 26569/opensips 
udp 0 0 0.0.0.0:161 0.0.0.0:* 26539/snmpd opensipsctl fifo OK:
***
sip01 ~ # opensipsctl fifo ps
Process:: ID=0 PID=26569 Type=attendant
Process:: ID=1 PID=26571 Type=SNMP AgentX
...
But this command shows no output:
***
snmpd -f -Dagentx -x tcp:localhost:705 2>&1


And as a result - SNMP monitoring does not work.

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Re: [OpenSIPS-Users] Need help to print Shared memory and package memory used in opensips in logs .

[Liviu Chircu]

Hi Sasmita,

Hmmm... indeed, the error is somewhat unhelpful to the script writer. 
The problem is that "statistics.so" is not loaded, hence "pvar $stat... 
is not defined". A similar thing happens with $json. I, myself, was a 
bit puzzled when I first hit this error.


Best regards,

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 06.03.2018 13:02, Sasmita Panda wrote:

version: opensips 1.11.2-notls (x86_64/linux)
flags: STATS: On, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, 
SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
MAX_URI_SIZE 1024, BUF_SIZE 65535

poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
svn revision: 2537:3262M
main.c compiled on 13:20:22 Nov 15 2017 with gcc 4.8.3


This is what exactly I am using .


*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, Mar 6, 2018 at 4:27 PM, Sasmita Panda > wrote:


Hi All ,

I am using opensips-1.11.2  .

*xlog("Real SHMEM used size is $stat(real_used_size) \n");*


Trying to print logs through this but its not working . giving
below error .
*
*
* ERROR:core:pv_parse_spec: pvar "stat"(inner_name) not found*
* ERROR:core:pv_parse_spec: wrong char [$/36] in
[$stat($real_used_size) #012] at [6 (5)]*
* ERROR:core:pv_parse_format: parsing PV spec failed*
* ERROR:core:fix_actions: wrong format [Real SHMEM used size is
$stat($real_used_size) #012] for value param*
* ERROR:core:fix_actions: fixing failed (code=-5) at cfg line 78*
* ERROR:core:main: failed to fix configuration with err code -5*
* INFO:core:cleanup: cleanup*

Please help me how to do this . Thanks in advance .



*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/




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Re: [OpenSIPS-Users] Need help to print Shared memory and package memory used in opensips in logs .

[Sasmita Panda]

version: opensips 1.11.2-notls (x86_64/linux)
flags: STATS: On, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM,
SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
svn revision: 2537:3262M
main.c compiled on 13:20:22 Nov 15 2017 with gcc 4.8.3


This is what exactly I am using .


*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*

On Tue, Mar 6, 2018 at 4:27 PM, Sasmita Panda  wrote:

> Hi All ,
>
> I am using opensips-1.11.2  .
>
> *xlog("Real SHMEM used size is $stat(real_used_size) \n");*
>
>
> Trying to print logs through this but its not working . giving below error
> .
>
> * ERROR:core:pv_parse_spec: pvar "stat"(inner_name) not found*
> * ERROR:core:pv_parse_spec: wrong char [$/36] in [$stat($real_used_size)
> #012] at [6 (5)]*
> * ERROR:core:pv_parse_format: parsing PV spec failed*
> * ERROR:core:fix_actions: wrong format [Real SHMEM used size is
> $stat($real_used_size) #012] for value param*
> * ERROR:core:fix_actions: fixing failed (code=-5) at cfg line 78*
> * ERROR:core:main: failed to fix configuration with err code -5*
> * INFO:core:cleanup: cleanup*
>
> Please help me how to do this . Thanks in advance .
>
>
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
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[OpenSIPS-Users] Need help to print Shared memory and package memory used in opensips in logs .

[Sasmita Panda]

Hi All ,

I am using opensips-1.11.2  .

*xlog("Real SHMEM used size is $stat(real_used_size) \n");*


Trying to print logs through this but its not working . giving below error .

* ERROR:core:pv_parse_spec: pvar "stat"(inner_name) not found*
* ERROR:core:pv_parse_spec: wrong char [$/36] in [$stat($real_used_size)
#012] at [6 (5)]*
* ERROR:core:pv_parse_format: parsing PV spec failed*
* ERROR:core:fix_actions: wrong format [Real SHMEM used size is
$stat($real_used_size) #012] for value param*
* ERROR:core:fix_actions: fixing failed (code=-5) at cfg line 78*
* ERROR:core:main: failed to fix configuration with err code -5*
* INFO:core:cleanup: cleanup*

Please help me how to do this . Thanks in advance .



*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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Re: [OpenSIPS-Users] Username in DB URL

[Bogdan-Andrei Iancu]

Just as an update, the fix was confirmed and backported on 2.2 and 2.3 also.

Thanks Olle!

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
OpenSIPS Summit 2018
  http://www.opensips.org/events/Summit-2018Amsterdam

On 03/02/2018 07:32 PM, Bogdan-Andrei Iancu wrote:

Hi Olle,

Could you give a try to this fix:
https://github.com/OpenSIPS/opensips/commit/5b0bc1e30ebfc1e67f4967c3e66cfa7577ee5b58

Currently this is available only on trunk version (still you can use 
the nightly builds from the official repos). If your tests are 
positive, I will do a backport ;)


Best regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   http://www.opensips-solutions.com
OpenSIPS Summit 2018
   http://www.opensips.org/events/Summit-2018Amsterdam
On 02/13/2018 04:11 PM, Olle Frimanson wrote:


Hi, thanks for the reply. We try to avoid installing from source and 
rather use RPM’s. We are also looking into using a SQL proxy and have 
that locally on the opensips server. But the very best solution would 
be to escape somehow.


I think there was a similar problem with the TLS crypto suite that 
was fixed in a nice way.


And I don’t think we can get Microsoft to change the usernames for 
MySQL J


BR / Olle

*Från:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Skickat:* den 6 februari 2018 18:20
*Till:* OpenSIPS users mailling list ; Olle 
Frimanson 

*Ämne:* Re: [OpenSIPS-Users] Username in DB URL

Hi Olle,

As Alex said there is no work around for this for now. Still the 
problem can be addressed by coding and it is not so hard to do. The 
question is what is the best/proper type of escaping to be supported 
into the URL - I guess the %xy is the proper one for an URL, as per 
RFC 1738 ).


Have you installed opensips form sources ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
   http://www.opensips-solutions.com
OpenSIPS Summit 2018
   http://www.opensips.org/events/Summit-2018Amsterdam

On 02/05/2018 08:08 PM, Olle Frimanson wrote:

Hi, we are experiencing an issue with a roleout in Microsoft
Azure, where the DB  username contains a ”@” sign. This seems to
be the default in MySQL in Azure.  Is there anyway to escape this
the URL for the clusterer module?

If you use mysql client it’s typical something like

Mysql -h host.mysql.azure.com -u opensips@host -popensips
opensips, which works fine

In clustererar where we use this this translates to:

modparam("clusterer",
"db_url","mysql://opensips@host:opens...@host.mysql.azure.com/opensips"
)

which fails

Any ideas on a workaround on this?

BR / Olle




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