Re: [OpenSIPS-Users] OpenSIPS Unable to parse msg received from

2018-08-31 Thread Serge S. Yuriev 

Hi

For me it's look like obviously fragmentation issue - you get decoupled 
parts of one large message. You can check via tcpdump IP headers 
correctly formatted. Try to make messages smaller or/and switch to TCP.


On 23/08/18 17:25, Abdul Basit wrote:
I upgraded home proxy server opensips just in case it may help resolving 
the issues.


# opensips -V
version: *opensips 2.4.2* (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
MAX_URI_SIZE 1024, BUF_SIZE 65535

poll method support: poll, epoll, sigio_rt, select.
main.c compiled on  with gcc 6.3.0

But result is same. I am continuously getting parse_mgs errors.
any idea about fixation of the issue?

--
regards,

abdul basit


On Thu, 23 Aug 2018 at 16:52, Abdul Basit > wrote:


Hi Alain Bieuzent,

I added sip msg validate code in start of main route but i didn't
get these additional log lines.
Still getting messages from opensips core:

Aug 23 13:22:24 homer-proxy /usr/sbin/opensips[25568]:
ERROR:core:receive_msg: Unable to parse msg received from
[x.x.x.x:61648]
Aug 23 13:22:27 homer-proxy /usr/sbin/opensips[25568]:
ERROR:core:parse_msg:
message=;tag=1c1737546331#015#012To:
#015#012Call-ID: c@x.x.x.x#015#012CSeq:
63697 REGISTER#015#012Authorization: Digest

username="yy",realm="x.x.x.x",nonce="5b7e9911db6acf694065c578ff32fa51ce24bb374d67",uri="sip:x.x.x.x",algorithm=MD5,response="debec3f66d2ac1ec488138871ffa1d60"#015#012Contact:
;expires=360#015#012Supported:
em,timer,replaces,path,resource-priority#015#012Allow:

REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE#015#012Expires:
360#015#012User-Agent: Audiocodes-Sip-Gateway-MP-118
FXS/v.5.20A.043.005#015#012Content-Length: 0#015#012#015#012>

Hi Bogdan,

Yes, I did it as per cfg in shared link. But again i am getting the
parse error messages. Looks like request is initiating from some
buggy sip client.

Aug 23 13:43:39 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:parse_first_line: bad request first line
Aug 23 13:43:39 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:parse_first_line: at line 0 char 34:
Aug 23 13:43:39 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:parse_first_line: parsed so far: AudiocodesGW 1372031417
1372031297
Aug 23 13:43:39 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:parse_msg: message=
Aug 23 13:43:39 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:receive_msg: Unable to parse msg received from
[x.x.x.x:64443]
Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:parse_to_param: unexpected char [C] in status 13:
<<;tag=#015#012>> .
Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:parse_to: unexpected end of header in state 13
Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:get_hdr_field: bad to header
Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:parse_to_param: unexpected char [C] in status 13:
<<;tag=#015#012>> .
Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:parse_to: unexpected end of header in state 13
Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:get_hdr_field: bad to header
Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]:
ERROR:tm:t_check: reply cannot be parsed

The impact I am getting is the homer UI is not showing complete sip
ladder.
--
regards,

abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445


On Thu, 9 Aug 2018 at 19:02, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hi Abdul,

have you checked the tutorial on HEP switching ?
https://blog.opensips.org/2017/10/12/opensips-as-hep-proxyswitch/

Maybe you are using the wrong routes in the script.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
   http://opensips.org/training/OpenSIPS_Bootcamp_2018/

On 08/09/2018 04:40 PM, Abdul Basit wrote:

Hi Alain,

Thank you for your suggestion.
I will update cfg and get back with results.

Do i need to upgrade opensips to 2.4 as well?

--
regards,

abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445


On Thu, 9 Aug 2018 at 17:35, Alain Bieuzent
mailto:alain.bieuz...@free.fr>> wrote:

Hi Abdul,

try to add this part of code at the begining of your
routing code to add more logs :

if(!sipmsg_validate(""))

    {

xlog("L_WARN", "Dropping mal formed Messages Retcode :
$retcode ");

[OpenSIPS-Users] tls_mgm

2018-08-31 Thread volga629 

Hello Everyone,
Recent versions of opensips tls_mgm module give this error.

opensips-3.0.0.b33b7a7e7-2.fc27.x86_64

[root@vprx00 ~]# opensips -V
version: opensips 3.0.0-dev (x86_64/linux)
flags: STATS: On, SHM_EXTRA_STATS, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, 
PKG_MALLOC, QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
MAX_URI_SIZE 1024, BUF_SIZE 65535

poll method support: poll, epoll, sigio_rt, select.
git revision: b33b7a7e7
main.c compiled on 12:56:54 Aug 14 2018 with gcc 7


Aug 31 09:08:26 [14440] DBG:core:load_module: loading module 
/usr/lib64/opensips/modules/proto_hep.so
Aug 31 09:08:26 [14440] DBG:core:load_module: loading module 
/usr/lib64/opensips/modules/proto_tls.so
Aug 31 09:08:26 [14440] DBG:core:add_module_dependency: adding type 2 
dependency proto_tls - (module tls_mgm)
Aug 31 09:08:26 [14440] DBG:core:add_module_dependency: adding type 0 
dependency proto_tls - (module proto_hep)
Aug 31 09:08:26 [14440] DBG:core:load_module: loading module 
/usr/lib64/opensips/modules/tls_mgm.so

Aug 31 09:08:26 [14440] DBG:core:register_module: register_pv: tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: tls_mgm matches 
module tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: found 
 in module tls_mgm [/usr/lib64/opensips/modules/]
Aug 31 09:08:26 [14440] ERROR:tls_mgm:split_param_val: No TLS domain 
name
Aug 31 09:08:26 [14440] CRITICAL:core:yyerror: parse error in config 
file /etc/opensips/opensips.cfg, line 205, column 18-19: Parameter 
 not found in module  - can't set
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: tls_mgm matches 
module tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: found 
 in module tls_mgm [/usr/lib64/opensips/modules/]
Aug 31 09:08:26 [14440] ERROR:tls_mgm:split_param_val: No TLS domain 
name
Aug 31 09:08:26 [14440] CRITICAL:core:yyerror: parse error in config 
file /etc/opensips/opensips.cfg, line 206, column 18-19: Parameter 
 not found in module  - can't set
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: tls_mgm matches 
module tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: found 
 in module tls_mgm [/usr/lib64/opensips/modules/]
Aug 31 09:08:26 [14440] ERROR:tls_mgm:split_param_val: No TLS domain 
name
Aug 31 09:08:26 [14440] CRITICAL:core:yyerror: parse error in config 
file /etc/opensips/opensips.cfg, line 207, column 18-19: Parameter 
 not found in module  - can't set
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: tls_mgm matches 
module tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: found 
 in module tls_mgm [/usr/lib64/opensips/modules/]
Aug 31 09:08:26 [14440] ERROR:tls_mgm:split_param_val: No TLS domain 
name
Aug 31 09:08:26 [14440] CRITICAL:core:yyerror: parse error in config 
file /etc/opensips/opensips.cfg, line 208, column 18-19: Parameter 
 not found in module  - can't set
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: tls_mgm matches 
module tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: found 
 in module tls_mgm [/usr/lib64/opensips/modules/]
Aug 31 09:08:26 [14440] ERROR:tls_mgm:split_param_val: No TLS domain 
name
Aug 31 09:08:26 [14440] CRITICAL:core:yyerror: parse error in config 
file /etc/opensips/opensips.cfg, line 209, column 18-19: Parameter 
 not found in module  - can't set
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: tls_mgm matches 
module tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: found  
in module tls_mgm [/usr/lib64/opensips/modules/]
Aug 31 09:08:26 [14440] ERROR:tls_mgm:split_param_val: No TLS domain 
name
Aug 31 09:08:26 [14440] CRITICAL:core:yyerror: parse error in config 
file /etc/opensips/opensips.cfg, line 210, column 18-19: Parameter 
 not found in module  - can't set
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: tls_mgm matches 
module tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: found  in 
module tls_mgm [/usr/lib64/opensips/modules/]
Aug 31 09:08:26 [14440] ERROR:tls_mgm:split_param_val: No TLS domain 
name
Aug 31 09:08:26 [14440] CRITICAL:core:yyerror: parse error in config 
file /etc/opensips/opensips.cfg, line 211, column 18-19: Parameter 
 not found in module  - can't set
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: tls_mgm matches 
module tls_mgm
Aug 31 09:08:26 [14440] DBG:core:set_mod_param_regex: found 
 in module tls_mgm [/usr/lib64/opensips/modules/]
Aug 31 09:08:26 [14440] ERROR:tls_mgm:split_param_val: No TLS domain 
name
Aug 31 09:08:26 [14440] CRITICAL:core:yyerror: parse error in config 
file /etc/opensips/opensips.cfg, line 212, column 18-19: Parameter 
 not found in module  - can't set




Any help thank you

volga629


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Re: [OpenSIPS-Users] Doubt about call center module

2018-08-31 Thread Daniel Zanutti 
You are correct, sorry.

I'll fix and start testing again.

Thanks

On Fri, Aug 31, 2018 at 10:10 AM Bogdan-Andrei Iancu 
wrote:

> As I said, in the cc_flows, you have no value for the "message_queue"
> column - this is a must, it has to be an URL to provide playback for the
> call queuing.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
> OpenSIPS Bootcamp 2018
>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>
> On 08/31/2018 04:06 PM, Daniel Zanutti wrote:
>
> Hi Bogdan
>
> Here it is table cc_flows:
> id  flowid  priority  skillprependcid  message_welcome
> message_queue
> --  --    ---  --  ---
> ---
>  1  fila-1   256  suporte  fila-1
>
>
> Also table agents:
> id  agentid location logstate
> skills   last_call_end
> --  --  ---  
> ---  ---
>  1  1...@plat5.domain.com  sip:1...@plat5.domain.com:5060 1
> suporte   1535650312
>
> Thanks
>
> On Fri, Aug 31, 2018 at 5:02 AM Bogdan-Andrei Iancu 
> wrote:
>
>> Hi Daniel,
>>
>> It is not about the B2B scenario, but about how you provisioned the flow
>> in DB. Could you simply dump the output of "select * from cc_flows" ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   http://www.opensips-solutions.com
>> OpenSIPS Bootcamp 2018
>>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>>
>> On 08/30/2018 08:34 PM, Daniel Zanutti wrote:
>>
>> Hi Bogdan
>>
>> Yes, It's the same scenario and same message. The call flow is:
>>
>> Asterisk Dials(port 5070) -> Opensips (port 5060) forward to Queue ->
>> Calls local user
>>
>> I'm using standard Queue scenario:
>> 
>> 
>> 
>> 
>> 
>> server1
>> 
>> 
>> client1
>> message
>> 
>> 1
>> 
>> 
>> 
>> 1
>> 
>> 
>>
>> And SIP message is the same on all calls, just changed Call-id/tags:
>>
>> U 10.10.10.10:5070 -> 10.10.10.10:5060
>> INVITE sip:fila-1@10.10.10.10:5060 SIP/2.0.
>> Via: SIP/2.0/UDP 10.10.10.10:5070;branch=z9hG4bK2abb2acc;rport.
>> Max-Forwards: 70.
>> From: ;tag=as6440e239.
>> To: .
>> Contact: .
>> Call-ID: 357cf76348e4e68325d065e85282320a@10.10.10.10:5070.
>> CSeq: 102 INVITE.
>> User-Agent: PBX SIPTEK.
>> Date: Thu, 30 Aug 2018 17:30:30 GMT.
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
>> PUBLISH, MESSAGE.
>> Supported: replaces, timer.
>> P-Asserted-Identity: "5511" .
>> Content-Type: application/sdp.
>> Content-Length: 353.
>> [SDP OMMITED]
>>
>> I updated to latest 2.4.2 GIT version (commit
>> 8b6830cdd96298682fcc298095ad1b718c54c77d), same problem is happening.
>>
>> Also you can access the server if you want, it's dedicated to this test.
>>
>> Thanks
>>
>>
>>
>>
>> On Thu, Aug 30, 2018 at 1:04 PM Bogdan-Andrei Iancu 
>> wrote:
>>
>>> Hi Daniel,
>>>
>>> Are you sure you configured a proper SIP URI as "message_queue" in the
>>> flow description ? My impression is you have an empty string there - and
>>> OpenSIPS is trying to put the call on the queue (as there is no agent), but
>>> the SIP URI is not valid.
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>>
>>> OpenSIPS Founder and Developer
>>>   http://www.opensips-solutions.com
>>> OpenSIPS Bootcamp 2018
>>>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>>>
>>> On 08/29/2018 10:26 PM, Daniel Zanutti wrote:
>>>
>>> Got some more info.
>>>
>>> *This is the first call that worked fine:*
>>> ..
>>>
>>> *This is the second call that had the problem:*
>>> .
>>> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
>>> DBG:call_center:cc_call_state_machine: selecting QUEUE
>>> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
>>> DBG:call_center:cc_queue_push_call:  QUEUE - adding call 0x7fd8510524a8
>>> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
>>> DBG:call_center:cc_queue_push_call: adding call on pos 0 (already 1 calls),
>>> l=(nil) h=(nil)
>>> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
>>> DBG:call_center:w_handle_call: new destination for call(0x7fd8510524a8) is
>>> (state=2)
>>> .
>>>
>>>
>>> On Mon, Aug 27, 2018 at 6:15 PM Daniel Zanutti 
>>> wrote:
>>>
 Trying to configure the call center modules, but found a problem when
 there is no agents available.

 If there is 1 agent available, call is sent to him with no problem:

 Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Recebida asterisk -
 Tentando entrar na fila fila-1
 Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Entrou na fila com sucesso
 (fila-1)!

Re: [OpenSIPS-Users] Doubt about call center module

2018-08-31 Thread Bogdan-Andrei Iancu 
As I said, in the cc_flows, you have no value for the "message_queue" 
column - this is a must, it has to be an URL to provide playback for the 
call queuing.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
  http://opensips.org/training/OpenSIPS_Bootcamp_2018/

On 08/31/2018 04:06 PM, Daniel Zanutti wrote:

Hi Bogdan

Here it is table cc_flows:
id  flowid  priority  skillprependcid message_welcome  
message_queue
--  --    ---  -- ---  
---

 1  fila-1   256  suporte  fila-1

Also table agents:
id  agentid locationlogstate  skills  
 last_call_end
--  -- ---  
  --- ---
 1 1...@plat5.domain.com  
sip:1...@plat5.domain.com:5060  
   1  suporte   1535650312


Thanks

On Fri, Aug 31, 2018 at 5:02 AM Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


Hi Daniel,

It is not about the B2B scenario, but about how you provisioned
the flow in DB. Could you simply dump the output of "select * from
cc_flows" ?

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
   http://opensips.org/training/OpenSIPS_Bootcamp_2018/

On 08/30/2018 08:34 PM, Daniel Zanutti wrote:

Hi Bogdan

Yes, It's the same scenario and same message. The call flow is:

Asterisk Dials(port 5070) -> Opensips (port 5060) forward to
Queue -> Calls local user

I'm using standard Queue scenario:





server1


client1
message

1



1



And SIP message is the same on all calls, just changed Call-id/tags:

U 10.10.10.10:5070  -> 10.10.10.10:5060

INVITE sip:fila-1@10.10.10.10:5060
 SIP/2.0.
Via: SIP/2.0/UDP 10.10.10.10:5070;branch=z9hG4bK2abb2acc;rport.
Max-Forwards: 70.
From: http://sip:5511@10.10.10.10:5070>>;tag=as6440e239.
To: http://sip:fila-1@10.10.10.10:5060>>.
Contact: http://sip:5511@10.10.10.10:5070>>.
Call-ID: 357cf76348e4e68325d065e85282320a@10.10.10.10:5070
.
CSeq: 102 INVITE.
User-Agent: PBX SIPTEK.
Date: Thu, 30 Aug 2018 17:30:30 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
P-Asserted-Identity: "5511" mailto:sip%3A5511@10.10.10.10>>.
Content-Type: application/sdp.
Content-Length: 353.
[SDP OMMITED]

I updated to latest 2.4.2 GIT version (commit
8b6830cdd96298682fcc298095ad1b718c54c77d), same problem is happening.

Also you can access the server if you want, it's dedicated to
this test.

Thanks




On Thu, Aug 30, 2018 at 1:04 PM Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hi Daniel,

Are you sure you configured a proper SIP URI as
"message_queue" in the flow description ? My impression is
you have an empty string there - and OpenSIPS is trying to
put the call on the queue (as there is no agent), but the SIP
URI is not valid.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
   http://opensips.org/training/OpenSIPS_Bootcamp_2018/

On 08/29/2018 10:26 PM, Daniel Zanutti wrote:

Got some more info.

*This is the first call that worked fine:*
..

*This is the second call that had the problem:*
.
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:call_center:cc_call_state_machine: selecting QUEUE
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:call_center:cc_queue_push_call:  QUEUE - adding call
0x7fd8510524a8
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:call_center:cc_queue_push_call: adding call on pos 0
(already 1 calls), l=(nil) h=(nil)
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:call_center:w_handle_call: new destination for
call(0x7fd8510524a8) is (state=2)
.


On Mon, Aug 27, 2018 at 6:15 PM Daniel Zanutti
mailto:daniel.zanu...@gmail.com>>
wrote:

Trying to configure the call center modules, but found a
problem when there is no agents available.

Re: [OpenSIPS-Users] Doubt about call center module

2018-08-31 Thread Daniel Zanutti 
Hi Bogdan

Here it is table cc_flows:
id  flowid  priority  skillprependcid  message_welcome
message_queue
--  --    ---  --  ---
---
 1  fila-1   256  suporte  fila-1


Also table agents:
id  agentid location logstate
skills   last_call_end
--  --  ---  
---  ---
 1  1...@plat5.domain.com  sip:1...@plat5.domain.com:5060 1
suporte   1535650312

Thanks

On Fri, Aug 31, 2018 at 5:02 AM Bogdan-Andrei Iancu 
wrote:

> Hi Daniel,
>
> It is not about the B2B scenario, but about how you provisioned the flow
> in DB. Could you simply dump the output of "select * from cc_flows" ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
> OpenSIPS Bootcamp 2018
>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>
> On 08/30/2018 08:34 PM, Daniel Zanutti wrote:
>
> Hi Bogdan
>
> Yes, It's the same scenario and same message. The call flow is:
>
> Asterisk Dials(port 5070) -> Opensips (port 5060) forward to Queue ->
> Calls local user
>
> I'm using standard Queue scenario:
> 
> 
> 
> 
> 
> server1
> 
> 
> client1
> message
> 
> 1
> 
> 
> 
> 1
> 
> 
>
> And SIP message is the same on all calls, just changed Call-id/tags:
>
> U 10.10.10.10:5070 -> 10.10.10.10:5060
> INVITE sip:fila-1@10.10.10.10:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.10.10.10:5070;branch=z9hG4bK2abb2acc;rport.
> Max-Forwards: 70.
> From: ;tag=as6440e239.
> To: .
> Contact: .
> Call-ID: 357cf76348e4e68325d065e85282320a@10.10.10.10:5070.
> CSeq: 102 INVITE.
> User-Agent: PBX SIPTEK.
> Date: Thu, 30 Aug 2018 17:30:30 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE.
> Supported: replaces, timer.
> P-Asserted-Identity: "5511" .
> Content-Type: application/sdp.
> Content-Length: 353.
> [SDP OMMITED]
>
> I updated to latest 2.4.2 GIT version (commit
> 8b6830cdd96298682fcc298095ad1b718c54c77d), same problem is happening.
>
> Also you can access the server if you want, it's dedicated to this test.
>
> Thanks
>
>
>
>
> On Thu, Aug 30, 2018 at 1:04 PM Bogdan-Andrei Iancu 
> wrote:
>
>> Hi Daniel,
>>
>> Are you sure you configured a proper SIP URI as "message_queue" in the
>> flow description ? My impression is you have an empty string there - and
>> OpenSIPS is trying to put the call on the queue (as there is no agent), but
>> the SIP URI is not valid.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   http://www.opensips-solutions.com
>> OpenSIPS Bootcamp 2018
>>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>>
>> On 08/29/2018 10:26 PM, Daniel Zanutti wrote:
>>
>> Got some more info.
>>
>> *This is the first call that worked fine:*
>> ..
>>
>> *This is the second call that had the problem:*
>> .
>> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
>> DBG:call_center:cc_call_state_machine: selecting QUEUE
>> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
>> DBG:call_center:cc_queue_push_call:  QUEUE - adding call 0x7fd8510524a8
>> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
>> DBG:call_center:cc_queue_push_call: adding call on pos 0 (already 1 calls),
>> l=(nil) h=(nil)
>> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
>> DBG:call_center:w_handle_call: new destination for call(0x7fd8510524a8) is
>> (state=2)
>> .
>>
>>
>> On Mon, Aug 27, 2018 at 6:15 PM Daniel Zanutti 
>> wrote:
>>
>>> Trying to configure the call center modules, but found a problem when
>>> there is no agents available.
>>>
>>> If there is 1 agent available, call is sent to him with no problem:
>>>
>>> Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Recebida asterisk -
>>> Tentando entrar na fila fila-1
>>> Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Entrou na fila com sucesso
>>> (fila-1)!
>>> Aug 27 18:11:01 plat5 /sbin/opensips[23569]: incoming reply
>>>
>>> But when there is no agent available, opensips refuses:
>>> Aug 27 18:11:07 plat5 /sbin/opensips[23569]: Recebida asterisk -
>>> Tentando entrar na fila fila-1
>>> Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
>>> ERROR:b2b_logic:b2b_process_scenario_init: Failed to get the value for the
>>> b2b client ruri
>>> Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
>>> ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID
>>> received)
>>> Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
>>> ERROR:call_center:w_handle_call: failed to set new destination for call
>>> Aug 27 18:11:07 plat5 /sbin/opensips[23569]: errnum: -1
>>>

Re: [OpenSIPS-Users] dialog replication

2018-08-31 Thread Slava Bendersky 

Hello Liviu,
Please check this diagram 
https://www.dropbox.com/s/od0z7wacc2yu3rr/opensips-cluster-design.pdf?dl=0
The error coming up on public inteface where no vips just round robin 
DNS A recoord.


volga629

On Thu, Aug 16, 2018 at 5:20 AM, Liviu Chircu  
wrote:

Hi Volga,

The 2.4 dialog clustering definitely supports "active-active" setups. 
 For this, you will need to use two virtual IPs, each being primary 
for one of the nodes, such that you avoid those "dialog created on 
unknown IP" errors when replicating dialogs to the backup node.


Best regards,

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

On 16.08.2018 08:02, volga...@networklab.ca wrote:

Hello Everyone,
Based on this thread 
https://opensips.org/pipermail/users/2014-August/029635.html dialog 
replication is not design for active/active scanario. My question 
when federated no sql cluster enabled is it really need it ?


tried enable and getting errors

Aug 15 23:42:22 aitossbc01 /usr/sbin/opensips[12909]: 
WARNING:dialog:fetch_socket_info: non-local socket public ip:5060>...ignoring
Aug 15 23:42:22 aitossbc01 /usr/sbin/opensips[12909]: 
ERROR:dialog:dlg_replicated_create: Replicated dialog doesn't match 
any listening sockets
Aug 15 23:42:22 aitossbc01 /usr/sbin/opensips[12909]: 
ERROR:dialog:receive_dlg_repl: Failed to process a binary packet!


 Dialog
loadmodule "dialog.so"
modparam("dialog", "db_url", "postgres://")
modparam("dialog", "db_mode", 1)
modparam("dialog","profiles_with_value","outbound; inbound")
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "default_timeout", 3600)
modparam("dialog", "options_ping_interval", 900)
modparam("dialog", "profiles_with_value", "caller ; domain")
modparam("dialog", "dialog_replication_cluster", 1)
modparam("dialog", "profile_replication_cluster", 1)
modparam("dialog", "dlg_sharing_tag", "vip1=active")


volga629


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[OpenSIPS-Users] Passing AVP to prefix core function

2018-08-31 Thread Alex A 
Dear All, I am working what appears to be a simple function for opensips 2.2.3, 
however cannot seem to get it working.. Essentially, extract the groupID from 
permissions module and add a prefix to R-URI on the egress side. 
https://www.opensips.org/Documentation/Script-CoreFunctions-2-2#toc26 
http://www.opensips.org/html/docs/modules/2.2.x/permissions.html#idp5689232 
Config route looks like this:     route[relay] {     if ( 
get_source_group("$avp(group)") ) {     # do something with $avp(group) 
    xlog("group is $avp(group)\n");                          
#prefix("$avp(group)");     };     #Add the string parameter in front of 
username in R-URI.     #prefix("$avp(group)");     
#prefix("$avp(group){s.substr,0,0}");     $avp(22) = "#";     
prefix("$avp(22)"); Prefix core function prefixes R-URI with variable name 
($avp(22)) instead of value of "#". I have tried various syntax versions 
that are commented out, however to no avail.. If I remove the quotes around the 
variable name: prefix($avp(22)); Opensips does not startup at all, complaining 
about: syntax error and bad argument, string expected Am I missing something 
simple? or prefix function is simply not designed to work with variables? Thank 
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Re: [OpenSIPS-Users] OpenSIPS Unable to parse msg received from

2018-08-31 Thread Abdul Basit 
Hi Alain Bieuzent,

I added sip msg validate code in start of main route but i didn't get these
additional log lines.
Still getting messages from opensips core:

Aug 23 13:22:24 homer-proxy /usr/sbin/opensips[25568]:
ERROR:core:receive_msg: Unable to parse msg received from [x.x.x.x:61648]
Aug 23 13:22:27 homer-proxy /usr/sbin/opensips[25568]:
ERROR:core:parse_msg:
message=;tag=1c1737546331#015#012To:
#015#012Call-ID: c@x.x.x.x#015#012CSeq: 63697
REGISTER#015#012Authorization: Digest username="yy",realm="x.x.x.x"
,nonce="5b7e9911db6acf694065c578ff32fa51ce24bb374d67",uri="sip:x.x.x.x"
,algorithm=MD5,response="debec3f66d2ac1ec488138871ffa1d60"#015#012Contact:
;expires=360#015#012Supported:
em,timer,replaces,path,resource-priority#015#012Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE#015#012Expires:
360#015#012User-Agent: Audiocodes-Sip-Gateway-MP-118
FXS/v.5.20A.043.005#015#012Content-Length: 0#015#012#015#012>

Hi Bogdan,

Yes, I did it as per cfg in shared link. But again i am getting the parse
error messages. Looks like request is initiating from some buggy sip client.

Aug 23 13:43:39 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:parse_first_line: bad request first line
Aug 23 13:43:39 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:parse_first_line: at line 0 char 34:
Aug 23 13:43:39 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:parse_first_line: parsed so far: AudiocodesGW 1372031417
1372031297
Aug 23 13:43:39 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:parse_msg: message=
Aug 23 13:43:39 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:receive_msg: Unable to parse msg received from [x.x.x.x:64443]
Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:parse_to_param: unexpected char [C] in status 13: <<;tag=#015#012
>> .
Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]: ERROR:core:parse_to:
unexpected end of header in state 13
Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:get_hdr_field: bad to header
Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:parse_to_param: unexpected char [C] in status 13: <<;tag=#015#012
>> .
Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]: ERROR:core:parse_to:
unexpected end of header in state 13
Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]:
ERROR:core:get_hdr_field: bad to header
Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]: ERROR:tm:t_check:
reply cannot be parsed

The impact I am getting is the homer UI is not showing complete sip ladder.

--
regards,

abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445


On Thu, 9 Aug 2018 at 19:02, Bogdan-Andrei Iancu 
wrote:

> Hi Abdul,
>
> have you checked the tutorial on HEP switching ?
> https://blog.opensips.org/2017/10/12/opensips-as-hep-proxyswitch/
>
> Maybe you are using the wrong routes in the script.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
> OpenSIPS Bootcamp 2018
>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>
> On 08/09/2018 04:40 PM, Abdul Basit wrote:
>
> Hi Alain,
>
> Thank you for your suggestion.
> I will update cfg and get back with results.
>
> Do i need to upgrade opensips to 2.4 as well?
>
>
> --
> regards,
>
> abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445
>
>
> On Thu, 9 Aug 2018 at 17:35, Alain Bieuzent 
> wrote:
>
>> Hi Abdul,
>>
>>
>>
>> try to add this part of code at the begining of your routing code to add
>> more logs :
>>
>>
>>
>> if(!sipmsg_validate(""))
>>
>> {
>>
>> xlog("L_WARN", "Dropping mal formed Messages Retcode : $retcode
>> ");
>>
>> xlog("L_WARN","--- error from [$si:$sp]\n+\n$mb\n\n");
>>
>> exit;
>>
>> }
>>
>>
>>
>> then you can find the retcode value there :
>> http://www.opensips.org/html/docs/modules/2.4.x/sipmsgops.html#func_sipmsg_validate
>>
>>
>>
>>
>>
>> *De : *Users  au nom de Abdul Basit <
>> basit.e...@gmail.com>
>> *Répondre à : *OpenSIPS users mailling list 
>> *Date : *jeudi 9 août 2018 à 14:18
>> *À : *OpenSIPS users mailling list 
>> *Objet : *[OpenSIPS-Users] OpenSIPS Unable to parse msg received from
>>
>>
>>
>> Hi Team,
>>
>>
>>
>> I am using opensips as hep proxy to distribute traffic to multiple homer
>> noes for trace capturing.
>>
>>
>>
>> # opensips -V
>>
>> version: opensips 2.3.2 (x86_64/linux)
>>
>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
>> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>
>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>>
>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>
>> main.c compiled on  with gcc 6.3.0
>>
>>
>>
>> Host operating system is
>>
>>
>>
>> # lsb_release -a
>>
>> No LSB modules are available.
>>
>> Distributor ID: Debian
>>
>> Description:Debian GNU/Linux 9.3 (stretch)
>>
>> Release:9.3
>>
>> Codename:   stretch
>>
>>
>>
>> I am

Re: [OpenSIPS-Users] OpenSIPS Unable to parse msg received from

2018-08-31 Thread Abdul Basit 
I upgraded home proxy server opensips just in case it may help resolving
the issues.

# opensips -V
version: *opensips 2.4.2* (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC,
FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.
main.c compiled on  with gcc 6.3.0

But result is same. I am continuously getting parse_mgs errors.
any idea about fixation of the issue?

--
regards,

abdul basit


On Thu, 23 Aug 2018 at 16:52, Abdul Basit  wrote:

> Hi Alain Bieuzent,
>
> I added sip msg validate code in start of main route but i didn't get
> these additional log lines.
> Still getting messages from opensips core:
>
> Aug 23 13:22:24 homer-proxy /usr/sbin/opensips[25568]:
> ERROR:core:receive_msg: Unable to parse msg received from [x.x.x.x:61648]
> Aug 23 13:22:27 homer-proxy /usr/sbin/opensips[25568]:
> ERROR:core:parse_msg: 
> message= 70#015#012From: ;tag=1c1737546331#015#012To:
> #015#012Call-ID: c@x.x.x.x#015#012CSeq: 63697
> REGISTER#015#012Authorization: Digest username="yy",realm="x.x.x.x"
> ,nonce="5b7e9911db6acf694065c578ff32fa51ce24bb374d67",uri=
> "sip:x.x.x.x",algorithm=MD5,response="debec3f66d2ac1ec488138871ffa1d60"#015#012Contact:
> ;expires=360#015#012Supported:
> em,timer,replaces,path,resource-priority#015#012Allow:
> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE#015#012Expires:
> 360#015#012User-Agent: Audiocodes-Sip-Gateway-MP-118
> FXS/v.5.20A.043.005#015#012Content-Length: 0#015#012#015#012>
>
> Hi Bogdan,
>
> Yes, I did it as per cfg in shared link. But again i am getting the parse
> error messages. Looks like request is initiating from some buggy sip client.
>
> Aug 23 13:43:39 homer-proxy /usr/sbin/opensips[26027]:
> ERROR:core:parse_first_line: bad request first line
> Aug 23 13:43:39 homer-proxy /usr/sbin/opensips[26027]:
> ERROR:core:parse_first_line: at line 0 char 34:
> Aug 23 13:43:39 homer-proxy /usr/sbin/opensips[26027]:
> ERROR:core:parse_first_line: parsed so far: AudiocodesGW 1372031417
> 1372031297
> Aug 23 13:43:39 homer-proxy /usr/sbin/opensips[26027]:
> ERROR:core:parse_msg: message=  x.x.x.x#015#012s=Phone-Call#015#012c=IN IP4 x.x.x.x#015#012t=0 0#015#012m
> =audio 6070 RTP/AVP 8 96#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:96
> telephone-event/8000#015#012a=fmtp:96 0-15#015#012a=ptime:20#015#012a=
> sendrecv#015#012a=rtcp:6071 IN IP4 x.x.x.x#015#012>
> Aug 23 13:43:39 homer-proxy /usr/sbin/opensips[26027]:
> ERROR:core:receive_msg: Unable to parse msg received from [x.x.x.x:64443]
> Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]:
> ERROR:core:parse_to_param: unexpected char [C] in status 13: <<;tag=
> #015#012>> .
> Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]:
> ERROR:core:parse_to: unexpected end of header in state 13
> Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]:
> ERROR:core:get_hdr_field: bad to header
> Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]:
> ERROR:core:parse_to_param: unexpected char [C] in status 13: <<;tag=
> #015#012>> .
> Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]:
> ERROR:core:parse_to: unexpected end of header in state 13
> Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]:
> ERROR:core:get_hdr_field: bad to header
> Aug 23 13:43:40 homer-proxy /usr/sbin/opensips[26027]: ERROR:tm:t_check:
> reply cannot be parsed
>
> The impact I am getting is the homer UI is not showing complete sip ladder.
>
> --
> regards,
>
> abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445
>
>
> On Thu, 9 Aug 2018 at 19:02, Bogdan-Andrei Iancu 
> wrote:
>
>> Hi Abdul,
>>
>> have you checked the tutorial on HEP switching ?
>> https://blog.opensips.org/2017/10/12/opensips-as-hep-proxyswitch/
>>
>> Maybe you are using the wrong routes in the script.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   http://www.opensips-solutions.com
>> OpenSIPS Bootcamp 2018
>>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>>
>> On 08/09/2018 04:40 PM, Abdul Basit wrote:
>>
>> Hi Alain,
>>
>> Thank you for your suggestion.
>> I will update cfg and get back with results.
>>
>> Do i need to upgrade opensips to 2.4 as well?
>>
>>
>> --
>> regards,
>>
>> abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445
>>
>>
>> On Thu, 9 Aug 2018 at 17:35, Alain Bieuzent 
>> wrote:
>>
>>> Hi Abdul,
>>>
>>>
>>>
>>> try to add this part of code at the begining of your routing code to add
>>> more logs :
>>>
>>>
>>>
>>> if(!sipmsg_validate(""))
>>>
>>> {
>>>
>>> xlog("L_WARN", "Dropping mal formed Messages Retcode : $retcode
>>> ");
>>>
>>> xlog("L_WARN","--- error from [$si:$sp]\n+\n$mb\n\n");
>>>
>>> exit;
>>>
>>> }
>>>
>>>
>>>
>>> then you can find the retcode value there :
>>> http://www.opensips.org/html/docs/modules/2.4.x/sipmsgops.html#func_sipmsg_validate

[OpenSIPS-Users] OpenSIPS Bootcamp - Early Birds closing

2018-08-31 Thread OpenSIPS Team 

*
   22nd-26th of October 2018,
   Cluj-Napoca, Romania
**DoubleTree by Hilton Hotel**
*



*Early Birds get to an end*

The /*1st of September*/ deadline for the Early Bird 10% discount is 
almost there, so do not mis the opportunity and secure your seat for a 
good price.


Register now 



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**

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platforms to enhance their businesses:


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Re: [OpenSIPS-Users] Ordinary presence server functions of OpenSIPS

2018-08-31 Thread Ben Newlin 
The free Bria client is capable of being a presence user agent. I believe 
Zoiper will do so as well.

The error response you are getting is 489 Bad Event. Looking at the Event 
header in the message you are sending from SIPp you have set 
“E_PRESENCE_PUBLISH”. That is not a defined event package for SIP; that is the 
name of the internal OpenSIPS event corresponding to presence publishing. Per 
RFC 3856 the event package for presence is “presence”.

Ben Newlin

From: Users  on behalf of Giovanni Maruzzelli 

Reply-To: "gmar...@opentelecom.it" , OpenSIPS users 
mailling list 
Date: Thursday, August 2, 2018 at 1:02 PM
To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] Ordinary presence server functions of OpenSIPS

study, Fatma, study :)

btw, if you can't "find a softphone capable of being a presence user agent" you 
may be in the wrong field of studies.

-giovanni

On 2 August 2018 at 15:52, Fatma Raissi 
mailto:raissifa...@gmail.com>> wrote:
Good morning,


Thanks again for your answer.
But I can't find a softphone capable of being a presence user agent.
Plus the presence information I need to publish is one variable which is 
"workload" of the machine.

Here is the SIP message I am using and joined the configuration file. Maybe you 
can Identify the problem. Thanks

test.xml







  
  
  
  
  
  

  

  


Reponse: 489 Bad event



Aug  2 06:49:44 [40701] DBG:core:get_hdr_field: cseq : <1> 
Aug  2 06:49:44 [40701] DBG:maxfwd:is_maxfwd_present: value = 70
Aug  2 06:49:44 [40701] DBG:uri:has_totag: no totag
Aug  2 06:49:44 [40701] DBG:core:parse_headers: flags=78
Aug  2 06:49:44 [40701] DBG:tm:t_lookup_request: start searching: hash=22792, 
isACK=0
Aug  2 06:49:44 [40701] DBG:tm:matching_3261: RFC3261 transaction matched, 
tid=nashds7
Aug  2 06:49:44 [40701] DBG:tm:t_lookup_request: REF_UNSAFE:[0x7f73f6c49708] 
after is 1
Aug  2 06:49:44 [40701] DBG:tm:t_lookup_request: transaction found 
(T=0x7f73f6c49708)
Aug  2 06:49:44 [40701] DBG:tm:t_retransmit_reply: buf=0x7f73f644f600: SIP/2.0 
4..., shmem=0x7f73f6c4c678: SIP/2.0 4
Aug  2 06:49:44 [40701] DBG:tm:t_check_trans: UNREF_UNSAFE: [0x7f73f6c49708] 
after is 0
Aug  2 06:49:44 [40701] DBG:core:destroy_avp_list: destroying list (nil)
Aug  2 06:49:44 [40701] DBG:core:receive_msg: cleaning up
Aug  2 06:49:44 [40700] DBG:core:parse_msg: SIP Request:
Aug  2 06:49:44 [40700] DBG:core:parse_msg:  method:  
Aug  2 06:49:44 [40700] DBG:core:parse_msg:  uri: 
http://127.0.0.1:5060>>
Aug  2 06:49:44 [40700] DBG:core:parse_msg:  version: 
Aug  2 06:49:44 [40700] DBG:core:parse_headers: flags=2
Aug  2 06:49:44 [40700] DBG:core:parse_via_param: found param type 235,  
= ; state=6
Aug  2 06:49:44 [40700] DBG:core:parse_via_param: found param type 232, 
 = ; state=16
Aug  2 06:49:44 [40700] DBG:core:parse_via: end of header reached, state=5
Aug  2 06:49:44 [40700] DBG:core:parse_headers: via found, flags=2
Aug  2 06:49:44 [40700] DBG:core:parse_headers: this is the first via
Aug  2 06:49:44 [40700] DBG:core:receive_msg: After parse_msg...
Aug  2 06:49:44 [40700] DBG:core:receive_msg: preparing to run routing 
scripts...
Aug  2 06:49:44 [40700] DBG:core:parse_headers: flags=100
Aug  2 06:49:44 [40700] DBG:core:_parse_to: end of header reached, state=9
Aug  2 06:49:44 [40700] DBG:core:_parse_to: display={}, 
ruri={sip:127.0.0.1:5060}
Aug  2 06:49:44 [40700] DBG:core:get_hdr_field:  [20]; 
uri=[sip:127.0.0.1:5060]
Aug  2 06:49:44 [40700] DBG:core:get_hdr_field: to body 
[sip:127.0.0.1:5060

Error! Filename not specified.



Error! Filename not specified.

Fatma RAISSI  - ENIT Junior Entreprise

Élève ingénieur en télécommunication
Membre d'honneur
Vice-Présidente du mandat 2016-2017
 Tel: (+216) 53 411 311 | Email: 
raissifa...@gmail.com



2018-08-02 13:48 GMT+02:00 Giovanni Maruzzelli 
mailto:gmar...@gmail.com>>:
Be ause they have working presence client embedded, and you seems not be able 
to model it in sipp.

Start with something known to work, softphones, trace the sip messages, then 
(if needed) do the sipp xml modelization.

-giovanni

On Thu, Aug 2, 2018, 13:45 Fatma Raissi 
mailto:raissifa...@gmail.com>> wrote:
Good morning Sir,


Thank you a lot for your answer.
But could you explain why would I use softphones while I have nothing to do 
with voice or voice over IP.

Cordially,



Error! Filename not specified.

Fatma RAISSI  - ENIT Junior Entreprise

Élève ingénieur en télécommunication
Membre d'honneur
Vice-Présidente du mandat 2016-2017
 Tel: (+216) 53 411 311 | Email: 
raissifa...@gmail.com



2018-08-02 10:34 GMT+02:00 Giovanni Maruzzelli 
mailto:gmar...@gmail.com>>:
Use softphones instead of sipp

On Wed, Aug 1, 2018, 12:01 Fatma Raissi 
mailto:raissifa...@gmail.com>> wrote:
Good morning Everyone,


I am using OpenSIPS as presence server. I need it just to accomplish very basic 
and simple presence server

Re: [OpenSIPS-Users] Doubt about call center module

2018-08-31 Thread Bogdan-Andrei Iancu 

Hi Daniel,

It is not about the B2B scenario, but about how you provisioned the flow 
in DB. Could you simply dump the output of "select * from cc_flows" ?


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
  http://opensips.org/training/OpenSIPS_Bootcamp_2018/

On 08/30/2018 08:34 PM, Daniel Zanutti wrote:

Hi Bogdan

Yes, It's the same scenario and same message. The call flow is:

Asterisk Dials(port 5070) -> Opensips (port 5060) forward to Queue -> 
Calls local user


I'm using standard Queue scenario:





server1


client1
message

1



1



And SIP message is the same on all calls, just changed Call-id/tags:

U 10.10.10.10:5070  -> 10.10.10.10:5060 

INVITE sip:fila-1@10.10.10.10:5060 
 SIP/2.0.

Via: SIP/2.0/UDP 10.10.10.10:5070;branch=z9hG4bK2abb2acc;rport.
Max-Forwards: 70.
From: >;tag=as6440e239.

To: http://sip:fila-1@10.10.10.10:5060>>.
Contact: >.
Call-ID: 357cf76348e4e68325d065e85282320a@10.10.10.10:5070 
.

CSeq: 102 INVITE.
User-Agent: PBX SIPTEK.
Date: Thu, 30 Aug 2018 17:30:30 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE.

Supported: replaces, timer.
P-Asserted-Identity: "5511" >.

Content-Type: application/sdp.
Content-Length: 353.
[SDP OMMITED]

I updated to latest 2.4.2 GIT version (commit 
8b6830cdd96298682fcc298095ad1b718c54c77d), same problem is happening.


Also you can access the server if you want, it's dedicated to this test.

Thanks




On Thu, Aug 30, 2018 at 1:04 PM Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


Hi Daniel,

Are you sure you configured a proper SIP URI as "message_queue" in
the flow description ? My impression is you have an empty string
there - and OpenSIPS is trying to put the call on the queue (as
there is no agent), but the SIP URI is not valid.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
   http://opensips.org/training/OpenSIPS_Bootcamp_2018/

On 08/29/2018 10:26 PM, Daniel Zanutti wrote:

Got some more info.

*This is the first call that worked fine:*
..

*This is the second call that had the problem:*
.
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:call_center:cc_call_state_machine: selecting QUEUE
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:call_center:cc_queue_push_call:  QUEUE - adding call
0x7fd8510524a8
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:call_center:cc_queue_push_call: adding call on pos 0 (already
1 calls), l=(nil) h=(nil)
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:call_center:w_handle_call: new destination for
call(0x7fd8510524a8) is  (state=2)
.


On Mon, Aug 27, 2018 at 6:15 PM Daniel Zanutti
mailto:daniel.zanu...@gmail.com>> wrote:

Trying to configure the call center modules, but found a
problem when there is no agents available.

If there is 1 agent available, call is sent to him with no
problem:

Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Recebida
asterisk - Tentando entrar na fila fila-1
Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Entrou na fila
com sucesso (fila-1)!
Aug 27 18:11:01 plat5 /sbin/opensips[23569]: incoming reply

But when there is no agent available, opensips refuses:
Aug 27 18:11:07 plat5 /sbin/opensips[23569]: Recebida
asterisk - Tentando entrar na fila fila-1
Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
ERROR:b2b_logic:b2b_process_scenario_init: Failed to get the
value for the b2b client ruri
Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
ERROR:call_center:set_call_leg: failed to init new b2bua call
(empty ID received)
Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
ERROR:call_center:w_handle_call: failed to set new
destination for call
Aug 27 18:11:07 plat5 /sbin/opensips[23569]: errnum: -1

Error -1 means flowID is invalid, but I sent the same value
on both calls.

This is the call:

cc_handle_call("$rU")

I'm using Opensips 2.4.2 with Debian 8.11.

Am I missing something or found a bug?

Thanks



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