Re: [OpenSIPS-Users] SIP SDP rfc1918 address fix
I kind of knew it what you going say ;-)
Because of legacy changes we have to stick with it.
Sent from my iPhone
> On Apr 28, 2017, at 9:39 AM, Bogdan-Andrei Iancu <bog...@opensips.org> wrote:
>
> :-| SER 0.10 ?? that's 12 year oldpart of history... cannot help with
> that.
>
> Try OpenSIPS 2.2 or 2.3
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
> http://www.opensips.org/events/Summit-2017Amsterdam.html
>
>> On 04/28/2017 04:24 PM, Satish Patel wrote:
>> Unfortunately we are using SER 0.10
>>
>> Sent from my iPhone
>>
>>> On Apr 27, 2017, at 5:50 AM, Bogdan-Andrei Iancu <bog...@opensips.org>
>>> wrote:
>>>
>>> Satish,
>>>
>>> I do not fine the err log you mentioned ("extract_mediaip: no `c=' in SDP")
>>> in the code of OpenSIPS - what version are you using ??
>>>
>>> Also I tried to to inject your SDP into OpenSIPS 2.3 and I do not get the
>>> any errors.
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developer
>>> http://www.opensips-solutions.com
>>>
>>> OpenSIPS Summit May 2017 Amsterdam
>>> http://www.opensips.org/events/Summit-2017Amsterdam.html
>>>
>>>> On 04/27/2017 03:43 AM, Satish Patel wrote:
>>>> Yes, whenever fix_nated_sdp() fiction run it produce that error which I
>>>> mentioned in my previous email. Every single time.
>>>>
>>>> Sent from my iPhone
>>>>
>>>>> On Apr 26, 2017, at 4:52 PM, Bogdan-Andrei Iancu <bog...@opensips.org>
>>>>> wrote:
>>>>>
>>>>> So below is the SDP OpenSIPS receives (from network) and when doing
>>>>> fix_nated_sdp() on that SDP leads to the "c=" errors ?
>>>>>
>>>>> Regards,
>>>>>
>>>>> Bogdan-Andrei Iancu
>>>>> OpenSIPS Founder and Developer
>>>>> http://www.opensips-solutions.com
>>>>>
>>>>> OpenSIPS Summit May 2017 Amsterdam
>>>>> http://www.opensips.org/events/Summit-2017Amsterdam.html
>>>>>
>>>>>> On 04/26/2017 08:44 PM, Satish Patel wrote:
>>>>>> Here is my payload again we have custom application which is using SER
>>>>>> so some of them are custom values, This is the payload after i apply
>>>>>> fix_nated_sdp() function.
>>>>>>
>>>>>>
>>>>>> Max-Forwards: 16.
>>>>>> Content-Type: application/sdp.
>>>>>> Content-Length: 418.
>>>>>> Supported: path, 100rel.
>>>>>> P-hint: LOCAL.
>>>>>> P-hint: ALIASED OUTBOUND.
>>>>>> P-hint: DIRECT-RTP.
>>>>>> .
>>>>>> v=0.
>>>>>> o=user1 53655765 2353687637 IN IP4 192.168.1.8.
>>>>>> s=-.
>>>>>> c=IN IP4 173.71.121.4.
>>>>>> t=0 0.
>>>>>> m=audio 6000 RTP/AVP 0.
>>>>>> a=rtpmap:127 VANI/32000.
>>>>>> a=fmtp:127 ver=3;mode=3;sub-types=1,7;codecs=0x26.
>>>>>> a=rtpmap:111 SIREN14-3D/32000.
>>>>>> a=fmtp:111 bitrate=32000.
>>>>>> a=vx_payload_hdr_ver:2.
>>>>>> a=rtpmap:0 PCMU/8000.
>>>>>> a=vx_join_audio:1.
>>>>>> a=vx_join_text:0.
>>>>>> a=vx_jc:60.
>>>>>> a=setup:both.
>>>>>> a=vx_rtcp:0.
>>>>>> a=direction:active.
>>>>>> a=oldmediaip:192.168.1.8.
>>>>>>
>>>>>> On Wed, Apr 26, 2017 at 6:18 AM, Bogdan-Andrei Iancu
>>>>>> <bog...@opensips.org> wrote:
>>>>>>> Hi Satish,
>>>>>>>
>>>>>>> For the mime test, you can use the has_body() function:
>>>>>>> http://www.opensips.org/html/docs/modules/2.2.x/sipmsgops.html#idp3886992
>>>>>>>
>>>>>>> About the error - could you post the actual SDP payload generating those
>>>>>>> errors ?
>>>>>>>
>>>>>>> Regards,
>>>>>>>
>>>>>>> Bogdan-Andrei Iancu
>>>>>>> OpenSIPS Founder and Developer
>>>>>>> http://www.opensips-solutio
Re: [OpenSIPS-Users] SIP SDP rfc1918 address fix
Unfortunately we are using SER 0.10
Sent from my iPhone
> On Apr 27, 2017, at 5:50 AM, Bogdan-Andrei Iancu <bog...@opensips.org> wrote:
>
> Satish,
>
> I do not fine the err log you mentioned ("extract_mediaip: no `c=' in SDP")
> in the code of OpenSIPS - what version are you using ??
>
> Also I tried to to inject your SDP into OpenSIPS 2.3 and I do not get the any
> errors.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
> http://www.opensips.org/events/Summit-2017Amsterdam.html
>
>> On 04/27/2017 03:43 AM, Satish Patel wrote:
>> Yes, whenever fix_nated_sdp() fiction run it produce that error which I
>> mentioned in my previous email. Every single time.
>>
>> Sent from my iPhone
>>
>>> On Apr 26, 2017, at 4:52 PM, Bogdan-Andrei Iancu <bog...@opensips.org>
>>> wrote:
>>>
>>> So below is the SDP OpenSIPS receives (from network) and when doing
>>> fix_nated_sdp() on that SDP leads to the "c=" errors ?
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developer
>>> http://www.opensips-solutions.com
>>>
>>> OpenSIPS Summit May 2017 Amsterdam
>>> http://www.opensips.org/events/Summit-2017Amsterdam.html
>>>
>>>> On 04/26/2017 08:44 PM, Satish Patel wrote:
>>>> Here is my payload again we have custom application which is using SER
>>>> so some of them are custom values, This is the payload after i apply
>>>> fix_nated_sdp() function.
>>>>
>>>>
>>>> Max-Forwards: 16.
>>>> Content-Type: application/sdp.
>>>> Content-Length: 418.
>>>> Supported: path, 100rel.
>>>> P-hint: LOCAL.
>>>> P-hint: ALIASED OUTBOUND.
>>>> P-hint: DIRECT-RTP.
>>>> .
>>>> v=0.
>>>> o=user1 53655765 2353687637 IN IP4 192.168.1.8.
>>>> s=-.
>>>> c=IN IP4 173.71.121.4.
>>>> t=0 0.
>>>> m=audio 6000 RTP/AVP 0.
>>>> a=rtpmap:127 VANI/32000.
>>>> a=fmtp:127 ver=3;mode=3;sub-types=1,7;codecs=0x26.
>>>> a=rtpmap:111 SIREN14-3D/32000.
>>>> a=fmtp:111 bitrate=32000.
>>>> a=vx_payload_hdr_ver:2.
>>>> a=rtpmap:0 PCMU/8000.
>>>> a=vx_join_audio:1.
>>>> a=vx_join_text:0.
>>>> a=vx_jc:60.
>>>> a=setup:both.
>>>> a=vx_rtcp:0.
>>>> a=direction:active.
>>>> a=oldmediaip:192.168.1.8.
>>>>
>>>> On Wed, Apr 26, 2017 at 6:18 AM, Bogdan-Andrei Iancu
>>>> <bog...@opensips.org> wrote:
>>>>> Hi Satish,
>>>>>
>>>>> For the mime test, you can use the has_body() function:
>>>>> http://www.opensips.org/html/docs/modules/2.2.x/sipmsgops.html#idp3886992
>>>>>
>>>>> About the error - could you post the actual SDP payload generating those
>>>>> errors ?
>>>>>
>>>>> Regards,
>>>>>
>>>>> Bogdan-Andrei Iancu
>>>>> OpenSIPS Founder and Developer
>>>>> http://www.opensips-solutions.com
>>>>>
>>>>> OpenSIPS Summit May 2017 Amsterdam
>>>>> http://www.opensips.org/events/Summit-2017Amsterdam.html
>>>>>
>>>>>
>>>>>> On 04/25/2017 10:35 PM, Satish Patel wrote:
>>>>>> We have some custome Voice solution and in-house media server so right
>>>>>> now i don't care about PORT all i need correct IP address.
>>>>>>
>>>>>> I have tried following and it fixed issue but i am seeing following
>>>>>> error in logs
>>>>>>
>>>>>> if (method=="INVITE") {
>>>>>> if(search("^Content-Type:.*application/sdp")) {
>>>>>> fix_nated_sdp("3");
>>>>>> };
>>>>>> };
>>>>>>
>>>>>>
>>>>>> Error:
>>>>>>
>>>>>> ERROR: extract_mediaip: no `c=' in SDP
>>>>>> ERROR: extract_mediaip: no `c=' in SDP
>>>>>>
>>>>>> Do you know what does that means and how to fix that issue?
>>>>>>
>>>>>> On Mon, Apr 24, 2017 at 11:41 PM, Alex Balashov
>>>>>>
Re: [OpenSIPS-Users] SIP SDP rfc1918 address fix
Yes, whenever fix_nated_sdp() fiction run it produce that error which I
mentioned in my previous email. Every single time.
Sent from my iPhone
> On Apr 26, 2017, at 4:52 PM, Bogdan-Andrei Iancu <bog...@opensips.org> wrote:
>
> So below is the SDP OpenSIPS receives (from network) and when doing
> fix_nated_sdp() on that SDP leads to the "c=" errors ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
> http://www.opensips.org/events/Summit-2017Amsterdam.html
>
>> On 04/26/2017 08:44 PM, Satish Patel wrote:
>> Here is my payload again we have custom application which is using SER
>> so some of them are custom values, This is the payload after i apply
>> fix_nated_sdp() function.
>>
>>
>> Max-Forwards: 16.
>> Content-Type: application/sdp.
>> Content-Length: 418.
>> Supported: path, 100rel.
>> P-hint: LOCAL.
>> P-hint: ALIASED OUTBOUND.
>> P-hint: DIRECT-RTP.
>> .
>> v=0.
>> o=user1 53655765 2353687637 IN IP4 192.168.1.8.
>> s=-.
>> c=IN IP4 173.71.121.4.
>> t=0 0.
>> m=audio 6000 RTP/AVP 0.
>> a=rtpmap:127 VANI/32000.
>> a=fmtp:127 ver=3;mode=3;sub-types=1,7;codecs=0x26.
>> a=rtpmap:111 SIREN14-3D/32000.
>> a=fmtp:111 bitrate=32000.
>> a=vx_payload_hdr_ver:2.
>> a=rtpmap:0 PCMU/8000.
>> a=vx_join_audio:1.
>> a=vx_join_text:0.
>> a=vx_jc:60.
>> a=setup:both.
>> a=vx_rtcp:0.
>> a=direction:active.
>> a=oldmediaip:192.168.1.8.
>>
>> On Wed, Apr 26, 2017 at 6:18 AM, Bogdan-Andrei Iancu
>> <bog...@opensips.org> wrote:
>>> Hi Satish,
>>>
>>> For the mime test, you can use the has_body() function:
>>> http://www.opensips.org/html/docs/modules/2.2.x/sipmsgops.html#idp3886992
>>>
>>> About the error - could you post the actual SDP payload generating those
>>> errors ?
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developer
>>> http://www.opensips-solutions.com
>>>
>>> OpenSIPS Summit May 2017 Amsterdam
>>> http://www.opensips.org/events/Summit-2017Amsterdam.html
>>>
>>>
>>>> On 04/25/2017 10:35 PM, Satish Patel wrote:
>>>> We have some custome Voice solution and in-house media server so right
>>>> now i don't care about PORT all i need correct IP address.
>>>>
>>>> I have tried following and it fixed issue but i am seeing following
>>>> error in logs
>>>>
>>>> if (method=="INVITE") {
>>>> if(search("^Content-Type:.*application/sdp")) {
>>>> fix_nated_sdp("3");
>>>> };
>>>> };
>>>>
>>>>
>>>> Error:
>>>>
>>>> ERROR: extract_mediaip: no `c=' in SDP
>>>> ERROR: extract_mediaip: no `c=' in SDP
>>>>
>>>> Do you know what does that means and how to fix that issue?
>>>>
>>>> On Mon, Apr 24, 2017 at 11:41 PM, Alex Balashov
>>>> <abalas...@evaristesys.com> wrote:
>>>>> The intent of my questions was to get what you think about what you
>>>>> actually want to accomplish. fix_nated_sdp() allows you to replace the
>>>>> IP with the received signalling IP:
>>>>>
>>>>> http://www.opensips.org/html/docs/modules/2.1.x/nathelper.html#id293899
>>>>>
>>>>> But what about the port?
>>>>>
>>>>>> On Mon, Apr 24, 2017 at 11:39:14PM -0400, Satish Patel wrote:
>>>>>>
>>>>>> after google found bunch of post where people suggesting use
>>>>>> fix_nated_sdp() is that right approach ?
>>>>>>
>>>>>> On Mon, Apr 24, 2017 at 11:25 PM, Alex Balashov
>>>>>> <abalas...@evaristesys.com> wrote:
>>>>>>> Yes, but RTP can come from a different address than the signalling
>>>>>>> (SIP). What sense would there be in substituting the source of the SIP
>>>>>>> message in there?
>>>>>>>
>>>>>>>> On Mon, Apr 24, 2017 at 11:23:30PM -0400, Satish Patel wrote:
>>>>>>>>
>>>>>>>> I meant "origin public address of client" if c line isn't public then
>>>>>>>> media never work.
>>>>>>>&
Re: [OpenSIPS-Users] SIP SDP rfc1918 address fix
Here is my payload again we have custom application which is using SER
so some of them are custom values, This is the payload after i apply
fix_nated_sdp() function.
Max-Forwards: 16.
Content-Type: application/sdp.
Content-Length: 418.
Supported: path, 100rel.
P-hint: LOCAL.
P-hint: ALIASED OUTBOUND.
P-hint: DIRECT-RTP.
.
v=0.
o=user1 53655765 2353687637 IN IP4 192.168.1.8.
s=-.
c=IN IP4 173.71.121.4.
t=0 0.
m=audio 6000 RTP/AVP 0.
a=rtpmap:127 VANI/32000.
a=fmtp:127 ver=3;mode=3;sub-types=1,7;codecs=0x26.
a=rtpmap:111 SIREN14-3D/32000.
a=fmtp:111 bitrate=32000.
a=vx_payload_hdr_ver:2.
a=rtpmap:0 PCMU/8000.
a=vx_join_audio:1.
a=vx_join_text:0.
a=vx_jc:60.
a=setup:both.
a=vx_rtcp:0.
a=direction:active.
a=oldmediaip:192.168.1.8.
On Wed, Apr 26, 2017 at 6:18 AM, Bogdan-Andrei Iancu
<bog...@opensips.org> wrote:
> Hi Satish,
>
> For the mime test, you can use the has_body() function:
> http://www.opensips.org/html/docs/modules/2.2.x/sipmsgops.html#idp3886992
>
> About the error - could you post the actual SDP payload generating those
> errors ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
> http://www.opensips.org/events/Summit-2017Amsterdam.html
>
>
> On 04/25/2017 10:35 PM, Satish Patel wrote:
>>
>> We have some custome Voice solution and in-house media server so right
>> now i don't care about PORT all i need correct IP address.
>>
>> I have tried following and it fixed issue but i am seeing following
>> error in logs
>>
>> if (method=="INVITE") {
>> if(search("^Content-Type:.*application/sdp")) {
>> fix_nated_sdp("3");
>> };
>> };
>>
>>
>> Error:
>>
>> ERROR: extract_mediaip: no `c=' in SDP
>> ERROR: extract_mediaip: no `c=' in SDP
>>
>> Do you know what does that means and how to fix that issue?
>>
>> On Mon, Apr 24, 2017 at 11:41 PM, Alex Balashov
>> <abalas...@evaristesys.com> wrote:
>>>
>>> The intent of my questions was to get what you think about what you
>>> actually want to accomplish. fix_nated_sdp() allows you to replace the
>>> IP with the received signalling IP:
>>>
>>> http://www.opensips.org/html/docs/modules/2.1.x/nathelper.html#id293899
>>>
>>> But what about the port?
>>>
>>> On Mon, Apr 24, 2017 at 11:39:14PM -0400, Satish Patel wrote:
>>>
>>>> after google found bunch of post where people suggesting use
>>>> fix_nated_sdp() is that right approach ?
>>>>
>>>> On Mon, Apr 24, 2017 at 11:25 PM, Alex Balashov
>>>> <abalas...@evaristesys.com> wrote:
>>>>>
>>>>> Yes, but RTP can come from a different address than the signalling
>>>>> (SIP). What sense would there be in substituting the source of the SIP
>>>>> message in there?
>>>>>
>>>>> On Mon, Apr 24, 2017 at 11:23:30PM -0400, Satish Patel wrote:
>>>>>
>>>>>> I meant "origin public address of client" if c line isn't public then
>>>>>> media never work.
>>>>>>
>>>>>> c=IN IP4 192.168.1.8.
>>>>>>
>>>>>> It should be
>>>>>>
>>>>>> c=IN IP4
>>>>>>
>>>>>> On Mon, Apr 24, 2017 at 11:04 PM, Alex Balashov
>>>>>> <abalas...@evaristesys.com> wrote:
>>>>>>>
>>>>>>> Satish,
>>>>>>>
>>>>>>> When you say "origin public address", do you mean the external source
>>>>>>> address and port of the SIP message, or the incoming RTP stream?
>>>>>>>
>>>>>>> On Mon, Apr 24, 2017 at 11:00:40PM -0400, Satish Patel wrote:
>>>>>>>
>>>>>>>> In my INVITE/SDP i am seeing sometime rfc1918 address which i want
>>>>>>>> fix
>>>>>>>> and replace it with origin public address. ex
>>>>>>>>
>>>>>>>> I am seeing following info in INVITE
>>>>>>>>
>>>>>>>> v=0.
>>>>>>>> o=amsip 0 0 IN IP4 192.168.1.8.
>>>>>>>> s= .
>>>>>>>> c=IN IP4 192.168.1.8.
>>>>>>>> t=0 0.
>>>>>>>> m=audio 22530 RTP/AVP 127 111 0 101.
>>>>>>>>
>>>>>>>&g
Re: [OpenSIPS-Users] SIP SDP rfc1918 address fix
after google found bunch of post where people suggesting use fix_nated_sdp() is that right approach ? On Mon, Apr 24, 2017 at 11:25 PM, Alex Balashov <abalas...@evaristesys.com> wrote: > Yes, but RTP can come from a different address than the signalling > (SIP). What sense would there be in substituting the source of the SIP > message in there? > > On Mon, Apr 24, 2017 at 11:23:30PM -0400, Satish Patel wrote: > >> I meant "origin public address of client" if c line isn't public then >> media never work. >> >> c=IN IP4 192.168.1.8. >> >> It should be >> >> c=IN IP4 >> >> On Mon, Apr 24, 2017 at 11:04 PM, Alex Balashov >> <abalas...@evaristesys.com> wrote: >> > Satish, >> > >> > When you say "origin public address", do you mean the external source >> > address and port of the SIP message, or the incoming RTP stream? >> > >> > On Mon, Apr 24, 2017 at 11:00:40PM -0400, Satish Patel wrote: >> > >> >> In my INVITE/SDP i am seeing sometime rfc1918 address which i want fix >> >> and replace it with origin public address. ex >> >> >> >> I am seeing following info in INVITE >> >> >> >> v=0. >> >> o=amsip 0 0 IN IP4 192.168.1.8. >> >> s= . >> >> c=IN IP4 192.168.1.8. >> >> t=0 0. >> >> m=audio 22530 RTP/AVP 127 111 0 101. >> >> >> >> ___ >> >> Users mailing list >> >> Users@lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > >> > -- >> > Alex Balashov | Principal | Evariste Systems LLC >> > >> > Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) >> > Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ >> > >> > ___ >> > Users mailing list >> > Users@lists.opensips.org >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > Alex Balashov | Principal | Evariste Systems LLC > > Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) > Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP SDP rfc1918 address fix
I meant "origin public address of client" if c line isn't public then media never work. c=IN IP4 192.168.1.8. It should be c=IN IP4 On Mon, Apr 24, 2017 at 11:04 PM, Alex Balashov <abalas...@evaristesys.com> wrote: > Satish, > > When you say "origin public address", do you mean the external source > address and port of the SIP message, or the incoming RTP stream? > > On Mon, Apr 24, 2017 at 11:00:40PM -0400, Satish Patel wrote: > >> In my INVITE/SDP i am seeing sometime rfc1918 address which i want fix >> and replace it with origin public address. ex >> >> I am seeing following info in INVITE >> >> v=0. >> o=amsip 0 0 IN IP4 192.168.1.8. >> s= . >> c=IN IP4 192.168.1.8. >> t=0 0. >> m=audio 22530 RTP/AVP 127 111 0 101. >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > Alex Balashov | Principal | Evariste Systems LLC > > Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) > Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] SIP SDP rfc1918 address fix
In my INVITE/SDP i am seeing sometime rfc1918 address which i want fix and replace it with origin public address. ex I am seeing following info in INVITE v=0. o=amsip 0 0 IN IP4 192.168.1.8. s= . c=IN IP4 192.168.1.8. t=0 0. m=audio 22530 RTP/AVP 127 111 0 101. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dispatcher + Proxy + NAT question
Thanks Razvan, for experiment i have put dispatcher on Public Interface and second NIC on private interface which is connected to SIP proxy server. On dispatcher i am not handling any NAT stuff all it does (record_route) Do you think i should put NAT module in dispatcher and what kind of NAT function i should use? On Tue, Apr 18, 2017 at 4:37 AM, Răzvan Crainea <raz...@opensips.org> wrote: > Hi, Satish! > > You will probably need to have the dispatcher working as a FrontEnd, with > two interfaces that will work in bridge mode. > Since the dispatcher is the one facing the Public internet, he's the one > that should take care of any NAT handling. > > Best regards, > > Răzvan Crainea > OpenSIPS Solutions > www.opensips-solutions.com > > > On 04/16/2017 03:48 AM, Satish Patel wrote: >> >> Any help here? >> >> On Sat, Apr 15, 2017 at 10:04 AM, Satish Patel <satish@gmail.com> >> wrote: >>> >>> currently we have following design 1 dispatcher and 3 SIP Proxy and >>> every server on Public IP. Dispatcher using as a load-balancer to >>> distribute load on proxy. >>> >>> My dispatcher is very simple and stateless (I am not using record_route() >>> too) >>> >>> Everything working fine, but now we decided to move alls SIP Proxy to >>> Private IP address and just keep dispatcher on Public IP in that case >>> i have to add record_route() function in dispatcher to put itself in >>> path but question is how does NAT will work in this case? who will >>> perform NAT function? >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dispatcher + Proxy + NAT question
Any help here? On Sat, Apr 15, 2017 at 10:04 AM, Satish Patel <satish@gmail.com> wrote: > currently we have following design 1 dispatcher and 3 SIP Proxy and > every server on Public IP. Dispatcher using as a load-balancer to > distribute load on proxy. > > My dispatcher is very simple and stateless (I am not using record_route() too) > > Everything working fine, but now we decided to move alls SIP Proxy to > Private IP address and just keep dispatcher on Public IP in that case > i have to add record_route() function in dispatcher to put itself in > path but question is how does NAT will work in this case? who will > perform NAT function? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Dispatcher + Proxy + NAT question
currently we have following design 1 dispatcher and 3 SIP Proxy and every server on Public IP. Dispatcher using as a load-balancer to distribute load on proxy. My dispatcher is very simple and stateless (I am not using record_route() too) Everything working fine, but now we decided to move alls SIP Proxy to Private IP address and just keep dispatcher on Public IP in that case i have to add record_route() function in dispatcher to put itself in path but question is how does NAT will work in this case? who will perform NAT function? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] port number in record-route
Does this work if i use record_route_preset without PORT number
specify and let SER decide which is the best port to use?
something like this, without PORT specify
record_route_preset("62.32.212.10;nat=yes");
On Fri, Mar 31, 2017 at 4:00 AM, Aqs Younas <aqsyou...@gmail.com> wrote:
> This might help you.
> http://www.opensips.org/html/docs/modules/2.2.x/rr.html#id293864
>
> On 31 March 2017 at 06:31, Satish Patel <satish@gmail.com> wrote:
>>
>> is there a way i can re-write record-route port number?
>>
>> On Wed, Mar 29, 2017 at 6:29 PM, Alex Balashov
>> <abalas...@evaristesys.com> wrote:
>> > Record-Route headers contain URIs, and like any SIP URI, they can
>> > contain a port component. If that port component is omitted, 5060 is
>> > presumed.
>> >
>> > On March 29, 2017 6:27:30 PM EDT, Satish Patel <satish@gmail.com>
>> > wrote:
>> >>what is the use of port number in record-route?
>> >>
>> >>I am having major issue with that look like we are running sip server
>> >>on different port to protect ourself from sip scanner we are using
>> >>non-standard port like 6060/7070 multiple port on single server so it
>> >>will failover to other port if firewall block them.
>> >>
>> >>I am seeing record-route adding first port in listen: directive for
>> >>example
>> >>
>> >>listen=udp:x.x.x.x:7070 udp:x.x.x.x:7070 udp:x.x.x.x:5062
>> >>
>> >>In this case my record-route always using 7070 in header default
>> >>recordless request coming on 5062.
>> >>
>> >>I found one more issue here someone posted while ago
>> >>
>>
>> >> >>https://lists.cs.columbia.edu/pipermail/sip-implementors/2001-March/000601.html
>> >>
>> >>___
>> >>Users mailing list
>> >>Users@lists.opensips.org
>> >>http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> >
>> >
>> > -- Alex
>> >
>> > --
>> > Principal, Evariste Systems LLC (www.evaristesys.com)
>> >
>> > Sent from my Google Nexus.
>> >
>> > ___
>> > Users mailing list
>> > Users@lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
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>
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Re: [OpenSIPS-Users] port number in record-route
is there a way i can re-write record-route port number? On Wed, Mar 29, 2017 at 6:29 PM, Alex Balashov <abalas...@evaristesys.com> wrote: > Record-Route headers contain URIs, and like any SIP URI, they can contain a > port component. If that port component is omitted, 5060 is presumed. > > On March 29, 2017 6:27:30 PM EDT, Satish Patel <satish@gmail.com> wrote: >>what is the use of port number in record-route? >> >>I am having major issue with that look like we are running sip server >>on different port to protect ourself from sip scanner we are using >>non-standard port like 6060/7070 multiple port on single server so it >>will failover to other port if firewall block them. >> >>I am seeing record-route adding first port in listen: directive for >>example >> >>listen=udp:x.x.x.x:7070 udp:x.x.x.x:7070 udp:x.x.x.x:5062 >> >>In this case my record-route always using 7070 in header default >>recordless request coming on 5062. >> >>I found one more issue here someone posted while ago >> >>https://lists.cs.columbia.edu/pipermail/sip-implementors/2001-March/000601.html >> >>___ >>Users mailing list >>Users@lists.opensips.org >>http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- Alex > > -- > Principal, Evariste Systems LLC (www.evaristesys.com) > > Sent from my Google Nexus. > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] port number in record-route
what is the use of port number in record-route? I am having major issue with that look like we are running sip server on different port to protect ourself from sip scanner we are using non-standard port like 6060/7070 multiple port on single server so it will failover to other port if firewall block them. I am seeing record-route adding first port in listen: directive for example listen=udp:x.x.x.x:7070 udp:x.x.x.x:7070 udp:x.x.x.x:5062 In this case my record-route always using 7070 in header default recordless request coming on 5062. I found one more issue here someone posted while ago https://lists.cs.columbia.edu/pipermail/sip-implementors/2001-March/000601.html ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP default port routing issue
Thanks you, I think i missed your email, but you said interface in my case its PORT. we have multiple port listening for SIP request and we have notice INVITE comes on 5062 but when server send 200 OK it use 5060 instead of 5062 where it received invite. On Mon, Mar 27, 2017 at 3:29 AM, Răzvan Crainea <raz...@opensips.org> wrote: > Please register on the mailing list, I have already replied to this > thread[1], but I guess you didin't get the reply. > > [1] http://lists.opensips.org/pipermail/users/2017-March/036849.html > > Best regards, > > Răzvan Crainea > OpenSIPS Solutions > www.opensips-solutions.com > > > On 03/26/2017 08:16 PM, Satish Patel wrote: >> >> any suggestion? >> >> On Tue, Mar 21, 2017 at 6:48 PM, Satish Patel <satish@gmail.com> >> wrote: >>> >>> This is little tricky question, we are developing softphone and we put >>> logic in phone it will try to connect 5060 if it's blocked by some >>> country then it will try 5061 if that is block then try 5062 >>> >>> Now on OpenSIPS we are listening on all 3 ports 5060, 5061 and 5062. >>> Now problem is here INVITE goes on correct port but when server send >>> 200 OK mesg it will always pick first port in listen: directive, how >>> do i synchronize communication to specific port where INVITE comes >>> from? >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP default port routing issue
any suggestion? On Tue, Mar 21, 2017 at 6:48 PM, Satish Patel <satish@gmail.com> wrote: > This is little tricky question, we are developing softphone and we put > logic in phone it will try to connect 5060 if it's blocked by some > country then it will try 5061 if that is block then try 5062 > > Now on OpenSIPS we are listening on all 3 ports 5060, 5061 and 5062. > Now problem is here INVITE goes on correct port but when server send > 200 OK mesg it will always pick first port in listen: directive, how > do i synchronize communication to specific port where INVITE comes > from? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] SIP default port routing issue
This is little tricky question, we are developing softphone and we put logic in phone it will try to connect 5060 if it's blocked by some country then it will try 5061 if that is block then try 5062 Now on OpenSIPS we are listening on all 3 ports 5060, 5061 and 5062. Now problem is here INVITE goes on correct port but when server send 200 OK mesg it will always pick first port in listen: directive, how do i synchronize communication to specific port where INVITE comes from? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] dispatcher with t_relay performance
Before LB was stateless and it was working fine. but we added new NIC
on this and enabled mhomed=1 and it broke routing because coun't
figure out right socket.. so i changed forward() to t_relay() and it
works fine again.. now my question is does t_relay() impact
performance... or any kind of issue? to having stateful LB ?
On Mon, Mar 7, 2016 at 10:35 AM, SamyGo <govoi...@gmail.com> wrote:
> Oh, I thought it was a typo, 200,000 CPS ! Well I'd say to not spend much
> time thinking about t_relay() rather spend energy on designing an
> architecture that can give you the flexibility and scalability options.
>
> For example:
> A DNS SRV pointing to a layer of stateless dispatcher OpenSIPS. These
> stateless OpenSIPS just don't care about any business logic just do a rough
> load-balancing and "redirect" to the second layer OpenSIPS.
> The second layer of OpenSIPS do the business logic and stay in call i.e use
> t_relay()
>
> That is a simple example in which you can add as many OpenSIPS at both
> layers to manage your 200K CPS.
>
> There could be way too many different ways of handling your 200K CPS load,
> it all depends on your business logic, type of SIP requests and calls etc,
> location of the end users/regions, methods to tweak your business logic i.e
> use of caches and NoSQL DBs, and so much that only you may know at this
> point.
>
> Please go through this link: http://www.opensips.org/About/PerformanceTests
> to see results for different types of configurations. However, do keep in
> mind that those results may be done on older versions of OpenSIPS and you
> may want to stress test your setup separately to know what are your
> capabilities.
>
> Regards,
> Sammy
>
>
>
> On Mon, Mar 7, 2016 at 8:54 AM, Satish Patel <satish@gmail.com> wrote:
>>
>> We have 200,000 CPS and more in future. Just worried about t_relay() and
>> its performance. Any idea?
>>
>> --
>> Sent from my iPhone
>>
>> On Mar 6, 2016, at 2:44 PM, SamyGo <govoi...@gmail.com> wrote:
>>
>> I'd ask you to read difference between Load_balancer and Dispatcher
>> module. Dispatcher module is not an accurate measure but it is the only
>> option when it comes to load balancing REGISTER requests.
>>
>> Dispatcher is hence very light weight as compared to Load Balancer. For a
>> 200 CPS calls Load Balancer or Dispatcehr won't be putting any bigger impact
>> relative to the business logic itself. For example doing alot of DB queries,
>> engaging various other modules etc these things really define how light or
>> heavy your system is going to be.
>>
>> Regards,
>> Sammy
>>
>>
>> On Sun, Mar 6, 2016 at 10:36 AM, Satish Patel <satish@gmail.com>
>> wrote:
>>>
>>> Any thought on it???
>>>
>>> On Fri, Mar 4, 2016 at 1:30 PM, Satish Patel <satish@gmail.com>
>>> wrote:
>>> > We have dispatcher and we are using very simple code block like
>>> > following
>>> >
>>> > if (method=="REGISTER" || method=="INVITE" ) {
>>> > ds_select_dst("1", "2");
>>> > t_relay();
>>> >}
>>> >
>>> > Does t_relay will keep all transaction in memory? and what will be the
>>> > performance issue? we have ~200k cps calls.. what will be the impact?
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
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Re: [OpenSIPS-Users] dispatcher with t_relay performance
We have 200,000 CPS and more in future. Just worried about t_relay() and its
performance. Any idea?
--
Sent from my iPhone
> On Mar 6, 2016, at 2:44 PM, SamyGo <govoi...@gmail.com> wrote:
>
> I'd ask you to read difference between Load_balancer and Dispatcher module.
> Dispatcher module is not an accurate measure but it is the only option when
> it comes to load balancing REGISTER requests.
>
> Dispatcher is hence very light weight as compared to Load Balancer. For a 200
> CPS calls Load Balancer or Dispatcehr won't be putting any bigger impact
> relative to the business logic itself. For example doing alot of DB queries,
> engaging various other modules etc these things really define how light or
> heavy your system is going to be.
>
> Regards,
> Sammy
>
>
>> On Sun, Mar 6, 2016 at 10:36 AM, Satish Patel <satish@gmail.com> wrote:
>> Any thought on it???
>>
>> On Fri, Mar 4, 2016 at 1:30 PM, Satish Patel <satish@gmail.com> wrote:
>> > We have dispatcher and we are using very simple code block like following
>> >
>> > if (method=="REGISTER" || method=="INVITE" ) {
>> > ds_select_dst("1", "2");
>> > t_relay();
>> >}
>> >
>> > Does t_relay will keep all transaction in memory? and what will be the
>> > performance issue? we have ~200k cps calls.. what will be the impact?
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
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Re: [OpenSIPS-Users] dispatcher with t_relay performance
Any thought on it???
On Fri, Mar 4, 2016 at 1:30 PM, Satish Patel <satish@gmail.com> wrote:
> We have dispatcher and we are using very simple code block like following
>
> if (method=="REGISTER" || method=="INVITE" ) {
> ds_select_dst("1", "2");
> t_relay();
>}
>
> Does t_relay will keep all transaction in memory? and what will be the
> performance issue? we have ~200k cps calls.. what will be the impact?
___
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[OpenSIPS-Users] dispatcher with t_relay performance
We have dispatcher and we are using very simple code block like following
if (method=="REGISTER" || method=="INVITE" ) {
ds_select_dst("1", "2");
t_relay();
}
Does t_relay will keep all transaction in memory? and what will be the
performance issue? we have ~200k cps calls.. what will be the impact?
___
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Re: [OpenSIPS-Users] mhome asymmetric port issue
Thanks Razvan,
Problem is we have legacy application running on ser-0.10 older
version and we have did lots of other customization in ser I tried to
use force_send_socket() but look like that support isn't in RR module.
if i enable "mhomed=1" and use t_relay() in dispatcher then its
consuming REGISTER packet and sending AUTH challenge to client instead
of sending that REGISTER to backend dispatcher..
if i use forward(uri:host, uri:port); function then it doesn't
understand socket correctly. atleast t_relay() is working but
consuming REGISTER, we have very simple code like following
else if ( (method=="REGISTER") || (method=="INVITE") ) {
if ( !ds_select_dst("2", "2") ) {
xlog("L_ERR", "Unable to route REGISTER\n");
sl_send_reply("500","Unable to route REGISTER");
break;
}
..
..
t_relay()
On Thu, Mar 3, 2016 at 3:39 AM, Răzvan Crainea <raz...@opensips.org> wrote:
> Hi, Satish!
>
> By default, OpenSIPS uses the same interface to send the reply. However,
> when using mhomed=1, the operating system decides where the reply should be
> sent to. And in your case, the operating system simply chooses a different
> interface. So it seems this is the normal behavior, there's nothing wrong.
> If you really want to use the same interface for replies, you should use the
> force_send_socket() function to set the desired interface.
>
> Best regards,
> Răzvan
>
>
> On 03/02/2016 11:10 PM, Satish Patel wrote:
>>
>> mhome=1
>> listen=udp:10.0.0.1:6060 udp:10.0.0.1:5060 udp:192.168.100.1:6060
>> udp:192.168.100.1:5060
>>
>> From client when i send REGISTER to 5060 then server sending reply
>> back using port 6060, it should send reply back client using 5060
>> right???
>>
>> If i use mhome=0 everything works!
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> --
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
>
> ___
> Users mailing list
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
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[OpenSIPS-Users] mhome asymmetric port issue
mhome=1 listen=udp:10.0.0.1:6060 udp:10.0.0.1:5060 udp:192.168.100.1:6060 udp:192.168.100.1:5060 >From client when i send REGISTER to 5060 then server sending reply back using port 6060, it should send reply back client using 5060 right??? If i use mhome=0 everything works! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Dispatcher with multiple interface
I have following scenario [client]-Public-IP--[dispatcher]--LAN-IP--[proxy] Dispatcher has multi home interface, public and private, when client send request to dispatcher public Interface then dispatcher should use private LAN IP to send that request to Proxy can dispatcher do that? Current dispatcher only using public IP to send request to proxy. I wants it use LAN IP to forward request. How that will possible ? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIP 2.1 WebRTC missing Received header in usrloc
I am following this document: http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1 My sipML5 client successfully register but somehow its not calling each other. I check AOS and it looks strange, where is the received: header to contact client? AOR:: 1...@sip.example.com Contact:: sip:1001@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws Q= Expires:: 191 Callid:: 589a0c22-f016-8ac6-8721-3d535c0dd836 Cseq:: 5318 User-agent:: IM-client/OMA1.0 sipML5-v1.2015.03.18 State:: CS_NEW Flags:: 0 Cflags:: Socket:: udp:182.XX.XX.164:5060 Methods:: 4294967295 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Transparent Auth with WebRTC
Thanks Eric,
I tried following, now its forwarding REGISTER packet to asterisk but
authentication failed, I have check username/password is correct on
asterisk. Do you think it is because of realm ?
if (is_method(REGISTER)) {
rewritehostport(asterisk:5060);
route(relay);
exit;
}
On Wed, Jun 24, 2015 at 9:24 AM, Eric Tamme e...@uphreak.com wrote:
just t_relay the request to your other server... OpenSIPS wont
automatically challenge anything
On 06/24/2015 07:22 AM, Satish Patel wrote:
All,
I have special requirement which is little odd, I want to use WebRTC
with Opensips but all REGISTER process will done by other SIP server,
Example:
[UA][WebRTC-Opensips]---[Asterisk/Freeswitch]
UA will use WebRTC of Opensips but opensips forward all REGISTER request
to Asterisk/Freeswitch and user will authenticate their... In short
Opensips will just Proxy Auth request.
How it will be possible?
___
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___
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[OpenSIPS-Users] Transparent Auth with WebRTC
All, I have special requirement which is little odd, I want to use WebRTC with Opensips but all REGISTER process will done by other SIP server, Example: [UA][WebRTC-Opensips]---[Asterisk/Freeswitch] UA will use WebRTC of Opensips but opensips forward all REGISTER request to Asterisk/Freeswitch and user will authenticate their... In short Opensips will just Proxy Auth request. How it will be possible? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] siptarce enable/disable without restart service
But siptrace() has no connection with cache store (memory, etc)... it is
just raw sip messages store in database or send to Homer (HEP)
On Tue, Jun 2, 2015 at 11:53 AM, Jarrod Baumann jar...@unixc.org wrote:
You could wrap it in an if statement that checked a cache store (memory,
etc) value if that didn’t pose too great a performance issue? I haven’t
tried it, just thinking outloud.
On Jun 2, 2015, at 10:45 AM, Satish Patel satish@gmail.com wrote:
We have opensips 2.1, is there anyway we can enable disable siptrace
without restart opensips? because it will be very helpful.. we don't want
to keep it on for no reason..
route {
...
...
sip_trace();
}
___
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[OpenSIPS-Users] siptarce enable/disable without restart service
We have opensips 2.1, is there anyway we can enable disable siptrace
without restart opensips? because it will be very helpful.. we don't want
to keep it on for no reason..
route {
...
...
sip_trace();
}
___
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Re: [OpenSIPS-Users] siptarce enable/disable without restart service
nevermind, i found answer.
opensipsctl fifo sip_trace off
opensipsctl fifo sip_trace on
On Tue, Jun 2, 2015 at 11:58 AM, Satish Patel satish@gmail.com wrote:
But siptrace() has no connection with cache store (memory, etc)... it is
just raw sip messages store in database or send to Homer (HEP)
On Tue, Jun 2, 2015 at 11:53 AM, Jarrod Baumann jar...@unixc.org wrote:
You could wrap it in an if statement that checked a cache store (memory,
etc) value if that didn’t pose too great a performance issue? I haven’t
tried it, just thinking outloud.
On Jun 2, 2015, at 10:45 AM, Satish Patel satish@gmail.com wrote:
We have opensips 2.1, is there anyway we can enable disable siptrace
without restart opensips? because it will be very helpful.. we don't want
to keep it on for no reason..
route {
...
...
sip_trace();
}
___
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___
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___
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[OpenSIPS-Users] opensips 1.7.2 shared memory without compile
we have old opensips 1.7.2 but running with default options, we want to increase shared memory so do we need to re-compile it? As per this document http://www.opensips.org/Documentation/TroubleShooting-IncreaseMem Re-compile only for Private memory or for it apply to shared memory too? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] URGENT- Openstack VMware Guest VM created but Openstack got error
Sorry guys!!! On Sat, May 9, 2015 at 9:05 PM, Satish Patel satish@gmail.com wrote: Hi, We have integrate Openstack with vCenter VMware and when i create instance on Openstack GUI it created successfully on VMware but it throwing error on Openstack. Look like openstack trying 3 time to create VM even VM is already running on VMware. I don't know what is wrong here, why openstack doesn't know about running VMware VM status. Please need your suggesstion to solve this issue. 2015-05-09 20:48:48.597 29674 INFO oslo.messaging._drivers.impl_qpid [-] Connected to AMQP server on 172.16.120.14:5672 2015-05-09 20:48:48.625 29674 INFO nova.scheduler.filter_scheduler [req-1e440c82-21c4-4010-9d99-b2623c7547a1 b872b754f8774f64b3344e1ef9c245db 49de35ef22bb4bb08355fb2bb163a588] Attempting to build 1 instance(s) uuids: [u'060dba43-3065-4927-bf1f-50abdc72da7d'] 2015-05-09 20:48:48.695 29674 INFO nova.scheduler.filter_scheduler [req-1e440c82-21c4-4010-9d99-b2623c7547a1 b872b754f8774f64b3344e1ef9c245db 49de35ef22bb4bb08355fb2bb163a588] Choosing host WeighedHost [host: lnx1iico08x, weight: 1.0] for instance 060dba43-3065-4927-bf1f-50abdc72da7d 2015-05-09 20:49:48.299 29674 INFO nova.scheduler.filter_scheduler [req-1e440c82-21c4-4010-9d99-b2623c7547a1 b872b754f8774f64b3344e1ef9c245db 49de35ef22bb4bb08355fb2bb163a588] Attempting to build 1 instance(s) uuids: [u'060dba43-3065-4927-bf1f-50abdc72da7d'] 2015-05-09 20:49:48.302 29674 ERROR nova.scheduler.filter_scheduler [req-1e440c82-21c4-4010-9d99-b2623c7547a1 b872b754f8774f64b3344e1ef9c245db 49de35ef22bb4bb08355fb2bb163a588] [instance: 060dba43-3065-4927-bf1f-50abdc72da7d] Error from last host: lnx1iico08x (node domain-c18(ICO-VM)): [u'Traceback (most recent call last):\n', u' File /usr/lib/python2.6/site-packages/nova/compute/manager.py, line 1340, in _build_instance\nset_access_ip=set_access_ip)\n', u' File /usr/lib/python2.6/site-packages/nova/compute/manager.py, line 401, in decorated_function\nreturn function(self, context, *args, **kwargs)\n', u' File /usr/lib/python2.6/site-packages/nova/compute/manager.py, line 1779, in _spawn\nLOG.exception(_(\'Instance failed to spawn\'), instance=instance)\n', u' File /usr/lib/python2.6/site-packages/nova/openstack/common/excutils.py, line 68, in __exit__\nsix.reraise(self.type_, self.value, self.tb)\n', u' File /usr/lib/python2.6/site-packages/nova/compute/manager.py, line 1765, in _spawn\nblock_device_info)\n', u' File /usr/lib/python2.6/site-packages/nova/virt/vmwareapi/driver.py, line 854, in spawn\nadmin_password, network_info, block_device_info)\n', u' File /usr/lib/python2.6/site-packages/nova/virt/vmwareapi/vmops.py, line 757, in spawn\n_power_on_vm()\n', u' File /usr/lib/python2.6/site-packages/nova/virt/vmwareapi/vmops.py, line 540, in _power_on_vm\nself._session._wait_for_task(power_on_task)\n', u' File /usr/lib/python2.6/site-packages/nova/virt/vmwareapi/driver.py, line 1220, in _wait_for_task\nret_val = done.wait()\n', u' File /usr/lib/python2.6/site-packages/eventlet/event.py, line 116, in wait\nreturn hubs.get_hub().switch()\n', u' File /usr/lib/python2.6/site-packages/eventlet/hubs/hub.py, line 187, in switch\nreturn self.greenlet.switch()\n', uAttributeError: TaskInfo instance has no attribute 'name'\n] 2015-05-09 20:49:48.366 29674 INFO nova.scheduler.filter_scheduler [req-1e440c82-21c4-4010-9d99-b2623c7547a1 b872b754f8774f64b3344e1ef9c245db 49de35ef22bb4bb08355fb2bb163a588] Choosing host WeighedHost [host: lnx1iico08x, weight: 1.0] for instance 060dba43-3065-4927-bf1f-50abdc72da7d 2015-05-09 20:49:59.409 29674 INFO nova.scheduler.filter_scheduler [req-1e440c82-21c4-4010-9d99-b2623c7547a1 b872b754f8774f64b3344e1ef9c245db 49de35ef22bb4bb08355fb2bb163a588] Attempting to build 1 instance(s) uuids: [u'060dba43-3065-4927-bf1f-50abdc72da7d'] 2015-05-09 20:49:59.411 29674 ERROR nova.scheduler.filter_scheduler [req-1e440c82-21c4-4010-9d99-b2623c7547a1 b872b754f8774f64b3344e1ef9c245db 49de35ef22bb4bb08355fb2bb163a588] [instance: 060dba43-3065-4927-bf1f-50abdc72da7d] Error from last host: lnx1iico08x (node domain-c18(ICO-VM)): [u'Traceback (most recent call last):\n', u' File /usr/lib/python2.6/site-packages/nova/compute/manager.py, line 1340, in _build_instance\nset_access_ip=set_access_ip)\n', u' File /usr/lib/python2.6/site-packages/nova/compute/manager.py, line 401, in decorated_function\nreturn function(self, context, *args, **kwargs)\n', u' File /usr/lib/python2.6/site-packages/nova/compute/manager.py, line 1779, in _spawn\nLOG.exception(_(\'Instance failed to spawn\'), instance=instance)\n', u' File /usr/lib/python2.6/site-packages/nova/openstack/common/excutils.py, line 68, in __exit__\nsix.reraise(self.type_, self.value, self.tb)\n', u' File /usr/lib/python2.6/site-packages/nova/compute/manager.py, line 1765, in _spawn\nblock_device_info)\n', u' File /usr/lib
[OpenSIPS-Users] URGENT- Openstack VMware Guest VM created but Openstack got error
Hi, We have integrate Openstack with vCenter VMware and when i create instance on Openstack GUI it created successfully on VMware but it throwing error on Openstack. Look like openstack trying 3 time to create VM even VM is already running on VMware. I don't know what is wrong here, why openstack doesn't know about running VMware VM status. Please need your suggesstion to solve this issue. 2015-05-09 20:48:48.597 29674 INFO oslo.messaging._drivers.impl_qpid [-] Connected to AMQP server on 172.16.120.14:5672 2015-05-09 20:48:48.625 29674 INFO nova.scheduler.filter_scheduler [req-1e440c82-21c4-4010-9d99-b2623c7547a1 b872b754f8774f64b3344e1ef9c245db 49de35ef22bb4bb08355fb2bb163a588] Attempting to build 1 instance(s) uuids: [u'060dba43-3065-4927-bf1f-50abdc72da7d'] 2015-05-09 20:48:48.695 29674 INFO nova.scheduler.filter_scheduler [req-1e440c82-21c4-4010-9d99-b2623c7547a1 b872b754f8774f64b3344e1ef9c245db 49de35ef22bb4bb08355fb2bb163a588] Choosing host WeighedHost [host: lnx1iico08x, weight: 1.0] for instance 060dba43-3065-4927-bf1f-50abdc72da7d 2015-05-09 20:49:48.299 29674 INFO nova.scheduler.filter_scheduler [req-1e440c82-21c4-4010-9d99-b2623c7547a1 b872b754f8774f64b3344e1ef9c245db 49de35ef22bb4bb08355fb2bb163a588] Attempting to build 1 instance(s) uuids: [u'060dba43-3065-4927-bf1f-50abdc72da7d'] 2015-05-09 20:49:48.302 29674 ERROR nova.scheduler.filter_scheduler [req-1e440c82-21c4-4010-9d99-b2623c7547a1 b872b754f8774f64b3344e1ef9c245db 49de35ef22bb4bb08355fb2bb163a588] [instance: 060dba43-3065-4927-bf1f-50abdc72da7d] Error from last host: lnx1iico08x (node domain-c18(ICO-VM)): [u'Traceback (most recent call last):\n', u' File /usr/lib/python2.6/site-packages/nova/compute/manager.py, line 1340, in _build_instance\nset_access_ip=set_access_ip)\n', u' File /usr/lib/python2.6/site-packages/nova/compute/manager.py, line 401, in decorated_function\nreturn function(self, context, *args, **kwargs)\n', u' File /usr/lib/python2.6/site-packages/nova/compute/manager.py, line 1779, in _spawn\nLOG.exception(_(\'Instance failed to spawn\'), instance=instance)\n', u' File /usr/lib/python2.6/site-packages/nova/openstack/common/excutils.py, line 68, in __exit__\nsix.reraise(self.type_, self.value, self.tb)\n', u' File /usr/lib/python2.6/site-packages/nova/compute/manager.py, line 1765, in _spawn\nblock_device_info)\n', u' File /usr/lib/python2.6/site-packages/nova/virt/vmwareapi/driver.py, line 854, in spawn\nadmin_password, network_info, block_device_info)\n', u' File /usr/lib/python2.6/site-packages/nova/virt/vmwareapi/vmops.py, line 757, in spawn\n_power_on_vm()\n', u' File /usr/lib/python2.6/site-packages/nova/virt/vmwareapi/vmops.py, line 540, in _power_on_vm\nself._session._wait_for_task(power_on_task)\n', u' File /usr/lib/python2.6/site-packages/nova/virt/vmwareapi/driver.py, line 1220, in _wait_for_task\nret_val = done.wait()\n', u' File /usr/lib/python2.6/site-packages/eventlet/event.py, line 116, in wait\nreturn hubs.get_hub().switch()\n', u' File /usr/lib/python2.6/site-packages/eventlet/hubs/hub.py, line 187, in switch\nreturn self.greenlet.switch()\n', uAttributeError: TaskInfo instance has no attribute 'name'\n] 2015-05-09 20:49:48.366 29674 INFO nova.scheduler.filter_scheduler [req-1e440c82-21c4-4010-9d99-b2623c7547a1 b872b754f8774f64b3344e1ef9c245db 49de35ef22bb4bb08355fb2bb163a588] Choosing host WeighedHost [host: lnx1iico08x, weight: 1.0] for instance 060dba43-3065-4927-bf1f-50abdc72da7d 2015-05-09 20:49:59.409 29674 INFO nova.scheduler.filter_scheduler [req-1e440c82-21c4-4010-9d99-b2623c7547a1 b872b754f8774f64b3344e1ef9c245db 49de35ef22bb4bb08355fb2bb163a588] Attempting to build 1 instance(s) uuids: [u'060dba43-3065-4927-bf1f-50abdc72da7d'] 2015-05-09 20:49:59.411 29674 ERROR nova.scheduler.filter_scheduler [req-1e440c82-21c4-4010-9d99-b2623c7547a1 b872b754f8774f64b3344e1ef9c245db 49de35ef22bb4bb08355fb2bb163a588] [instance: 060dba43-3065-4927-bf1f-50abdc72da7d] Error from last host: lnx1iico08x (node domain-c18(ICO-VM)): [u'Traceback (most recent call last):\n', u' File /usr/lib/python2.6/site-packages/nova/compute/manager.py, line 1340, in _build_instance\nset_access_ip=set_access_ip)\n', u' File /usr/lib/python2.6/site-packages/nova/compute/manager.py, line 401, in decorated_function\nreturn function(self, context, *args, **kwargs)\n', u' File /usr/lib/python2.6/site-packages/nova/compute/manager.py, line 1779, in _spawn\nLOG.exception(_(\'Instance failed to spawn\'), instance=instance)\n', u' File /usr/lib/python2.6/site-packages/nova/openstack/common/excutils.py, line 68, in __exit__\nsix.reraise(self.type_, self.value, self.tb)\n', u' File /usr/lib/python2.6/site-packages/nova/compute/manager.py, line 1765, in _spawn\nblock_device_info)\n', u' File /usr/lib/python2.6/site-packages/nova/virt/vmwareapi/driver.py, line 854, in spawn\nadmin_password, network_info, block_device_info)\n', u' File
[OpenSIPS-Users] Opensips 2.1 Asynchronous explaine
Hi, I have installed 2.1 but i didn't understand use of Async mode. I was reading article http://www.opensips.org/Documentation/Script-Async-2-1 But don't understand how and where i can use in my script because in my script i am doing some SQL operation but i don't know how it can fit and how good i use it. Can someone explain in layman's terms? http://www.urbandictionary.com/define.php?term=layman%27s+termsdefid=2021603 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] userblacklist strange behavior
It wasn't working because i was using
$fn = $avp(user)
My string was double quote like 1001 because of that it won't
comparing true result with blacklist table and result was false when i
strip out from $fn then it works!
On Mon, May 4, 2015 at 5:34 AM, Liviu Chircu li...@opensips.org wrote:
Hi Satish,
Whether you use integers or string there, performance should be similar.
However, integers are limited to 4294967295 before overflowing - hence why
it's not working for you.
So, proper way of usage:
$avp(user) = 1001;
Best regards,
Liviu Chircu
OpenSIPS Developerhttp://www.opensips-solutions.com
On 30.04.2015 23:34, Satish Patel wrote:
mysql select * from userblacklist;
++--+++---+
| id | username | domain | prefix | whitelist |
++--+++---+
| 1 | 1001 ||| 0 |
| 2 | 9198362323 ||| 0 |
++--+++---+
If i set $avp(user) = 1001; then it works!
but if i set $avp(user) = 9198362323; doesn't work what is the
problem ?
route[user_blacklist] {
$avp(user) = 9198362323;
if (!check_user_blacklist($avp(user), $avp(i:82))) {
sl_send_reply(403, User Blocked);
exit;
};
}
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Re: [OpenSIPS-Users] ANIs check for source number
Do you think instead of using dialplan i can just use Regex like
^(011|\+)[1-9][0-9]{10,14} to match Source string? Dialplan is advance
method which use transformation etc.. we we don't need.
We just want to make sure Source address is compliant with ANIs or E.164
standard because out SIP provide doesn't allow random 4 or 5 digit string
in RPID or FROM.
What do you suggest?
On Thu, Apr 30, 2015 at 12:56 PM, Podrigal, Aron ar...@guaranteedplus.com
wrote:
Have a look here
http://www.opensips.org/html/docs/modules/1.11.x/dialplan.html#id294016
On Thu, Apr 30, 2015 at 12:49 PM, Satish Patel satish@gmail.com
wrote:
I believe dialplan only check destination number, doesn't it? How it
will work with Source number which is coming inside RPID/PAI/From: header?
On Thu, Apr 30, 2015 at 12:30 PM, Podrigal, Aron
ar...@guaranteedplus.com wrote:
You can use the dialplan module to check against valid ANIs and reply
with a 403.
On Thu, Apr 30, 2015 at 12:09 PM, Satish Patel satish@gmail.com
wrote:
Question: what happen if client send call using random string in
RPID/PAI/From header (non-standard ANIs) How to verify ANIs for those
headers? is there a blacklist or any method which scan ANIs for source and
block them?
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--
Aron Podrigal
-
//Be happy :-)
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___
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http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Aron Podrigal
-
//Be happy :-)
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Re: [OpenSIPS-Users] ANIs check for source number
I believe dialplan only check destination number, doesn't it? How it will work with Source number which is coming inside RPID/PAI/From: header? On Thu, Apr 30, 2015 at 12:30 PM, Podrigal, Aron ar...@guaranteedplus.com wrote: You can use the dialplan module to check against valid ANIs and reply with a 403. On Thu, Apr 30, 2015 at 12:09 PM, Satish Patel satish@gmail.com wrote: Question: what happen if client send call using random string in RPID/PAI/From header (non-standard ANIs) How to verify ANIs for those headers? is there a blacklist or any method which scan ANIs for source and block them? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Aron Podrigal - //Be happy :-) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] userblacklist strange behavior
mysql select * from userblacklist;
++--+++---+
| id | username | domain | prefix | whitelist |
++--+++---+
| 1 | 1001 ||| 0 |
| 2 | 9198362323 ||| 0 |
++--+++---+
If i set $avp(user) = 1001; then it works!
but if i set $avp(user) = 9198362323; doesn't work what is the problem ?
route[user_blacklist] {
$avp(user) = 9198362323;
if (!check_user_blacklist($avp(user), $avp(i:82))) {
sl_send_reply(403, User Blocked);
exit;
};
}
___
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Re: [OpenSIPS-Users] ANIs check for source number
i tried to using is_uri_user_e164 () but it will allow anything with '+'
sign, like if i set +1001 its allowing...
I want to match E.164 or ANI string match logic so it should be something
like 16463272823 44635364894 like those number..
i don't want send random string to my SIP provide so they can block me..
On Thu, Apr 30, 2015 at 4:38 PM, Newlin, Ben ben.new...@inin.com wrote:
There is also a function in the URI module for checking if the user
portion of a URI is an E.164 number.
http://www.opensips.org/html/docs/modules/1.11.x/uri.html#id294513
From: Podrigal, Aron
Reply-To: OpenSIPS users mailling list
Date: Thursday, April 30, 2015 at 1:20 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] ANIs check for source number
yes of course you can use just a regex. That would need to match any
standard ^\+?[1-9]\d{1,14}$ BTW, 011 (as countrycode plus number) I
think is invalid form, the stranded allows ANI to be up to 15 digit and
does not have a min length.
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Re: [OpenSIPS-Users] userblacklist strange behavior
nevermind, i found issue it was double quote string so use
*$(fn{s.select,1,\})
to get rid on it*
On Thu, Apr 30, 2015 at 4:34 PM, Satish Patel satish@gmail.com wrote:
mysql select * from userblacklist;
++--+++---+
| id | username | domain | prefix | whitelist |
++--+++---+
| 1 | 1001 ||| 0 |
| 2 | 9198362323 ||| 0 |
++--+++---+
If i set $avp(user) = 1001; then it works!
but if i set $avp(user) = 9198362323; doesn't work what is the problem
?
route[user_blacklist] {
$avp(user) = 9198362323;
if (!check_user_blacklist($avp(user), $avp(i:82))) {
sl_send_reply(403, User Blocked);
exit;
};
}
___
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Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] ANIs check for source number
Question: what happen if client send call using random string in RPID/PAI/From header (non-standard ANIs) How to verify ANIs for those headers? is there a blacklist or any method which scan ANIs for source and block them? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] From header ANI standard number using RPID
I want to overwrite FROM: header using RPID in opensips. but i don't know where to put code and how to use that rpid feature in subscriber table Does any one has sample code which i can try on my config because i don't know how to call rpid and modify FROM address in standard ANI format ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Manually load rpid from subscriber table
We are using IP base authentication so how do i load rpid value from subscriber table? or do i need to do SQL query function to load? is there any built in function in opensips to manually load rpid from subscriber table if we don't want to use Registration method? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips-cp dispatcher issue
I found issue, you have to add destination with port otherwise it won't give you status sip:10.10.10.10:5060 One more issue, in opensips-cp it is showing status Active and Inactive but it is not showing Probing status On Tue, Apr 21, 2015 at 6:56 PM, Podrigal, Aron ar...@guaranteedplus.com wrote: status can be either Enabled, Disabled, Probing so what you see in the web interface is I guess that it is not disabled. Correct me if I'm wrong. On Tue, Apr 21, 2015 at 6:44 PM, Satish Patel satish@gmail.com wrote: In command line it is showing root@dispatcher1:/var/www# opensipsctl dispatcher dump SET:: 1 URI:: sip:173.XXX.XXX.181 state=Probing But on Opensips-cp web interface it is showing DB state: Active Look like it is not syncing with opensips properly. what would be the issue? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Aron Podrigal - //Be happy :-) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips 1.11 postgress DB BUG
Hey Liviu, I have fixed script here is the pull request https://github.com/OpenSIPS/opensips/pull/476 On Wed, Apr 22, 2015 at 8:27 AM, Liviu Chircu li...@opensips.org wrote: Hi Aron/Satish, If you can put together a working Pull Request, I will gladly review it. If bash is not your strong suite, then could you please open a Bug Report [1]? Sooner or later, it will get solved :) [1] https://github.com/OpenSIPS/opensips/issues?q=is%3Aopen+is%3Aissue+label%3Abug Best regards, Liviu Chircu OpenSIPS Developerhttp://www.opensips-solutions.com On 21.04.2015 23:30, Podrigal, Aron wrote: MySQL has a option --password to provide the password on command line. postgres however does not. but here is some refs how you can accomplish what you would like. http://www.postgresql.org/docs/9.4/static/libpq-pgpass.html On Tue, Apr 21, 2015 at 4:26 PM, Satish Patel satish@gmail.com wrote: can't we make password less? just like MySQL? On Tue, Apr 21, 2015 at 1:56 PM, Podrigal, Aron ar...@guaranteedplus.com wrote: It executes a series of sql files ( I don't remember the dir location where its stored I think somewhere in the share dir) for each file it asks for the password. On Tue, Apr 21, 2015 at 1:44 PM, Satish Patel satish@gmail.com wrote: I am installing opensips 1.11.3 with postgress DB but i don't know what is going on here root@dopensips:/etc/opensips# opensipsdbctl create INFO: creating database opensips ... Password for user postgres: Password for user postgres: Password for user postgres: NOTICE: CREATE TABLE / UNIQUE will create implicit index version_t_name_idx for table version Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence acc_id_seq for serial column acc.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index acc_pkey for table acc NOTICE: CREATE TABLE will create implicit sequence missed_calls_id_seq for serial column missed_calls.id -- Aron Podrigal - //Be happy :-) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Aron Podrigal - //Be happy :-) ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips 1.11 postgress DB BUG
I just did for 1.11 branch, if you want i can do same for 2.1 latest branch On Wed, Apr 22, 2015 at 2:32 PM, Satish Patel satish@gmail.com wrote: Hey Liviu, I have fixed script here is the pull request https://github.com/OpenSIPS/opensips/pull/476 On Wed, Apr 22, 2015 at 8:27 AM, Liviu Chircu li...@opensips.org wrote: Hi Aron/Satish, If you can put together a working Pull Request, I will gladly review it. If bash is not your strong suite, then could you please open a Bug Report [1]? Sooner or later, it will get solved :) [1] https://github.com/OpenSIPS/opensips/issues?q=is%3Aopen+is%3Aissue+label%3Abug Best regards, Liviu Chircu OpenSIPS Developerhttp://www.opensips-solutions.com On 21.04.2015 23:30, Podrigal, Aron wrote: MySQL has a option --password to provide the password on command line. postgres however does not. but here is some refs how you can accomplish what you would like. http://www.postgresql.org/docs/9.4/static/libpq-pgpass.html On Tue, Apr 21, 2015 at 4:26 PM, Satish Patel satish@gmail.com wrote: can't we make password less? just like MySQL? On Tue, Apr 21, 2015 at 1:56 PM, Podrigal, Aron ar...@guaranteedplus.com wrote: It executes a series of sql files ( I don't remember the dir location where its stored I think somewhere in the share dir) for each file it asks for the password. On Tue, Apr 21, 2015 at 1:44 PM, Satish Patel satish@gmail.com wrote: I am installing opensips 1.11.3 with postgress DB but i don't know what is going on here root@dopensips:/etc/opensips# opensipsdbctl create INFO: creating database opensips ... Password for user postgres: Password for user postgres: Password for user postgres: NOTICE: CREATE TABLE / UNIQUE will create implicit index version_t_name_idx for table version Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence acc_id_seq for serial column acc.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index acc_pkey for table acc NOTICE: CREATE TABLE will create implicit sequence missed_calls_id_seq for serial column missed_calls.id -- Aron Podrigal - //Be happy :-) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Aron Podrigal - //Be happy :-) ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] opensips 1.11 postgress DB BUG
I am installing opensips 1.11.3 with postgress DB but i don't know what is going on here root@dopensips:/etc/opensips# opensipsdbctl create INFO: creating database opensips ... Password for user postgres: Password for user postgres: Password for user postgres: NOTICE: CREATE TABLE / UNIQUE will create implicit index version_t_name_idx for table version Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence acc_id_seq for serial column acc.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index acc_pkey for table acc NOTICE: CREATE TABLE will create implicit sequence missed_calls_id_seq for serial column missed_calls.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index missed_calls_pkey for table missed_calls Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence domain_id_seq for serial column domain.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index domain_pkey for table domain NOTICE: CREATE TABLE / UNIQUE will create implicit index domain_domain_idx for table domain Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence grp_id_seq for serial column grp.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index grp_pkey for table grp NOTICE: CREATE TABLE / UNIQUE will create implicit index grp_account_group_idx for table grp NOTICE: CREATE TABLE will create implicit sequence re_grp_id_seq for serial column re_grp.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index re_grp_pkey for table re_grp Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence address_id_seq for serial column address.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index address_pkey for table address Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence aliases_id_seq for serial column aliases.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index aliases_pkey for table aliases NOTICE: CREATE TABLE / UNIQUE will create implicit index aliases_alias_idx for table aliases Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence location_id_seq for serial column location.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index location_pkey for table location NOTICE: CREATE TABLE / UNIQUE will create implicit index location_account_contact_idx for table location Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence silo_id_seq for serial column silo.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index silo_pkey for table silo Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence dbaliases_id_seq for serial column dbaliases.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index dbaliases_pkey for table dbaliases NOTICE: CREATE TABLE / UNIQUE will create implicit index dbaliases_alias_idx for table dbaliases Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence uri_id_seq for serial column uri.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index uri_pkey for table uri NOTICE: CREATE TABLE / UNIQUE will create implicit index uri_account_idx for table uri Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence rtpproxy_sockets_id_seq for serial column rtpproxy_sockets.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index rtpproxy_sockets_pkey for table rtpproxy_sockets Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence speed_dial_id_seq for serial column speed_dial.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index speed_dial_pkey for table speed_dial NOTICE: CREATE TABLE / UNIQUE will create implicit index speed_dial_speed_dial_idx for table speed_dial Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence usr_preferences_id_seq for serial column usr_preferences.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index usr_preferences_pkey for table usr_preferences Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence subscriber_id_seq for serial column subscriber.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index subscriber_pkey for table subscriber NOTICE: CREATE TABLE / UNIQUE will create implicit index subscriber_account_idx for table subscriber Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence pdt_id_seq for serial column pdt.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index pdt_pkey for table pdt NOTICE: CREATE TABLE / UNIQUE will create implicit index pdt_sdomain_prefix_idx for table pdt Password for user postgres: NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index dialog_pkey for table dialog Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence dispatcher_id_seq for serial column dispatcher.id NOTICE: CREATE TABLE / PRIMARY KEY will
[OpenSIPS-Users] does dispatcher use cache?
Does dispatcher module use internal cache? Because we have many group and i don't want opensips hit everytime SQL table to read IP address. Does opensips load dispatcher table in memory to not hit SQL for each call? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips 1.11 postgress DB BUG
can't we make password less? just like MySQL? On Tue, Apr 21, 2015 at 1:56 PM, Podrigal, Aron ar...@guaranteedplus.com wrote: It executes a series of sql files ( I don't remember the dir location where its stored I think somewhere in the share dir) for each file it asks for the password. On Tue, Apr 21, 2015 at 1:44 PM, Satish Patel satish@gmail.com wrote: I am installing opensips 1.11.3 with postgress DB but i don't know what is going on here root@dopensips:/etc/opensips# opensipsdbctl create INFO: creating database opensips ... Password for user postgres: Password for user postgres: Password for user postgres: NOTICE: CREATE TABLE / UNIQUE will create implicit index version_t_name_idx for table version Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence acc_id_seq for serial column acc.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index acc_pkey for table acc NOTICE: CREATE TABLE will create implicit sequence missed_calls_id_seq for serial column missed_calls.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index missed_calls_pkey for table missed_calls Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence domain_id_seq for serial column domain.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index domain_pkey for table domain NOTICE: CREATE TABLE / UNIQUE will create implicit index domain_domain_idx for table domain Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence grp_id_seq for serial column grp.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index grp_pkey for table grp NOTICE: CREATE TABLE / UNIQUE will create implicit index grp_account_group_idx for table grp NOTICE: CREATE TABLE will create implicit sequence re_grp_id_seq for serial column re_grp.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index re_grp_pkey for table re_grp Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence address_id_seq for serial column address.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index address_pkey for table address Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence aliases_id_seq for serial column aliases.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index aliases_pkey for table aliases NOTICE: CREATE TABLE / UNIQUE will create implicit index aliases_alias_idx for table aliases Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence location_id_seq for serial column location.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index location_pkey for table location NOTICE: CREATE TABLE / UNIQUE will create implicit index location_account_contact_idx for table location Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence silo_id_seq for serial column silo.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index silo_pkey for table silo Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence dbaliases_id_seq for serial column dbaliases.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index dbaliases_pkey for table dbaliases NOTICE: CREATE TABLE / UNIQUE will create implicit index dbaliases_alias_idx for table dbaliases Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence uri_id_seq for serial column uri.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index uri_pkey for table uri NOTICE: CREATE TABLE / UNIQUE will create implicit index uri_account_idx for table uri Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence rtpproxy_sockets_id_seq for serial column rtpproxy_sockets.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index rtpproxy_sockets_pkey for table rtpproxy_sockets Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence speed_dial_id_seq for serial column speed_dial.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index speed_dial_pkey for table speed_dial NOTICE: CREATE TABLE / UNIQUE will create implicit index speed_dial_speed_dial_idx for table speed_dial Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence usr_preferences_id_seq for serial column usr_preferences.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index usr_preferences_pkey for table usr_preferences Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence subscriber_id_seq for serial column subscriber.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index subscriber_pkey for table subscriber NOTICE: CREATE TABLE / UNIQUE will create implicit index subscriber_account_idx for table subscriber Password for user postgres: NOTICE: CREATE TABLE will create implicit sequence pdt_id_seq for serial column pdt.id
[OpenSIPS-Users] opensips-cp dispatcher issue
In command line it is showing root@dispatcher1:/var/www# opensipsctl dispatcher dump SET:: 1 URI:: sip:173.XXX.XXX.181 state=Probing But on Opensips-cp web interface it is showing DB state: Active Look like it is not syncing with opensips properly. what would be the issue? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Anybody is interested to develop this scenario using OpenSIPS?
I got confused in your diagram so i just wanted to clean, anyway i think your diagram should be like following [UA]-[Opensips]--[SIP Provider] | | | | | | [Asterisk] [Asterisk] This is what i understand, Please correct me if i am wrong: 1. Asterisk with handle VoiceMail, Conference, IVR etc service. 2. Opensips will send all Outbound calls to PSTN directly. 3. Did you create Asterisk Database for Username etc? I think we may need to Asterisk view table to both can read user DB. Its not a big deal. Regarding hour it may it may take 5/6 hours or may be less or more depend on what kind of issue we see. I will charge overall. 15,000/- Let me know if i am missing anything. On Fri, Apr 17, 2015 at 2:10 AM, Chandramouli P mouli...@gmail.com wrote: Hello All, If anybody is interested to develop this below written scenario using OpenSIPs, please let me know. Global SIP Users --- OpenSIPS --- Asterisk media server1 --- | |-- VoIP provider for PSTN calls | | --- Asterisk media server2 --- *Assumptions:* OpenSIPS public IP address (eth0): 104.131.65.66 OpenSIPS private IP address (eth1): 10.10.10.1 Asterisk media server1 private IP address (eth1): 10.10.10.2 Asterisk media server2 private IP address (eth1): 10.10.10.3 MySQL DB server private IP address (eth1): 10.10.10.4 VoIP provider public ip address: 123.456.789.111 1) All servers are hosted in Digital ocean and in private network 2) All SIP users, voice mail users, dial rules will be stores in MySQL database 3) I must give OpenSIPS proxy server public ip address in to my VoIP provider. My provider will allow incoming/out going traffic through this IP address only. But, call should go through our media servers only. Because, dial rules will be stored in MySQL database. 4) SIP users will connects to openSIPs proxy server from globally 5) I will provide you the whole environment with the installed OpenSIPs (Ubuntu), installed Asterisk (CentOS) servers, and installed MySQL database tables. 6) I will configure Asterisk in real time and data base. *Task:* You need to provide me OpenSIPs working configuration file to fulfill the below needs for the above environment: 1) Nat traversal 2) SIP registrations through proxy (As I said, we store all sip users details in MySQL database table) 3) Load balancing (We will give two media servers) with fail over 4) PSTN inbound/outbound calling through media servers by using MySQL data base tables (Because, we store users dial rules in db table). But, we give our Proxy server ip address to our VoIP provider for authentication purpose. Please do not reply me, if you are a learner. Only experienced professional with OpenSIPS are welcome. Thank you. Chandra. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] lookup(location) extract received field value
Thanks Vlad,
$du did magic, it is extracting Received:: value.
On Tue, Apr 14, 2015 at 11:34 AM, Vlad Paiu vladp...@opensips.org wrote:
Hello,
From you ul show output, you don't have to do anything special for this to
work - OpenSIPS will automatically relay the call to the Received:: value
that's displayed in the ul show output, setting it as $du, while the actual
Request-URI of the message will contain the private Contact that the client
registered.
Best Regards,
Vlad Paiu
OpenSIPS Developerhttp://www.opensips-solutions.com
On 14.04.2015 18:09, Satish Patel wrote:
Hi,
I have following User registred over public IP but that client doesn't
support STUN so contact info showing private IP 192.168.1.6
lookup function default extract Contact:: sip:1001@192.168.1.6:27098
Is there a way i can extract Received:: sip:173.XX.XX.215:27098 so i
can create new URI and send call to that?
if (lookup(location)) {
..
..
}
[root@sip ~]# opensipsctl ul show
Domain:: location table=512 records=1
AOR:: 1...@sip.example.com
Contact::
sip:1001@192.168.1.6:27098;rinstance=e223da1c59d774db Q=
Expires:: 3585
Callid::
NjIyYzg5NzU0NGNlYjFhZTEyMDZlNDk2NTgzMDUzYjY
Cseq:: 2
User-agent:: X-Lite 4.7.1 74247-44615bc7-W6.1
Received:: sip:173.XX.XX.215:27098
State:: CS_SYNC
Flags:: 0
Cflags:: NAT
Socket:: udp:182.XX.XX.164:5060
Methods:: 5951
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[OpenSIPS-Users] lookup(location) extract received field value
Hi,
I have following User registred over public IP but that client doesn't
support STUN so contact info showing private IP 192.168.1.6
lookup function default extract Contact:: sip:1001@192.168.1.6:27098
Is there a way i can extract Received:: sip:173.XX.XX.215:27098 so i
can create new URI and send call to that?
if (lookup(location)) {
..
..
}
[root@sip ~]# opensipsctl ul show
Domain:: location table=512 records=1
AOR:: 1...@sip.example.com
Contact:: sip:1001@192.168.1.6:27098;rinstance=e223da1c59d774db
Q=
Expires:: 3585
Callid:: NjIyYzg5NzU0NGNlYjFhZTEyMDZlNDk2NTgzMDUzYjY
Cseq:: 2
User-agent:: X-Lite 4.7.1 74247-44615bc7-W6.1
Received:: sip:173.XX.XX.215:27098
State:: CS_SYNC
Flags:: 0
Cflags:: NAT
Socket:: udp:182.XX.XX.164:5060
Methods:: 5951
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Re: [OpenSIPS-Users] opensipsctl fifo dlg_list to_uri:: shutdown Opensips
Here is the gdb bt full of crash, please look into this bug... (gdb) bt full #0 core_hash (cmd_tree=value optimized out, param=value optimized out) at ../../hash_func.h:54 end = 0x0 p = 0x0 v = value optimized out h = 0 s2 = 0x0 #1 process_mi_params (cmd_tree=value optimized out, param=value optimized out) at dlg_hash.c:1340 node = value optimized out d_entry = value optimized out dlg = value optimized out p2 = value optimized out h_entry = 8015792 p1 = 0x7f4f49420f18 #2 mi_print_dlgs (cmd_tree=value optimized out, param=value optimized out) at dlg_hash.c:1369 rpl_tree = 0x0 rpl = 0x0 idx = 0 cnt = 0 #3 0x7f4f4635e696 in run_mi_cmd (fifo_stream=0x134e7e0) at ../../mi/mi.h:107 ret = value optimized out #4 mi_fifo_server (fifo_stream=0x134e7e0) at fifo_fnc.c:578 mi_cmd = value optimized out mi_rpl = value optimized out hdl = 0x0 line_len = 29 file_sep = value optimized out command = value optimized out file = value optimized out f = 0x7f4f49413438 reply_stream = 0x136bb20 __FUNCTION__ = mi_fifo_server #5 0x7f4f4635fef6 in fifo_process (rank=value optimized out) at mi_fifo.c:214 fifo_stream = 0x134e7e0 __FUNCTION__ = fifo_process #6 0x004806d1 in start_module_procs () at sr_module.c:765 m = 0x7f4f493e4390 n = 0 l = 0 x = 0 __FUNCTION__ = start_module_procs #7 0x00430ee5 in main_loop () at main.c:714 chd_rank = 0 rc = 32767 startup_done = 0x0 __FUNCTION__ = main_loop #8 0x004336e9 in main (argc=9, argv=0x7fffa88ef3b8) at main.c:1277 cfg_log_stderr = 0 cfg_stream = 0x1319010 ---Type return to continue, or q return to quit--- c = -1 r = 0 tmp = 0x0 tmp_len = 0 port = 5421776 proto = 0 options = 0x5380c8 f:cCm:M:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:o: ret = -1 seed = 2219233510 rfd = 4 __FUNCTION__ = main (gdb) On Mon, Apr 13, 2015 at 11:30 AM, Satish Patel satish@gmail.com wrote: Thanks Jeff for reply, Opensips crash with error child process 6645 exited by a signal 11 I have generated core dump and following is my gdb output. look like i have to recompile it with debug support. How do i compile it with debug support? (gdb) bt full #0 0x7f74f2962ea0 in mi_print_dlgs () from /opt/opensips/lib64/opensips/modules/dialog.so No symbol table info available. #1 0x7f74f518995c in mi_fifo_server () from /opt/opensips/lib64/opensips/modules/mi_fifo.so No symbol table info available. #2 0x7f74f518b440 in ?? () from /opt/opensips/lib64/opensips/modules/mi_fifo.so No symbol table info available. #3 0x0047a9fe in start_module_procs () No symbol table info available. #4 0x00431e67 in main () No symbol table info available. On Mon, Apr 13, 2015 at 8:03 AM, Jeff Pyle jeff.p...@fidelityvoice.com wrote: Satish, What causes you to believe it's killing your opensips? Is opensips running before you run the command, but not after you run the command? If you can verify this is the case, you should follow the relevant steps from the Crash Troubleshooting http://www.opensips.org/Documentation/TroubleShooting-Crash page. Look at the *green* from your error: ERROR: /tmp/opensips_fifo does not exist ERROR: *Make sure you have the line 'modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)' in your config*ERROR: *and also have loaded the mi_fifo module.* Have you done what it says? opensipsctl uses the MI interface http://www.opensips.org/Documentation/Interface-MI-2-1 to manipulate data within opensips. You will need to configure an MI module, such as the mi_fifo http://www.opensips.org/html/docs/modules/2.1.x/mi_fifo.html one mentioned above, to work with opensipsctl. Then, edit opensipsctlrc for your particular environment. Only after all these things are done will opensipsctl produce any output. - Jeff On Sun, Apr 12, 2015 at 3:53 PM, Satish Patel satish@gmail.com wrote: Bump!! Please help -- Sent from my iPhone On Apr 10, 2015, at 7:56 AM, Satish Patel satish@gmail.com wrote: Any thought? Why that command killing my opensips? I think it's a BUG. On production it will be dangerous if it kill service with random command. -- Sent from my iPhone On Apr 9, 2015, at 11:25 PM, Satish Patel satish@gmail.com wrote: Is this a bug? or its normal behavior? It is very dangerous, I was just playing with command and i type following command which kill my opensips process, when i restarted service then it back. I am using opensips 2.1 [root@sip ~]# opensipsctl fifo dlg_list to_uri:: [root@sip ~]# opensipsctl fifo dlg_list ERROR: /tmp/opensips_fifo does
Re: [OpenSIPS-Users] opensipsctl fifo dlg_list to_uri:: shutdown Opensips
Thanks Jeff for reply, Opensips crash with error child process 6645 exited by a signal 11 I have generated core dump and following is my gdb output. look like i have to recompile it with debug support. How do i compile it with debug support? (gdb) bt full #0 0x7f74f2962ea0 in mi_print_dlgs () from /opt/opensips/lib64/opensips/modules/dialog.so No symbol table info available. #1 0x7f74f518995c in mi_fifo_server () from /opt/opensips/lib64/opensips/modules/mi_fifo.so No symbol table info available. #2 0x7f74f518b440 in ?? () from /opt/opensips/lib64/opensips/modules/mi_fifo.so No symbol table info available. #3 0x0047a9fe in start_module_procs () No symbol table info available. #4 0x00431e67 in main () No symbol table info available. On Mon, Apr 13, 2015 at 8:03 AM, Jeff Pyle jeff.p...@fidelityvoice.com wrote: Satish, What causes you to believe it's killing your opensips? Is opensips running before you run the command, but not after you run the command? If you can verify this is the case, you should follow the relevant steps from the Crash Troubleshooting http://www.opensips.org/Documentation/TroubleShooting-Crash page. Look at the *green* from your error: ERROR: /tmp/opensips_fifo does not exist ERROR: *Make sure you have the line 'modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)' in your config*ERROR: *and also have loaded the mi_fifo module.* Have you done what it says? opensipsctl uses the MI interface http://www.opensips.org/Documentation/Interface-MI-2-1 to manipulate data within opensips. You will need to configure an MI module, such as the mi_fifo http://www.opensips.org/html/docs/modules/2.1.x/mi_fifo.html one mentioned above, to work with opensipsctl. Then, edit opensipsctlrc for your particular environment. Only after all these things are done will opensipsctl produce any output. - Jeff On Sun, Apr 12, 2015 at 3:53 PM, Satish Patel satish@gmail.com wrote: Bump!! Please help -- Sent from my iPhone On Apr 10, 2015, at 7:56 AM, Satish Patel satish@gmail.com wrote: Any thought? Why that command killing my opensips? I think it's a BUG. On production it will be dangerous if it kill service with random command. -- Sent from my iPhone On Apr 9, 2015, at 11:25 PM, Satish Patel satish@gmail.com wrote: Is this a bug? or its normal behavior? It is very dangerous, I was just playing with command and i type following command which kill my opensips process, when i restarted service then it back. I am using opensips 2.1 [root@sip ~]# opensipsctl fifo dlg_list to_uri:: [root@sip ~]# opensipsctl fifo dlg_list ERROR: /tmp/opensips_fifo does not exist ERROR: Make sure you have the line 'modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)' in your config ERROR: and also have loaded the mi_fifo module. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensipsctl fifo dlg_list to_uri:: shutdown Opensips
Bump!! Please help -- Sent from my iPhone On Apr 10, 2015, at 7:56 AM, Satish Patel satish@gmail.com wrote: Any thought? Why that command killing my opensips? I think it's a BUG. On production it will be dangerous if it kill service with random command. -- Sent from my iPhone On Apr 9, 2015, at 11:25 PM, Satish Patel satish@gmail.com wrote: Is this a bug? or its normal behavior? It is very dangerous, I was just playing with command and i type following command which kill my opensips process, when i restarted service then it back. I am using opensips 2.1 [root@sip ~]# opensipsctl fifo dlg_list to_uri:: [root@sip ~]# opensipsctl fifo dlg_list ERROR: /tmp/opensips_fifo does not exist ERROR: Make sure you have the line 'modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)' in your config ERROR: and also have loaded the mi_fifo module. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] URGENT: contact header changed between opensips and client
But issue only on this server. My other servers working fine. Why my router not changing contacts of other servers? I have many opensips but only issue with this single server :( one thing i notice, other opensips in different data center. Do you think ISP can do ALG? On Thu, Apr 9, 2015 at 10:39 AM, Newlin, Ben ben.new...@inin.com wrote: Something in between is manipulating the addressing in the SIP message. You said that ALG was disabled in the router, but either that is incorrect or there is some other piece of equipment changing the message. This is clear because the contact is not the only header that is changed. The ‘received’ parameter of the Via address was also changed from the public address of the UA to the private address. Only an entity within the private network could know that address, so it is almost certainly the router doing it. Ben Newlin *From:* users-boun...@lists.opensips.org [mailto: users-boun...@lists.opensips.org] *On Behalf Of *Satish Patel *Sent:* Thursday, April 09, 2015 10:13 AM *To:* OpenSIPS users mailling list *Subject:* Re: [OpenSIPS-Users] URGENT: contact header changed between opensips and client Any thought guys? Why contact address change in transit? is it normal? On Wed, Apr 8, 2015 at 12:12 PM, Satish Patel satish@gmail.com wrote: Very interesting thing going on between opensips and my client [UA] Opensips sending following contact header to Client but on client side i check it received different contact header, who is changing it? [UA]-[Opensips]--[FS] UA - 173.xx.xx.xx.215 Opensips - 182.xx.xx.164:5060 FS - 182.xx.xx.162:5061 My UA behind the NAT and I have ALG disabled on router. * [Opensips] sending following packet SDP OK 200 to [UA] 2015/04/08 07:42:22.820354 182.xx.xx.164:5060 - 173.xx.xx.215:49152 SIP/2.0 200 OK Via: SIP/2.0/UDP 173.xx.xx.215:49152;received=173.xx.xx.215;rport=49152;branch=z9hG4bKPjHB6245E6-SpIsEUs.kPIxxNAFMM0HTDo Record-Route: sip:182.xx.xx.164;lr;ftag=OdSDiepaL9YssVLqOv.3C82QrrsrrdUV;did=3fd.556edf93 CSeq: 722 INVITE Contact: sip:1646327x...@182.xx.xx.162:5061;transport=udp * [UA] Received following packet, look at Contact: it changed here. 162 to 164. U 182.xx.xx.164:5060 - 192.168.1.9:60895 IP/2.0 200 OK. Via: SIP/2.0/UDP 173.xx.xx.215:49152;received=192.168.1.9;rport=49152;branch=z9hG4bKPjHB6245E6-SpIsEUs.kPIxxNAFMM0HTDo. Record-Route: sip:182.xx.xx.164:5060;lr;ftag=OdSDiepaL9YssVLqOv.3C82QrrsrrdUV;did=3fd.556edf93 . CSeq: 722 INVITE. Contact: sip:1646327x...@182.xx.xx.164:5061;transport=udp. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] URGENT: contact header changed between opensips and client
Any thought guys? Why contact address change in transit? is it normal? On Wed, Apr 8, 2015 at 12:12 PM, Satish Patel satish@gmail.com wrote: Very interesting thing going on between opensips and my client [UA] Opensips sending following contact header to Client but on client side i check it received different contact header, who is changing it? [UA]-[Opensips]--[FS] UA - 173.xx.xx.xx.215 Opensips - 182.xx.xx.164:5060 FS - 182.xx.xx.162:5061 My UA behind the NAT and I have ALG disabled on router. * [Opensips] sending following packet SDP OK 200 to [UA] 2015/04/08 07:42:22.820354 182.xx.xx.164:5060 - 173.xx.xx.215:49152 SIP/2.0 200 OK Via: SIP/2.0/UDP 173.xx.xx.215:49152;received=173.xx.xx.215;rport=49152; branch=z9hG4bKPjHB6245E6-SpIsEUs.kPIxxNAFMM0HTDo Record-Route: sip:182.xx.xx.164;lr;ftag=OdSDiepaL9YssVLqOv. 3C82QrrsrrdUV;did=3fd.556edf93 CSeq: 722 INVITE Contact: sip:1646327x...@182.xx.xx.162:5061;transport=udp * [UA] Received following packet, look at Contact: it changed here. 162 to 164. U 182.xx.xx.164:5060 - 192.168.1.9:60895 IP/2.0 200 OK. Via: SIP/2.0/UDP 173.xx.xx.215:49152;received=192.168.1.9;rport=49152; branch=z9hG4bKPjHB6245E6-SpIsEUs.kPIxxNAFMM0HTDo. Record-Route: sip:182.xx.xx.164:5060;lr;ftag=OdSDiepaL9YssVLqOv. 3C82QrrsrrdUV;did=3fd.556edf93. CSeq: 722 INVITE. Contact: sip:1646327x...@182.xx.xx.164:5061;transport=udp. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] URGENT: contact header changed between opensips and client
I am getting correct IP. Public IP of my Server [root@opensips ~]# curl -s checkip.dyndns.org | sed -e 's/.*Current IP Address: //' -e 's/.*$//' 182.xx.xx.164 On Thu, Apr 9, 2015 at 11:42 AM, Gordon E. Sims, Jr. gs...@nexepe.com wrote: You could try running the following command from the OpenSIPS server: curl -s checkip.dyndns.org | sed -e 's/.*Current IP Address: //' -e 's/.*$//' See if its going out with the proper address. More than likely the NAT statement on the router is not setup properly for Reverse NAT Policy. Gordon From: Newlin, Ben ben.new...@inin.com Reply-To: OpenSIPS users mailling list users@lists.opensips.org Date: Thursday, April 9, 2015 at 9:39 AM To: OpenSIPS users mailling list users@lists.opensips.org Subject: Re: [OpenSIPS-Users] URGENT: contact header changed between opensips and client Something in between is manipulating the addressing in the SIP message. You said that ALG was disabled in the router, but either that is incorrect or there is some other piece of equipment changing the message. This is clear because the contact is not the only header that is changed. The ‘received’ parameter of the Via address was also changed from the public address of the UA to the private address. Only an entity within the private network could know that address, so it is almost certainly the router doing it. Ben Newlin *From:* users-boun...@lists.opensips.org [ mailto:users-boun...@lists.opensips.org users-boun...@lists.opensips.org] *On Behalf Of *Satish Patel *Sent:* Thursday, April 09, 2015 10:13 AM *To:* OpenSIPS users mailling list *Subject:* Re: [OpenSIPS-Users] URGENT: contact header changed between opensips and client Any thought guys? Why contact address change in transit? is it normal? On Wed, Apr 8, 2015 at 12:12 PM, Satish Patel satish@gmail.com wrote: Very interesting thing going on between opensips and my client [UA] Opensips sending following contact header to Client but on client side i check it received different contact header, who is changing it? [UA]-[Opensips]--[FS] UA - 173.xx.xx.xx.215 Opensips - 182.xx.xx.164:5060 FS - 182.xx.xx.162:5061 My UA behind the NAT and I have ALG disabled on router. * [Opensips] sending following packet SDP OK 200 to [UA] 2015/04/08 07:42:22.820354 182.xx.xx.164:5060 - 173.xx.xx.215:49152 SIP/2.0 200 OK Via: SIP/2.0/UDP 173.xx.xx.215:49152;received=173.xx.xx.215;rport=49152;branch=z9hG4bKPjHB6245E6-SpIsEUs.kPIxxNAFMM0HTDo Record-Route: sip:182.xx.xx.164;lr;ftag=OdSDiepaL9YssVLqOv.3C82QrrsrrdUV;did=3fd.556edf93 CSeq: 722 INVITE Contact: sip:1646327x...@182.xx.xx.162:5061;transport=udp * [UA] Received following packet, look at Contact: it changed here. 162 to 164. U 182.xx.xx.164:5060 - 192.168.1.9:60895 IP/2.0 200 OK. Via: SIP/2.0/UDP 173.xx.xx.215:49152;received=192.168.1.9;rport=49152;branch=z9hG4bKPjHB6245E6-SpIsEUs.kPIxxNAFMM0HTDo. Record-Route: sip:182.xx.xx.164:5060;lr;ftag=OdSDiepaL9YssVLqOv.3C82QrrsrrdUV;did=3fd.556edf93 . CSeq: 722 INVITE. Contact: sip:1646327x...@182.xx.xx.164:5061;transport=udp. Think before you Print. This e-mail may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply e-mail and delete all copies of this message. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] URGENT: contact header changed between opensips and client
Yes, both Opensips in different location with different ISP. I think its somewhere ISP doing ALG which causing this issue. Even i have tried to change post number instead of default 5060 but still issue is there. I thought ALG only work on default 5060 but i am wrong. Anyway good to know it wasn't my configuration issue. On Thu, Apr 9, 2015 at 1:59 PM, Newlin, Ben ben.new...@inin.com wrote: ISPs can do ALG, but in this case it would be the ISP where the UA is located so the fact that the server is in a different data center wouldn’t matter (unless the UA is also on a different ISP from the others). Is this server using a different address range than the others? Is it possible that some configuration in your router is dependent on the IP ranges from the first data center, so that messages from the new data center don’t get the same treatment? Bottom line though, your traces clearly indicate the message being sent correctly by OpenSIPS and it is being altered in a way that could only occur in the UA’s private network. Ben Newlin *From:* users-boun...@lists.opensips.org [mailto: users-boun...@lists.opensips.org] *On Behalf Of *Satish Patel *Sent:* Thursday, April 09, 2015 10:47 AM *To:* OpenSIPS users mailling list *Subject:* Re: [OpenSIPS-Users] URGENT: contact header changed between opensips and client But issue only on this server. My other servers working fine. Why my router not changing contacts of other servers? I have many opensips but only issue with this single server :( one thing i notice, other opensips in different data center. Do you think ISP can do ALG? On Thu, Apr 9, 2015 at 10:39 AM, Newlin, Ben ben.new...@inin.com wrote: Something in between is manipulating the addressing in the SIP message. You said that ALG was disabled in the router, but either that is incorrect or there is some other piece of equipment changing the message. This is clear because the contact is not the only header that is changed. The ‘received’ parameter of the Via address was also changed from the public address of the UA to the private address. Only an entity within the private network could know that address, so it is almost certainly the router doing it. Ben Newlin *From:* users-boun...@lists.opensips.org [mailto: users-boun...@lists.opensips.org] *On Behalf Of *Satish Patel *Sent:* Thursday, April 09, 2015 10:13 AM *To:* OpenSIPS users mailling list *Subject:* Re: [OpenSIPS-Users] URGENT: contact header changed between opensips and client Any thought guys? Why contact address change in transit? is it normal? On Wed, Apr 8, 2015 at 12:12 PM, Satish Patel satish@gmail.com wrote: Very interesting thing going on between opensips and my client [UA] Opensips sending following contact header to Client but on client side i check it received different contact header, who is changing it? [UA]-[Opensips]--[FS] UA - 173.xx.xx.xx.215 Opensips - 182.xx.xx.164:5060 FS - 182.xx.xx.162:5061 My UA behind the NAT and I have ALG disabled on router. * [Opensips] sending following packet SDP OK 200 to [UA] 2015/04/08 07:42:22.820354 182.xx.xx.164:5060 - 173.xx.xx.215:49152 SIP/2.0 200 OK Via: SIP/2.0/UDP 173.xx.xx.215:49152;received=173.xx.xx.215;rport=49152;branch=z9hG4bKPjHB6245E6-SpIsEUs.kPIxxNAFMM0HTDo Record-Route: sip:182.xx.xx.164;lr;ftag=OdSDiepaL9YssVLqOv.3C82QrrsrrdUV;did=3fd.556edf93 CSeq: 722 INVITE Contact: sip:1646327x...@182.xx.xx.162:5061;transport=udp * [UA] Received following packet, look at Contact: it changed here. 162 to 164. U 182.xx.xx.164:5060 - 192.168.1.9:60895 IP/2.0 200 OK. Via: SIP/2.0/UDP 173.xx.xx.215:49152;received=192.168.1.9;rport=49152;branch=z9hG4bKPjHB6245E6-SpIsEUs.kPIxxNAFMM0HTDo. Record-Route: sip:182.xx.xx.164:5060;lr;ftag=OdSDiepaL9YssVLqOv.3C82QrrsrrdUV;did=3fd.556edf93 . CSeq: 722 INVITE. Contact: sip:1646327x...@182.xx.xx.164:5061;transport=udp. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] URGENT: contact header changed between opensips and client
Very interesting thing going on between opensips and my client [UA] Opensips sending following contact header to Client but on client side i check it received different contact header, who is changing it? [UA]-[Opensips]--[FS] UA - 173.xx.xx.xx.215 Opensips - 182.xx.xx.164:5060 FS - 182.xx.xx.162:5061 My UA behind the NAT and I have ALG disabled on router. * [Opensips] sending following packet SDP OK 200 to [UA] 2015/04/08 07:42:22.820354 182.xx.xx.164:5060 - 173.xx.xx.215:49152 SIP/2.0 200 OK Via: SIP/2.0/UDP 173.xx.xx.215:49152;received=173.xx.xx.215;rport=49152; branch=z9hG4bKPjHB6245E6-SpIsEUs.kPIxxNAFMM0HTDo Record-Route: sip:182.xx.xx.164;lr;ftag=OdSDiepaL9YssVLqOv. 3C82QrrsrrdUV;did=3fd.556edf93 CSeq: 722 INVITE Contact: sip:1646327x...@182.xx.xx.162:5061;transport=udp * [UA] Received following packet, look at Contact: it changed here. 162 to 164. U 182.xx.xx.164:5060 - 192.168.1.9:60895 IP/2.0 200 OK. Via: SIP/2.0/UDP 173.xx.xx.215:49152;received=192.168.1.9;rport=49152; branch=z9hG4bKPjHB6245E6-SpIsEUs.kPIxxNAFMM0HTDo. Record-Route: sip:182.xx.xx.164:5060;lr;ftag=OdSDiepaL9YssVLqOv. 3C82QrrsrrdUV;did=3fd.556edf93. CSeq: 722 INVITE. Contact: sip:1646327x...@182.xx.xx.164:5061;transport=udp. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Regex on R-URI question
I have following R-URI, want to remove sip: and anything after ; sip:12345678@176.74.234.222:31156;rinstance=19e78a48990c0005 I want following from above string 12345678@176.74.234.222:31156 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] variable compare string
Never mind, i figure out
if ($avp(route_info) == 'NULL')
On Tue, Apr 7, 2015 at 11:07 AM, Satish Patel satish@gmail.com wrote:
I have following SQL query
avp_db_query(SELECT billing_customer,route_info FROM inbound WHERE
(did='$tU'),$avp(bparty);$avp(route_info));
I want to check if $avp(route_info) has NULL entry in DB table, But
following code always giving me true. Even DB has value. what is wrong with
code?
if ($avp(route_info) = NULL) {
xlog(Route_info is NULL);
}
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Re: [OpenSIPS-Users] loose_route() sending ACK itself
It is URGENT!!
can some one help? This is very strange issue and i am stuck here :(
loose_route() sending ACK/BYE itself instead of next hope :(
I have removed all entries from domain table but no luck :(
On Thu, Mar 26, 2015 at 12:09 AM, Satish Patel satish@gmail.com wrote:
Hi,
senario:
[UA]-[Opensips]-[Freeswitch]
UA sending correct ACK to freeswitch but Opensips loose_route() sending it
to itself and it break dialog, If use fix_dialog_route() then it works, I
don't have any IP address added in domain table also.
How do i check whether Freeswitch using loose_route for strict route?
I have following script:
if (has_totag()) {
if (loose_route()) {
if (is_method(BYE)) {
#setflag(ACC_DO); # do accounting ...
#setflag(ACC_FAILED); # ... even if the
transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is useless,
do RR for
# re-INVITEs alos, as some buggy clients
do change route set
# during the dialog.
record_route();
}
if (check_route_param(nat=yes))
setflag(NAT);
# route it out to whatever destination was set by
loose_route()
# in $du (destination URI).
route(relay);
} else {
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but stateful
ACK; must be an ACK after
# a 487 or e.g. 404 from upstream
server
xlog(non loose-route section\n);
#t_relay();
exit;
} else {
# ACK without matching transaction
-
# ignore and discard
xlog(ACK without matching
transaction\n);
exit;
}
}
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Re: [OpenSIPS-Users] loose_route() sending ACK itself
Thanks Vlad,
I have sent you screenshot of sip trace to your private address because of
security reason, could you take a look and respond.
On Tue, Apr 7, 2015 at 1:08 PM, Vlad Paiu vladp...@opensips.org wrote:
Hello,
Looking in your SIP trace, I see that in the 200OK Contact, you have
Contact: sip:72.XX.XX.140;did=7de.9accc6f5. , and when OpenSIPS is
routing the ACK out, it is routing it to
U 182.XX.XX.164:5060 - 72.XX.XX.140:5060
ACK sip:72.XX.XX.140;did=7de.9accc6f5 SIP/2.0.
so not sure where exactly is the loop.
Best Regards,
Vlad Paiu
OpenSIPS Developerhttp://www.opensips-solutions.com
On 07.04.2015 19:32, Satish Patel wrote:
It is URGENT!!
can some one help? This is very strange issue and i am stuck here :(
loose_route() sending ACK/BYE itself instead of next hope :(
I have removed all entries from domain table but no luck :(
On Thu, Mar 26, 2015 at 12:09 AM, Satish Patel satish@gmail.com
wrote:
Hi,
senario:
[UA]-[Opensips]-[Freeswitch]
UA sending correct ACK to freeswitch but Opensips loose_route() sending
it to itself and it break dialog, If use fix_dialog_route() then it works,
I don't have any IP address added in domain table also.
How do i check whether Freeswitch using loose_route for strict route?
I have following script:
if (has_totag()) {
if (loose_route()) {
if (is_method(BYE)) {
#setflag(ACC_DO); # do accounting ...
#setflag(ACC_FAILED); # ... even if the
transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is
useless, do RR for
# re-INVITEs alos, as some buggy clients
do change route set
# during the dialog.
record_route();
}
if (check_route_param(nat=yes))
setflag(NAT);
# route it out to whatever destination was set by
loose_route()
# in $du (destination URI).
route(relay);
} else {
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but stateful
ACK; must be an ACK after
# a 487 or e.g. 404 from upstream
server
xlog(non loose-route section\n);
#t_relay();
exit;
} else {
# ACK without matching
transaction -
# ignore and discard
xlog(ACK without matching
transaction\n);
exit;
}
}
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[OpenSIPS-Users] unknown command get_dialog_info, missing loadmodule?
I am using opensips 2.1 and trying to use get_dialog_info I do have dialog module loaded. Did you remove that function in 2.1? loadmodule dialog ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] loose_route() sending ACK itself
Hi,
senario:
[UA]-[Opensips]-[Freeswitch]
UA sending correct ACK to freeswitch but Opensips loose_route() sending it
to itself and it break dialog, If use fix_dialog_route() then it works, I
don't have any IP address added in domain table also.
How do i check whether Freeswitch using loose_route for strict route?
I have following script:
if (has_totag()) {
if (loose_route()) {
if (is_method(BYE)) {
#setflag(ACC_DO); # do accounting ...
#setflag(ACC_FAILED); # ... even if the
transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is useless,
do RR for
# re-INVITEs alos, as some buggy clients do
change route set
# during the dialog.
record_route();
}
if (check_route_param(nat=yes))
setflag(NAT);
# route it out to whatever destination was set by
loose_route()
# in $du (destination URI).
route(relay);
} else {
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but stateful
ACK; must be an ACK after
# a 487 or e.g. 404 from upstream
server
xlog(non loose-route section\n);
#t_relay();
exit;
} else {
# ACK without matching transaction
-
# ignore and discard
xlog(ACK without matching
transaction\n);
exit;
}
}
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] Contest and prices - OpenSIPS 2.1 testing
Wow! so every person will get T-shirt who reported bug or one person among all bug reporter? On Tue, Mar 24, 2015 at 1:06 PM, Răzvan Crainea raz...@opensips.org wrote: Hi, All! Hurry up, we already have three bugs reported[1] :). [1] http://www.opensips.org/Community/BugHuntContest Cheers, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 03/23/2015 08:38 PM, Bogdan-Andrei Iancu wrote: Hi all, Starting this week and all the way to the date of 2.1 stable release (from RC to GA), we will open a weekly contest that will help with the testing :). What is the contest for ? For the best bug found :). Whoever finds the uglies bug in 2.1-rc will get an official OpenSIPS T-shirt, like these guys did :) : http://farm4.staticflickr.com/3947/15725260732_826db90980_z.jpg So, we pay you for for finding the best bug in OpenSIPS - attractive job ?? CONTEST IS ON ! And we already have a strong candidate for this week. Best Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Which port router open
Default SIP port 5060 UDP also you need media port call RTP -- Sent from my iPhone On Mar 19, 2015, at 7:29 AM, Mattia Adducchio m.adducc...@progel.net wrote: Hello Everyone, I'm trying to setup my personal sip server. In this moment it works only in my network, but now I want to open the router port for external access. I have open the port 5060 but maybe it's not enough. Thank you, Mattia ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:dialog:dlg_validate_dialog:
Great! will give it a shot!
Just surprised why it is not matching both dlg and req? does
fix_route_dialog();
has any impact on system when you have very high CPS etc?
It would be good if fix issue from root, instead of external resources
which eat CPU ticks :)
dlg=[sip:16463737221@188.178.235.222:5061;transport=udp] ,
req=[sip:188.178.235.222;lr;ftag=840e2e35;did=1f4.ca6a6956]
On Thu, Mar 19, 2015 at 12:24 PM, Vlad Paiu vladp...@opensips.org wrote:
Hello,
Just to recap, you are saying that the Contact the user agent is sending
is broken and you are happy that OpenSIPS is properly fixing the message,
but you want to get rid of the ERRORs in the log ? If this is the case, you
can use setdebug [1] for this.
Try something like
setdebug(-3)
if ($DLG_status!=NULL !validate_dialog() ) {
xlog( in-dialog bogus request \n);
fix_route_dialog();
}
setdebug()
http://www.opensips.org/Documentation/Script-CoreFunctions-2-1#toc48
Best Regards,
Vlad Paiu
OpenSIPS Developerhttp://www.opensips-solutions.com
On 18.03.2015 22:47, Satish Patel wrote:
I know you guys are super busy in OpenSIPS 2.1 release, but any suggestion
on above issue?
On Wed, Mar 18, 2015 at 12:17 AM, Satish Patel satish@gmail.com
wrote:
I am getting following error in log, I can understand my contact: and
Route: values mismatching here. why it is happening? is there a way to get
rid on this error?
Following is scenario. Only getting error in BYE message.
[UA][OpenSIP]---[Freeswitch]-[Opensip]-[SIP
Provide]
ERROR:dialog:dlg_validate_dialog: failed to validate remote contact:
dlg=[sip:16463737221@188.178.235.222:5061;transport=udp] ,
req=[sip:188.178.235.222;lr;ftag=840e2e35;did=1f4.ca6a6956]
I am using fix_route_dialog() in loose_route()
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route() || match_dialog()) {
if ($DLG_status!=NULL !validate_dialog() ) {
xlog( in-dialog bogus request \n);
fix_route_dialog();
}
xlog(L_INFO, Loose route failed on
$hdr(route)\n);
if (is_method(BYE)) {
#setflag(ACC_DO); # do accounting ...
#setflag(ACC_FAILED); # ... even if the
transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is
useless, do RR for
# re-INVITEs alos, as some buggy clients
do change route set
# during the dialog.
record_route();
}
if (check_route_param(nat=yes))
setflag(NAT);
# route it out to whatever destination was set by
loose_route()
# in $du (destination URI).
route(relay);
} else {
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but stateful
ACK; must be an ACK after
# a 487 or e.g. 404 from upstream
server
xlog(non loose-route section\n);
t_relay();
exit;
} else {
# ACK without matching
transaction -
# ignore and discard
xlog(ACK without matching
transaction\n);
exit;
}
}
xlog(L_INFO, destination uri after
loose_route: $du\n);
sl_send_reply(404,Not here);
}
exit;
}
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Re: [OpenSIPS-Users] Dispatcher user specific route question - 2.1
Thanks Vlad, Superb! so it will do round-robin? or fail-over? On Thu, Mar 19, 2015 at 12:30 PM, Vlad Paiu vladp...@opensips.org wrote: Hello, If you want to do dispatching between multiple setids, ds_select_dst() allows that. See the docs at [1] , you can provide a comma separated list of setids - so your $avp(zone) can contain '1,2' and OpenSIPS will try to first send to the servers in setid 1, and then, if those fail, to the servers in setid 2. [1] http://www.opensips.org/html/docs/modules/1.11.x/dispatcher#id294368 Best Regards, Vlad Paiu OpenSIPS Developerhttp://www.opensips-solutions.com On 19.03.2015 06:17, Satish Patel wrote: I have add extra zone column in subscriber table, +--+-+ | username | zone | +--+-+ |1001 |1| |1002 |2| +--+-+ In dispatcher table I have following two Freeswitch in two groups. +---+-++ | setid | destination | description| +---+--+---+ | 1 | sip:fs1.example.com | Freeswitch-1 | | 2 | sip:fs2.example.com | Freeswitch-2 | +---+--+---+ in opensips.cfg script i am query subscriber table base on incoming username and storing zone in avp(zone) variable, and calling same variable in following code if ( !ds_select_dst($avp(zone), 4, FM10)) Question: now either user belongs to zone 1 or 2, so it is *NOT* going to do load-balancing between two. But if I want to allow some user to do load-balancing then how it will be possible in above scenario? Can i set setid on fly so i can pass request along with user request and set same group for both switch and user call load-balance on both switch? Any other idea? ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dispatcher user specific route question - 2.1
Thanks! for quick answer!! On Thu, Mar 19, 2015 at 12:41 PM, Vlad Paiu vladp...@opensips.org wrote: Hello, It will do fail-over. Best Regards, Vlad Paiu OpenSIPS Developerhttp://www.opensips-solutions.com On 19.03.2015 18:39, Satish Patel wrote: Thanks Vlad, Superb! so it will do round-robin? or fail-over? On Thu, Mar 19, 2015 at 12:30 PM, Vlad Paiu vladp...@opensips.org wrote: Hello, If you want to do dispatching between multiple setids, ds_select_dst() allows that. See the docs at [1] , you can provide a comma separated list of setids - so your $avp(zone) can contain '1,2' and OpenSIPS will try to first send to the servers in setid 1, and then, if those fail, to the servers in setid 2. [1] http://www.opensips.org/html/docs/modules/1.11.x/dispatcher#id294368 Best Regards, Vlad Paiu OpenSIPS Developerhttp://www.opensips-solutions.com On 19.03.2015 06:17, Satish Patel wrote: I have add extra zone column in subscriber table, +--+-+ | username | zone | +--+-+ |1001 |1| |1002 |2| +--+-+ In dispatcher table I have following two Freeswitch in two groups. +---+-++ | setid | destination | description| +---+--+---+ | 1 | sip:fs1.example.com | Freeswitch-1 | | 2 | sip:fs2.example.com | Freeswitch-2 | +---+--+---+ in opensips.cfg script i am query subscriber table base on incoming username and storing zone in avp(zone) variable, and calling same variable in following code if ( !ds_select_dst($avp(zone), 4, FM10)) Question: now either user belongs to zone 1 or 2, so it is *NOT* going to do load-balancing between two. But if I want to allow some user to do load-balancing then how it will be possible in above scenario? Can i set setid on fly so i can pass request along with user request and set same group for both switch and user call load-balance on both switch? Any other idea? ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:dialog:dlg_validate_dialog:
I know you guys are super busy in OpenSIPS 2.1 release, but any suggestion
on above issue?
On Wed, Mar 18, 2015 at 12:17 AM, Satish Patel satish@gmail.com wrote:
I am getting following error in log, I can understand my contact: and
Route: values mismatching here. why it is happening? is there a way to get
rid on this error?
Following is scenario. Only getting error in BYE message.
[UA][OpenSIP]---[Freeswitch]-[Opensip]-[SIP
Provide]
ERROR:dialog:dlg_validate_dialog: failed to validate remote contact:
dlg=[sip:16463737221@188.178.235.222:5061;transport=udp] ,
req=[sip:188.178.235.222;lr;ftag=840e2e35;did=1f4.ca6a6956]
I am using fix_route_dialog() in loose_route()
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route() || match_dialog()) {
if ($DLG_status!=NULL !validate_dialog() ) {
xlog( in-dialog bogus request \n);
fix_route_dialog();
}
xlog(L_INFO, Loose route failed on
$hdr(route)\n);
if (is_method(BYE)) {
#setflag(ACC_DO); # do accounting ...
#setflag(ACC_FAILED); # ... even if the
transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is useless,
do RR for
# re-INVITEs alos, as some buggy clients
do change route set
# during the dialog.
record_route();
}
if (check_route_param(nat=yes))
setflag(NAT);
# route it out to whatever destination was set by
loose_route()
# in $du (destination URI).
route(relay);
} else {
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but stateful
ACK; must be an ACK after
# a 487 or e.g. 404 from upstream
server
xlog(non loose-route section\n);
t_relay();
exit;
} else {
# ACK without matching transaction
-
# ignore and discard
xlog(ACK without matching
transaction\n);
exit;
}
}
xlog(L_INFO, destination uri after loose_route:
$du\n);
sl_send_reply(404,Not here);
}
exit;
}
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[OpenSIPS-Users] Dispatcher user specific route question
I have two Freeswitch in dispatcher table: +---+-+--+ | setid | destination | description | +---+-+--+ | 1 | sip:fs1.example.com | Freeswitch-1 | | 2 | sip:fs2.example.com | Freeswitch-2 | +---+-+--+ I have created zone column in subscriber table. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Dispatcher user specific route question - 2.1
I have add extra zone column in subscriber table, +--+-+ | username | zone | +--+-+ |1001 |1| |1002 |2| +--+-+ In dispatcher table I have following two Freeswitch in two groups. +---+-++ | setid | destination | description| +---+--+---+ | 1 | sip:fs1.example.com | Freeswitch-1 | | 2 | sip:fs2.example.com | Freeswitch-2 | +---+--+---+ in opensips.cfg script i am query subscriber table base on incoming username and storing zone in avp(zone) variable, and calling same variable in following code if ( !ds_select_dst($avp(zone), 4, FM10)) Question: now either user belongs to zone 1 or 2, so it is *NOT* going to do load-balancing between two. But if I want to allow some user to do load-balancing then how it will be possible in above scenario? Can i set setid on fly so i can pass request along with user request and set same group for both switch and user call load-balance on both switch? Any other idea? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] ERROR:dialog:dlg_validate_dialog:
I am getting following error in log, I can understand my contact: and
Route: values mismatching here. why it is happening? is there a way to get
rid on this error?
Following is scenario. Only getting error in BYE message.
[UA][OpenSIP]---[Freeswitch]-[Opensip]-[SIP
Provide]
ERROR:dialog:dlg_validate_dialog: failed to validate remote contact:
dlg=[sip:16463737221@188.178.235.222:5061;transport=udp] ,
req=[sip:188.178.235.222;lr;ftag=840e2e35;did=1f4.ca6a6956]
I am using fix_route_dialog() in loose_route()
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route() || match_dialog()) {
if ($DLG_status!=NULL !validate_dialog() ) {
xlog( in-dialog bogus request \n);
fix_route_dialog();
}
xlog(L_INFO, Loose route failed on
$hdr(route)\n);
if (is_method(BYE)) {
#setflag(ACC_DO); # do accounting ...
#setflag(ACC_FAILED); # ... even if the
transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is useless,
do RR for
# re-INVITEs alos, as some buggy clients do
change route set
# during the dialog.
record_route();
}
if (check_route_param(nat=yes))
setflag(NAT);
# route it out to whatever destination was set by
loose_route()
# in $du (destination URI).
route(relay);
} else {
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but stateful
ACK; must be an ACK after
# a 487 or e.g. 404 from upstream
server
xlog(non loose-route section\n);
t_relay();
exit;
} else {
# ACK without matching transaction
-
# ignore and discard
xlog(ACK without matching
transaction\n);
exit;
}
}
xlog(L_INFO, destination uri after loose_route:
$du\n);
sl_send_reply(404,Not here);
}
exit;
}
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Re: [OpenSIPS-Users] 500 Server error in REGISTER message
I haven't done anything related stateless. also in my config, i haven't manually specify that 500 error anywhere where i can doubt. I don't know from where it is coming. must be internally from opensips. On Tue, Mar 17, 2015 at 1:14 PM, Eric Tamme e...@uphreak.com wrote: Ah - nm, i see it in an sl callback Mar 17 22:19:01 sip2 /usr/local/opensips-2-head/sbin/opensips[31285]: DBG:sl:sl_reply_error: error text is Server error occurred (1/SL) ... so are you doing anything statless in your config? This looks like it might be siptrace related. On 03/17/2015 11:11 AM, Eric Tamme wrote: I do not see the 500 from opensips in this log. On 03/17/2015 11:07 AM, Satish Patel wrote: Here is the debug 4 logs http://pastebin.com/CdPxFrNp 173.48.111.111 - UA 188.79.242.164 - OpenSIPs On Tue, Mar 17, 2015 at 12:45 PM, Eric Tamme e...@uphreak.com wrote: This is a ladder diagram, not a sip trace. A ladder diagram is not useful in this case. Turn your debug up to 4, capture the log of the register/500 happening and submit a link to the pastebin. DO NOT paste the contents into an email. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 500 Server error in REGISTER message
Even after disabled siptrace it is happening. no luck :( On Tue, Mar 17, 2015 at 1:20 PM, Eric Tamme e...@uphreak.com wrote: Turn of your sip tracing and see if the issue occurs. Its running some sl_callbacks (which i assume are realated to siptrace). On 03/17/2015 11:19 AM, Satish Patel wrote: I haven't done anything related stateless. also in my config, i haven't manually specify that 500 error anywhere where i can doubt. I don't know from where it is coming. must be internally from opensips. On Tue, Mar 17, 2015 at 1:14 PM, Eric Tamme e...@uphreak.com wrote: Ah - nm, i see it in an sl callback Mar 17 22:19:01 sip2 /usr/local/opensips-2-head/sbin/opensips[31285]: DBG:sl:sl_reply_error: error text is Server error occurred (1/SL) ... so are you doing anything statless in your config? This looks like it might be siptrace related. On 03/17/2015 11:11 AM, Eric Tamme wrote: I do not see the 500 from opensips in this log. On 03/17/2015 11:07 AM, Satish Patel wrote: Here is the debug 4 logs http://pastebin.com/CdPxFrNp 173.48.111.111 - UA 188.79.242.164 - OpenSIPs On Tue, Mar 17, 2015 at 12:45 PM, Eric Tamme e...@uphreak.com wrote: This is a ladder diagram, not a sip trace. A ladder diagram is not useful in this case. Turn your debug up to 4, capture the log of the register/500 happening and submit a link to the pastebin. DO NOT paste the contents into an email. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 500 Server error in REGISTER message
Great! Guys!! here is my rectified code. look like it works, is that
correct?
if (!save(location)) {
xlog(L_ERR, Saving contact failed - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n);
sl_reply_error();
exit;
}
exit;
}
On Tue, Mar 17, 2015 at 2:13 PM, Eric Tamme e...@uphreak.com wrote:
because the if statment does not evailuate true, so it skips the line
immediately after it. This is how unbraced functions work. it then
continues executing after and sends the error.
On 03/17/2015 12:10 PM, Satish Patel wrote:
Sorry forgot to post link
http://lists.opensips.org/pipermail/users/2012-August/022705.html
also interesting thing, I am not seeing xlog in opensips.log, why?
if ( 0 ) setflag(TCP_PERSISTENT);
if (!save(location))
xlog(Saving contact failed - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n);
sl_reply_error();
exit;
}
On Tue, Mar 17, 2015 at 2:09 PM, Satish Patel satish@gmail.com
wrote:
I have check on book example and it doesn't have any brace also. just
wonder!
Look at this link, someone posted link here, even they don't have curly
brace
On Tue, Mar 17, 2015 at 1:54 PM, Eric Tamme e...@uphreak.com wrote:
You are missing the left curly brace after your if statment im
suprised your script runs at all
On 03/17/2015 11:48 AM, Satish Patel wrote:
Eric,
I found what was the issue, I sent you REGISTER method snippet before,
if you look at it, If remove/comment out sl_reply_error(); line in
following code, it stopped sending 500 Error. Very interesting.. Do you
think i need to put that in curly braces { } ?
if (!save(location))
xlog(L_ERR, Saving contact failed - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n);
sl_reply_error();
exit;
}
On Tue, Mar 17, 2015 at 1:27 PM, Satish Patel satish@gmail.com
wrote:
Even after disabled siptrace it is happening. no luck :(
On Tue, Mar 17, 2015 at 1:20 PM, Eric Tamme e...@uphreak.com wrote:
Turn of your sip tracing and see if the issue occurs. Its running
some sl_callbacks (which i assume are realated to siptrace).
On 03/17/2015 11:19 AM, Satish Patel wrote:
I haven't done anything related stateless. also in my config, i
haven't manually specify that 500 error anywhere where i can doubt. I
don't know from where it is coming. must be internally from opensips.
On Tue, Mar 17, 2015 at 1:14 PM, Eric Tamme e...@uphreak.com wrote:
Ah - nm, i see it in an sl callback
Mar 17 22:19:01 sip2 /usr/local/opensips-2-head/sbin/opensips[31285]:
DBG:sl:sl_reply_error: error text is Server error occurred (1/SL)
... so are you doing anything statless in your config? This looks like
it might be siptrace related.
On 03/17/2015 11:11 AM, Eric Tamme wrote:
I do not see the 500 from opensips in this log.
On 03/17/2015 11:07 AM, Satish Patel wrote:
Here is the debug 4 logs http://pastebin.com/CdPxFrNp
173.48.111.111 - UA
188.79.242.164 - OpenSIPs
On Tue, Mar 17, 2015 at 12:45 PM, Eric Tamme e...@uphreak.com
wrote:
This is a ladder diagram, not a sip trace. A ladder diagram is not
useful in this case.
Turn your debug up to 4, capture the log of the register/500
happening and submit a link to the pastebin. DO NOT paste the contents
into an email.
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Re: [OpenSIPS-Users] 500 Server error in REGISTER message
Eric,
I found what was the issue, I sent you REGISTER method snippet before, if
you look at it, If remove/comment out sl_reply_error(); line in
following code, it stopped sending 500 Error. Very interesting.. Do you
think i need to put that in curly braces { } ?
if (!save(location))
xlog(L_ERR, Saving contact failed - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n);
sl_reply_error();
exit;
}
On Tue, Mar 17, 2015 at 1:27 PM, Satish Patel satish@gmail.com wrote:
Even after disabled siptrace it is happening. no luck :(
On Tue, Mar 17, 2015 at 1:20 PM, Eric Tamme e...@uphreak.com wrote:
Turn of your sip tracing and see if the issue occurs. Its running some
sl_callbacks (which i assume are realated to siptrace).
On 03/17/2015 11:19 AM, Satish Patel wrote:
I haven't done anything related stateless. also in my config, i
haven't manually specify that 500 error anywhere where i can doubt. I
don't know from where it is coming. must be internally from opensips.
On Tue, Mar 17, 2015 at 1:14 PM, Eric Tamme e...@uphreak.com wrote:
Ah - nm, i see it in an sl callback
Mar 17 22:19:01 sip2 /usr/local/opensips-2-head/sbin/opensips[31285]:
DBG:sl:sl_reply_error: error text is Server error occurred (1/SL)
... so are you doing anything statless in your config? This looks like it
might be siptrace related.
On 03/17/2015 11:11 AM, Eric Tamme wrote:
I do not see the 500 from opensips in this log.
On 03/17/2015 11:07 AM, Satish Patel wrote:
Here is the debug 4 logs http://pastebin.com/CdPxFrNp
173.48.111.111 - UA
188.79.242.164 - OpenSIPs
On Tue, Mar 17, 2015 at 12:45 PM, Eric Tamme e...@uphreak.com wrote:
This is a ladder diagram, not a sip trace. A ladder diagram is not
useful in this case.
Turn your debug up to 4, capture the log of the register/500 happening
and submit a link to the pastebin. DO NOT paste the contents into an
email.
___
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___
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___
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Re: [OpenSIPS-Users] 500 Server error in REGISTER message
Here is the debug 4 logs http://pastebin.com/CdPxFrNp 173.48.111.111 - UA 188.79.242.164 - OpenSIPs On Tue, Mar 17, 2015 at 12:45 PM, Eric Tamme e...@uphreak.com wrote: This is a ladder diagram, not a sip trace. A ladder diagram is not useful in this case. Turn your debug up to 4, capture the log of the register/500 happening and submit a link to the pastebin. DO NOT paste the contents into an email. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 500 Server error in REGISTER message
Sorry forgot to post link
http://lists.opensips.org/pipermail/users/2012-August/022705.html
also interesting thing, I am not seeing xlog in opensips.log, why?
if ( 0 ) setflag(TCP_PERSISTENT);
if (!save(location))
xlog(Saving contact failed - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n);
sl_reply_error();
exit;
}
On Tue, Mar 17, 2015 at 2:09 PM, Satish Patel satish@gmail.com wrote:
I have check on book example and it doesn't have any brace also. just
wonder!
Look at this link, someone posted link here, even they don't have curly
brace
On Tue, Mar 17, 2015 at 1:54 PM, Eric Tamme e...@uphreak.com wrote:
You are missing the left curly brace after your if statment im
suprised your script runs at all
On 03/17/2015 11:48 AM, Satish Patel wrote:
Eric,
I found what was the issue, I sent you REGISTER method snippet before,
if you look at it, If remove/comment out sl_reply_error(); line in
following code, it stopped sending 500 Error. Very interesting.. Do you
think i need to put that in curly braces { } ?
if (!save(location))
xlog(L_ERR, Saving contact failed - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n);
sl_reply_error();
exit;
}
On Tue, Mar 17, 2015 at 1:27 PM, Satish Patel satish@gmail.com
wrote:
Even after disabled siptrace it is happening. no luck :(
On Tue, Mar 17, 2015 at 1:20 PM, Eric Tamme e...@uphreak.com wrote:
Turn of your sip tracing and see if the issue occurs. Its running
some sl_callbacks (which i assume are realated to siptrace).
On 03/17/2015 11:19 AM, Satish Patel wrote:
I haven't done anything related stateless. also in my config, i
haven't manually specify that 500 error anywhere where i can doubt. I
don't know from where it is coming. must be internally from opensips.
On Tue, Mar 17, 2015 at 1:14 PM, Eric Tamme e...@uphreak.com wrote:
Ah - nm, i see it in an sl callback
Mar 17 22:19:01 sip2 /usr/local/opensips-2-head/sbin/opensips[31285]:
DBG:sl:sl_reply_error: error text is Server error occurred (1/SL)
... so are you doing anything statless in your config? This looks like
it might be siptrace related.
On 03/17/2015 11:11 AM, Eric Tamme wrote:
I do not see the 500 from opensips in this log.
On 03/17/2015 11:07 AM, Satish Patel wrote:
Here is the debug 4 logs http://pastebin.com/CdPxFrNp
173.48.111.111 - UA
188.79.242.164 - OpenSIPs
On Tue, Mar 17, 2015 at 12:45 PM, Eric Tamme e...@uphreak.com wrote:
This is a ladder diagram, not a sip trace. A ladder diagram is not
useful in this case.
Turn your debug up to 4, capture the log of the register/500
happening and submit a link to the pastebin. DO NOT paste the contents
into an email.
___
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http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
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listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
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Re: [OpenSIPS-Users] 500 Server error in REGISTER message
I got those code from Book Building Telephony System with OpenSIPS 1.6
Here is the code from book
if (is_method(REGISTER))
{
# authenticate the REGISTER requests (uncomment to enable auth)
##if (!www_authorize(, subscriber))
##{
## www_challenge(, 0);
## exit;
##}
##
##if (!db_check_to())
##{
## sl_send_reply(403,Forbidden auth ID);
## exit;
##}
if (!save(location))
sl_reply_error();
exit;
}
}
On Tue, Mar 17, 2015 at 1:48 PM, Satish Patel satish@gmail.com wrote:
Eric,
I found what was the issue, I sent you REGISTER method snippet before, if
you look at it, If remove/comment out sl_reply_error(); line in
following code, it stopped sending 500 Error. Very interesting.. Do you
think i need to put that in curly braces { } ?
if (!save(location))
xlog(L_ERR, Saving contact failed - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n);
sl_reply_error();
exit;
}
On Tue, Mar 17, 2015 at 1:27 PM, Satish Patel satish@gmail.com
wrote:
Even after disabled siptrace it is happening. no luck :(
On Tue, Mar 17, 2015 at 1:20 PM, Eric Tamme e...@uphreak.com wrote:
Turn of your sip tracing and see if the issue occurs. Its running some
sl_callbacks (which i assume are realated to siptrace).
On 03/17/2015 11:19 AM, Satish Patel wrote:
I haven't done anything related stateless. also in my config, i
haven't manually specify that 500 error anywhere where i can doubt. I
don't know from where it is coming. must be internally from opensips.
On Tue, Mar 17, 2015 at 1:14 PM, Eric Tamme e...@uphreak.com wrote:
Ah - nm, i see it in an sl callback
Mar 17 22:19:01 sip2 /usr/local/opensips-2-head/sbin/opensips[31285]:
DBG:sl:sl_reply_error: error text is Server error occurred (1/SL)
... so are you doing anything statless in your config? This looks like it
might be siptrace related.
On 03/17/2015 11:11 AM, Eric Tamme wrote:
I do not see the 500 from opensips in this log.
On 03/17/2015 11:07 AM, Satish Patel wrote:
Here is the debug 4 logs http://pastebin.com/CdPxFrNp
173.48.111.111 - UA
188.79.242.164 - OpenSIPs
On Tue, Mar 17, 2015 at 12:45 PM, Eric Tamme e...@uphreak.com wrote:
This is a ladder diagram, not a sip trace. A ladder diagram is not
useful in this case.
Turn your debug up to 4, capture the log of the register/500 happening
and submit a link to the pastebin. DO NOT paste the contents into an
email.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
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listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
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___
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Re: [OpenSIPS-Users] 500 Server error in REGISTER message
I have check on book example and it doesn't have any brace also. just
wonder!
Look at this link, someone posted link here, even they don't have curly
brace
On Tue, Mar 17, 2015 at 1:54 PM, Eric Tamme e...@uphreak.com wrote:
You are missing the left curly brace after your if statment im
suprised your script runs at all
On 03/17/2015 11:48 AM, Satish Patel wrote:
Eric,
I found what was the issue, I sent you REGISTER method snippet before,
if you look at it, If remove/comment out sl_reply_error(); line in
following code, it stopped sending 500 Error. Very interesting.. Do you
think i need to put that in curly braces { } ?
if (!save(location))
xlog(L_ERR, Saving contact failed - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n);
sl_reply_error();
exit;
}
On Tue, Mar 17, 2015 at 1:27 PM, Satish Patel satish@gmail.com
wrote:
Even after disabled siptrace it is happening. no luck :(
On Tue, Mar 17, 2015 at 1:20 PM, Eric Tamme e...@uphreak.com wrote:
Turn of your sip tracing and see if the issue occurs. Its running some
sl_callbacks (which i assume are realated to siptrace).
On 03/17/2015 11:19 AM, Satish Patel wrote:
I haven't done anything related stateless. also in my config, i
haven't manually specify that 500 error anywhere where i can doubt. I
don't know from where it is coming. must be internally from opensips.
On Tue, Mar 17, 2015 at 1:14 PM, Eric Tamme e...@uphreak.com wrote:
Ah - nm, i see it in an sl callback
Mar 17 22:19:01 sip2 /usr/local/opensips-2-head/sbin/opensips[31285]:
DBG:sl:sl_reply_error: error text is Server error occurred (1/SL)
... so are you doing anything statless in your config? This looks like it
might be siptrace related.
On 03/17/2015 11:11 AM, Eric Tamme wrote:
I do not see the 500 from opensips in this log.
On 03/17/2015 11:07 AM, Satish Patel wrote:
Here is the debug 4 logs http://pastebin.com/CdPxFrNp
173.48.111.111 - UA
188.79.242.164 - OpenSIPs
On Tue, Mar 17, 2015 at 12:45 PM, Eric Tamme e...@uphreak.com wrote:
This is a ladder diagram, not a sip trace. A ladder diagram is not
useful in this case.
Turn your debug up to 4, capture the log of the register/500 happening
and submit a link to the pastebin. DO NOT paste the contents into an
email.
___
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http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
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___
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___
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___
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___
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listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Re: [OpenSIPS-Users] 500 Server error in REGISTER message
I have set debug level 9 but still not seeing any 500 in logs debug=9 log_stderror=no log_facility=LOG_LOCAL7 On Tue, Mar 17, 2015 at 1:11 PM, Eric Tamme e...@uphreak.com wrote: I do not see the 500 from opensips in this log. On 03/17/2015 11:07 AM, Satish Patel wrote: Here is the debug 4 logs http://pastebin.com/CdPxFrNp 173.48.111.111 - UA 188.79.242.164 - OpenSIPs On Tue, Mar 17, 2015 at 12:45 PM, Eric Tamme e...@uphreak.com wrote: This is a ladder diagram, not a sip trace. A ladder diagram is not useful in this case. Turn your debug up to 4, capture the log of the register/500 happening and submit a link to the pastebin. DO NOT paste the contents into an email. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 500 Server error in REGISTER message
Following is trace, OpenSIPs sending 500 to UA ( SIP Phone), Here is the pastebin. http://pastebin.com/UPhNVSGZ [SIP-Phone][OpenSIP Proxy] 11:33:21.331372 │ REGISTER │ ▒ │ ── │ ▒ 11:33:21.331515 │ 401 Unauthorized │ ▒ │ ── │ ▒ 11:33:21.331520 │ 401 Unauthorized │ ▒ │ │ ▒ 11:33:21.434248 │ REGISTER │ ▒ │ ── │ ▒ 11:33:21.434499 │ 200 OK│ ▒ │ ── │ ▒ 11:33:21.434504 │ 200 OK│ ▒ │ │ ▒ 11:33:21.434518 │ 500 Server error occurred │ ▒ │ ── │ ▒ 11:33:21.434520 │ 500 Server error occurred │ ▒ │ │ ▒ 11:33:21.546953 │ REGISTER │ ▒ │ ── │ ▒ 11:33:21.547265 │ 401 Unauthorized │ ▒ │ ── │ ▒ 11:33:21.547267 │ 401 Unauthorized │ ▒ │ │ ▒ 11:33:21.644232 │ REGISTER │ ▒ │ ── │ ▒ 11:33:21.644437 │ 200 OK│ ▒ │ ── │ ▒ 11:33:21.62 │ 200 OK│ ▒ │ │ ▒ 11:33:21.644452 │ 500 Server error occurred │ ▒ │ ── │ ▒ 11:33:21.644455 │ 500 Server error occurred │ ▒ │ │ ▒ 11:33:21.747068 │ REGISTER │ ▒ │ ── │ ▒ 11:33:21.747211 │ 401 Unauthorized │ ▒ │ ── │ ▒ 11:33:21.747215 │ 401 Unauthorized │ ▒ │ │ ▒ 11:33:21.847082 │ REGISTER │ │ │ ── │ │ 11:33:21.847256 │ 200 OK│ │ │ ── │ │ 11:33:21.847261 │ 200 OK│ On Tue, Mar 17, 2015 at 12:26 PM, Terrance Devor ter.de...@gmail.com wrote: Can you please provide a sip trace. Who is sending the 500? Your media server? T ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 500 Server error in REGISTER message
Guys, any suggestion? it is Opensips 2.1.x Master branch. Is it a bug? On Fri, Mar 13, 2015 at 4:03 PM, Satish Patel satish@gmail.com wrote: Hello, Why we are getting 500 error in REGISTER time? Even i got register successfully, is it normal? 01:26:01.078025 │ REGISTER │ │ │ ── │ │ 01:26:01.078032 │ REGISTER │ │ │ │ │ │ 01:26:01.078358 │ 401 Unauthorized │ │ │ │ ── │ │ │ 01:26:01.078360 │ 401 Unauthorized │ │ ▒ │ │ │ ▒ 01:26:01.408019 │ REGISTER │ │ ▒ │ ── │ │ ▒ 01:26:01.408025 │ REGISTER │ │ ▒ │ │ │ ▒ 01:26:01.408364 │ 200 OK│ │ ▒ │ ── │ │ ▒ 01:26:01.408365 │ 200 OK│ │ ▒ │ │ │ ▒ 01:26:01.408381 │ 500 Server error occurred │ │ ▒ │ ── │ │ ▒ 01:26:01.408382 │ 500 Server error occurred │ │ ▒ │ │ │ ▒ 01:26:01.730160 │ REGISTER │ │ ▒ │ ── │ │ ▒ 01:26:01.730166 │ REGISTER │ │ ▒ │ │ │ ▒ 01:26:01.730454 │ 401 Unauthorized │ │ ▒ │ ── │ │ ▒ 01:26:01.730455 │ 401 Unauthorized │ │ ▒ │ │ │ ▒ 01:26:02.072783 │ REGISTER │ │ ▒ │ ── │ │ ▒ 01:26:02.072789 │ REGISTER │ │ ▒ │ │ │ ▒ 01:26:02.073108 │ 200 OK│ │ ▒ │ ── │ │ ▒ 01:26:02.073109 │ 200 OK│ │ ▒ │ │ │ ▒ 01:26:02.073128 │ 500 Server error occurred │ │ ▒ │ ── │ │ │ │ │ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 500 Server error in REGISTER message
Thanks for reply, I am not seeing any memory error in logs, This is testing box, so there is no load on CPU, I am registering single client. I am using multi-domain with mysql DB. This is what in debug i am seeing DBG:auth:check_response: authorization is OK ... DBG:core:mk_proxy: doing DNS lookup... DBG:sl:run_sl_callbacks: callback id 0 entered DBG:sl:sl_reply_error: error text is Server error occurred (1/SL) On Fri, Mar 13, 2015 at 4:16 PM, Babil Golam Sarwar gsba...@gmail.com wrote: Hi Satish, you should check the server logs. This could be caused by memory allocation failure, low memory and/or high CPU activity experienced by OpenSIPS on the server side. You can enable console debugging by adding the following to `opensips.cfg': debug=4 fork=no log_stderror=yes After that, restart OpenSIPS and watch the debug log output and system CPU usage. Regards, Babil (Golam Sarwar) PGP Key Fingerprint : D3A1 EED0 5BA0 72D3 A011 75CB 8EA6 7D99 F433 E92D PGP Key Download URL: http://bit.ly/gsbabil-pgp-key On Fri, Mar 13, 2015 at 1:03 PM, Satish Patel satish@gmail.com wrote: Hello, Why we are getting 500 error in REGISTER time? Even i got register successfully, is it normal? 01:26:01.078025 │ REGISTER │ │ │ ── │ │ 01:26:01.078032 │ REGISTER │ │ │ │ │ │ 01:26:01.078358 │ 401 Unauthorized │ │ │ │ ── │ │ │ 01:26:01.078360 │ 401 Unauthorized │ │ ▒ │ │ │ ▒ 01:26:01.408019 │ REGISTER │ │ ▒ │ ── │ │ ▒ 01:26:01.408025 │ REGISTER │ │ ▒ │ │ │ ▒ 01:26:01.408364 │ 200 OK│ │ ▒ │ ── │ │ ▒ 01:26:01.408365 │ 200 OK│ │ ▒ │ │ │ ▒ 01:26:01.408381 │ 500 Server error occurred │ │ ▒ │ ── │ │ ▒ 01:26:01.408382 │ 500 Server error occurred │ │ ▒ │ │ │ ▒ 01:26:01.730160 │ REGISTER │ │ ▒ │ ── │ │ ▒ 01:26:01.730166 │ REGISTER │ │ ▒ │ │ │ ▒ 01:26:01.730454 │ 401 Unauthorized │ │ ▒ │ ── │ │ ▒ 01:26:01.730455 │ 401 Unauthorized │ │ ▒ │ │ │ ▒ 01:26:02.072783 │ REGISTER │ │ ▒ │ ── │ │ ▒ 01:26:02.072789 │ REGISTER │ │ ▒ │ │ │ ▒ 01:26:02.073108 │ 200 OK│ │ ▒ │ ── │ │ ▒ 01:26:02.073109 │ 200 OK│ │ ▒ │ │ │ ▒ 01:26:02.073128 │ 500 Server error occurred │ │ ▒ │ ── │ │ │ │ │ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Planning OpenSIPS 2.1 release - heads up
Fetch MASTER 2.1.x and after compile i try to run and got this error, [root@opensips ]# /usr/local/opensips-2-head/sbin/opensips -c -f opensips.cfg Mar 13 01:03:43 [22204] CRITICAL:core:yyerror: parse error in config file opensips.cfg, line 60, column 1-12: syntax error Mar 13 01:03:43 [22204] CRITICAL:core:yyerror: parse error in config file opensips.cfg, line 60, column 1-12: Mar 13 01:03:43 [22204] ERROR:core:main: bad config file (2 errors) It is not supporting following option, after removing them it parse file successfully. disable_tcp=yes disable_tls=yes On Thu, Mar 12, 2015 at 3:16 PM, Satish Patel satish@gmail.com wrote: On download page, http://www.opensips.org/Downloads/Downloads, currently we have following. GIT clone of development stable version 2.1.1 (MASTER): Does on march 18th release will be 2.1.2 ? On Thu, Mar 12, 2015 at 11:26 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote: The D day for releasing 2.1 is 18th of March ! Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 24.02.2015 16:22, Bogdan-Andrei Iancu wrote: Hello all, Everybody is looking forward for the next OpenSIPS release, the major 2.1, which is planned for end of February - at least this was the plan so far. I our desire to make of 2.1 a radical improvement, we committed to several major changes/redesigns and new valuable functionalities. And they do burn time to get them done, especially as we want them : in the best possible way. In oder to finish all we committed for, the release date is now estimated for first half of March - some important parts of code still need to be settled down and we do not want to make any kind of compromise in quality. Just to give you a heads up on what OpenSIPS 2.1 will bring: - internal re-design around async reactors for load-balancing tasks inside OpenSIPS - async I/O support in scripts for db queries, exec, rest client - refactoring of the networking and protocol related code (protos are now modules) - webSockets support - Quality Based Routing new module - Compression new module - Fraud Detection new module - Emergency Call handling new module - rtpengine support - partitioning support in Dynamic Routing, Diaplan and Dispatcher and many other - when the coding taks is done, we will proceed withe bothersome task of updating docs, new and migration. Whoever gives a try to 2.1 version, please report any problems or crashes asap to us ! better have them done now rather than later :P Best regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Planning OpenSIPS 2.1 release - heads up
On download page, http://www.opensips.org/Downloads/Downloads, currently we have following. GIT clone of development stable version 2.1.1 (MASTER): Does on march 18th release will be 2.1.2 ? On Thu, Mar 12, 2015 at 11:26 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote: The D day for releasing 2.1 is 18th of March ! Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 24.02.2015 16:22, Bogdan-Andrei Iancu wrote: Hello all, Everybody is looking forward for the next OpenSIPS release, the major 2.1, which is planned for end of February - at least this was the plan so far. I our desire to make of 2.1 a radical improvement, we committed to several major changes/redesigns and new valuable functionalities. And they do burn time to get them done, especially as we want them : in the best possible way. In oder to finish all we committed for, the release date is now estimated for first half of March - some important parts of code still need to be settled down and we do not want to make any kind of compromise in quality. Just to give you a heads up on what OpenSIPS 2.1 will bring: - internal re-design around async reactors for load-balancing tasks inside OpenSIPS - async I/O support in scripts for db queries, exec, rest client - refactoring of the networking and protocol related code (protos are now modules) - webSockets support - Quality Based Routing new module - Compression new module - Fraud Detection new module - Emergency Call handling new module - rtpengine support - partitioning support in Dynamic Routing, Diaplan and Dispatcher and many other - when the coding taks is done, we will proceed withe bothersome task of updating docs, new and migration. Whoever gives a try to 2.1 version, please report any problems or crashes asap to us ! better have them done now rather than later :P Best regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:siptrace:pipport2su: bad protocol
I set debug=6 and here is the logs most of time error pop'ed up near tm module DBG:tm:run_trans_callbacks: trans=0x7f9570f06710, callback type 1024, id 1 entered DBG:siptrace:trace_onreq_out: trace on req out DBG:core:parse_headers: flags=40 DBG:siptrace:trace_msg_out: trace msg out ERROR:siptrace:pipport2su: bad protocol On Tue, Mar 10, 2015 at 12:12 PM, Satish Patel satish@gmail.com wrote: I am configuring siptrace with homer server and getting following error, ERROR:siptrace:pipport2su: bad protocol ERROR:siptrace:pipport2su: bad protocol ERROR:siptrace:pipport2su: bad protocol Razvan fix following patch but still getting above error in log https://github.com/OpenSIPS/opensips/commit/178b0cc26b05a81947de150fe1c2df36d600ccaa Am i missing something? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] ERROR:siptrace:pipport2su: bad protocol
I am configuring siptrace with homer server and getting following error, ERROR:siptrace:pipport2su: bad protocol ERROR:siptrace:pipport2su: bad protocol ERROR:siptrace:pipport2su: bad protocol Razvan fix following patch but still getting above error in log https://github.com/OpenSIPS/opensips/commit/178b0cc26b05a81947de150fe1c2df36d600ccaa Am i missing something? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] URGENT: ERROR:siptrace:pipport2su: bad protocol udp
Superb, definitely going to give a try, I have a silly question. Can I apply
that patch manually on my beach because if I try new master then fires it
require any change in mysql?
Reason we have couple of box running 2.1.x so want to keep it same. But any way
let me test code first. And get back to you.
--
Sent from my iPhone
On Mar 9, 2015, at 5:03 AM, Răzvan Crainea raz...@opensips.org wrote:
Hi, Satish!
Indeed, there was an issue with the protocol check. I fixed it in the master
branch. Please give it another try and let me know if you still have problems.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 03/08/2015 09:31 PM, Satish Patel wrote:
I got your point, but our plan is to use 2.1.x and we are already using it
since last 6 month without issue.
But it should work in 2.1.x right?
On Sun, Mar 8, 2015 at 3:20 PM, shah...@voip-demos.com wrote:
OpenSIPS v2.x is much different then v1.x. So, if you are not familiar with
it, you should better use 1.x
Thank you.
On 2015-03-08 19:52, Satish Patel wrote:
I tried same configuration on 1.11 version and it works! so look like
something wrong in 2.1.x version please fix that bug as soon as possible
On Sun, Mar 8, 2015 at 2:04 PM, Satish Patel satish@gmail.com wrote:
sorry for push but it wired error!
I have configure siptrace to send packet to Homer but getting following
error in logs
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
Opensips 2.1.x
SIP Capture agent
loadmodule siptrace.so
modparam(siptrace, duplicate_uri, sip:192.168.1.200:9060)
modparam(siptrace, duplicate_with_hep, 1)
modparam(siptrace, trace_to_database, 0)
modparam(siptrace, trace_flag, 22)
modparam(siptrace, trace_on, 1)
#HEPv2 == timestamp will be included to HEP header
modparam(siptrace, hep_version, 1)
modparam(siptrace, db_url,
mysql://opensips:xx@localhost/opensips)
...
...
route{
setflag(22);
sip_trace();
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Re: [OpenSIPS-Users] URGENT: ERROR:siptrace:pipport2su: bad protocol udp
Sorry It was branch
My iPhone is over smart :(
--
Sent from my iPhone
On Mar 9, 2015, at 9:12 AM, Satish Patel satish@gmail.com wrote:
Superb, definitely going to give a try, I have a silly question. Can I apply
that patch manually on my beach because if I try new master then fires it
require any change in mysql?
Reason we have couple of box running 2.1.x so want to keep it same. But any
way let me test code first. And get back to you.
--
Sent from my iPhone
On Mar 9, 2015, at 5:03 AM, Răzvan Crainea raz...@opensips.org wrote:
Hi, Satish!
Indeed, there was an issue with the protocol check. I fixed it in the master
branch. Please give it another try and let me know if you still have
problems.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 03/08/2015 09:31 PM, Satish Patel wrote:
I got your point, but our plan is to use 2.1.x and we are already using it
since last 6 month without issue.
But it should work in 2.1.x right?
On Sun, Mar 8, 2015 at 3:20 PM, shah...@voip-demos.com wrote:
OpenSIPS v2.x is much different then v1.x. So, if you are not familiar
with it, you should better use 1.x
Thank you.
On 2015-03-08 19:52, Satish Patel wrote:
I tried same configuration on 1.11 version and it works! so look like
something wrong in 2.1.x version please fix that bug as soon as possible
On Sun, Mar 8, 2015 at 2:04 PM, Satish Patel satish@gmail.com
wrote:
sorry for push but it wired error!
I have configure siptrace to send packet to Homer but getting
following error in logs
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
Opensips 2.1.x
SIP Capture agent
loadmodule siptrace.so
modparam(siptrace, duplicate_uri, sip:192.168.1.200:9060)
modparam(siptrace, duplicate_with_hep, 1)
modparam(siptrace, trace_to_database, 0)
modparam(siptrace, trace_flag, 22)
modparam(siptrace, trace_on, 1)
#HEPv2 == timestamp will be included
to HEP header
modparam(siptrace, hep_version, 1)
modparam(siptrace, db_url,
mysql://opensips:xx@localhost/opensips)
...
...
route{
setflag(22);
sip_trace();
___
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___
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___
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Re: [OpenSIPS-Users] URGENT: ERROR:siptrace:pipport2su: bad protocol udp
Thanks Razvan,
It is working great!! you guys are awesome!
On Mon, Mar 9, 2015 at 9:43 AM, Satish Patel satish@gmail.com wrote:
Sorry It was branch
My iPhone is over smart :(
--
Sent from my iPhone
On Mar 9, 2015, at 9:12 AM, Satish Patel satish@gmail.com wrote:
Superb, definitely going to give a try, I have a silly question. Can I
apply that patch manually on my beach because if I try new master then
fires it require any change in mysql?
Reason we have couple of box running 2.1.x so want to keep it same. But
any way let me test code first. And get back to you.
--
Sent from my iPhone
On Mar 9, 2015, at 5:03 AM, Răzvan Crainea raz...@opensips.org wrote:
Hi, Satish!
Indeed, there was an issue with the protocol check. I fixed it in the
master branch. Please give it another try and let me know if you still have
problems.
Best regards,
Răzvan Crainea
OpenSIPS Solutionswww.opensips-solutions.com
On 03/08/2015 09:31 PM, Satish Patel wrote:
I got your point, but our plan is to use 2.1.x and we are already using
it since last 6 month without issue.
But it should work in 2.1.x right?
On Sun, Mar 8, 2015 at 3:20 PM, shah...@voip-demos.com wrote:
OpenSIPS v2.x is much different then v1.x. So, if you are not familiar
with it, you should better use 1.x
Thank you.
On 2015-03-08 19:52, Satish Patel wrote:
I tried same configuration on 1.11 version and it works! so look like
something wrong in 2.1.x version please fix that bug as soon as possible
On Sun, Mar 8, 2015 at 2:04 PM, Satish Patel satish@gmail.com
wrote:
sorry for push but it wired error!
I have configure siptrace to send packet to Homer but getting
following error in logs
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
Opensips 2.1.x
SIP Capture agent
loadmodule siptrace.so
modparam(siptrace, duplicate_uri, sip:192.168.1.200:9060)
modparam(siptrace, duplicate_with_hep, 1)
modparam(siptrace, trace_to_database, 0)
modparam(siptrace, trace_flag, 22)
modparam(siptrace, trace_on, 1)
#HEPv2 == timestamp will be included to HEP header
modparam(siptrace, hep_version, 1)
modparam(siptrace, db_url, mysql://opensips:xx@localhost
/opensips)
...
...
route{
setflag(22);
sip_trace();
___
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___
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___
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Re: [OpenSIPS-Users] URGENT: ERROR:siptrace:pipport2su: bad protocol udp
Hey Razvan,
Can i take following patch and directly apply to my existing install branch
instead of downloading new Master?
https://github.com/OpenSIPS/opensips/commit/178b0cc26b05a81947de150fe1c2df36d600ccaa
Currently i am running:
[root@sip ]# opensips -V
version: opensips 2.1.1dev-tls (x86_64/linux)
flags: STATS: On, USE_IPV6, USE_TCP, USE_TLS, 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.
git revision: b3beb20
main.c compiled on 11:44:56 Dec 31 2014 with gcc 4.4.7
On Mon, Mar 9, 2015 at 10:39 AM, Satish Patel satish@gmail.com wrote:
Thanks Razvan,
It is working great!! you guys are awesome!
On Mon, Mar 9, 2015 at 9:43 AM, Satish Patel satish@gmail.com wrote:
Sorry It was branch
My iPhone is over smart :(
--
Sent from my iPhone
On Mar 9, 2015, at 9:12 AM, Satish Patel satish@gmail.com wrote:
Superb, definitely going to give a try, I have a silly question. Can I
apply that patch manually on my beach because if I try new master then
fires it require any change in mysql?
Reason we have couple of box running 2.1.x so want to keep it same. But
any way let me test code first. And get back to you.
--
Sent from my iPhone
On Mar 9, 2015, at 5:03 AM, Răzvan Crainea raz...@opensips.org wrote:
Hi, Satish!
Indeed, there was an issue with the protocol check. I fixed it in the
master branch. Please give it another try and let me know if you still have
problems.
Best regards,
Răzvan Crainea
OpenSIPS Solutionswww.opensips-solutions.com
On 03/08/2015 09:31 PM, Satish Patel wrote:
I got your point, but our plan is to use 2.1.x and we are already using
it since last 6 month without issue.
But it should work in 2.1.x right?
On Sun, Mar 8, 2015 at 3:20 PM, shah...@voip-demos.com wrote:
OpenSIPS v2.x is much different then v1.x. So, if you are not familiar
with it, you should better use 1.x
Thank you.
On 2015-03-08 19:52, Satish Patel wrote:
I tried same configuration on 1.11 version and it works! so look like
something wrong in 2.1.x version please fix that bug as soon as possible
On Sun, Mar 8, 2015 at 2:04 PM, Satish Patel satish@gmail.com
wrote:
sorry for push but it wired error!
I have configure siptrace to send packet to Homer but getting
following error in logs
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
Opensips 2.1.x
SIP Capture agent
loadmodule siptrace.so
modparam(siptrace, duplicate_uri, sip:192.168.1.200:9060)
modparam(siptrace, duplicate_with_hep, 1)
modparam(siptrace, trace_to_database, 0)
modparam(siptrace, trace_flag, 22)
modparam(siptrace, trace_on, 1)
#HEPv2 == timestamp will be included to HEP header
modparam(siptrace, hep_version, 1)
modparam(siptrace, db_url, mysql://opensips:xx@localhost
/opensips)
...
...
route{
setflag(22);
sip_trace();
___
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___
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___
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Re: [OpenSIPS-Users] URGENT: ERROR:siptrace:pipport2su: bad protocol udp
Thanks, I did exactly whatever you told me, i just patch 5 line in
siptrace.c file in my current branch (2.1.1dev-tls git revision: b3beb20)
Now it is working but still i am getting following error in logs. ( it is
not saying UDP this time)
ERROR:siptrace:pipport2su: bad protocol
ERROR:siptrace:pipport2su: bad protocol
And Interesting thing, I have Homer running on other box it is getting
following error. Look like it is related to above issue. could you please
take a look
ERROR: sipcapture [hep.c:139]: hepv2_received(): ERROR:
sipcapture:hep_msg_received: unknow protocol [1]
ERROR: sipcapture [hep.c:139]: hepv2_received(): ERROR:
sipcapture:hep_msg_received: unknow protocol [1]
On Mon, Mar 9, 2015 at 1:04 PM, Răzvan Crainea raz...@opensips.org wrote:
Hi, Satish!
Not sure what's the order of your messages, but yes, you can apply the
patch directly on your branch, without cloning the entire Master.
Best regards,
Răzvan Crainea
OpenSIPS Solutionswww.opensips-solutions.com
On 03/09/2015 05:43 PM, Satish Patel wrote:
Hey Razvan,
Can i take following patch and directly apply to my existing install
branch instead of downloading new Master?
https://github.com/OpenSIPS/opensips/commit/178b0cc26b05a81947de150fe1c2df36d600ccaa
Currently i am running:
[root@sip ]# opensips -V
version: opensips 2.1.1dev-tls (x86_64/linux)
flags: STATS: On, USE_IPV6, USE_TCP, USE_TLS, 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.
git revision: b3beb20
main.c compiled on 11:44:56 Dec 31 2014 with gcc 4.4.7
On Mon, Mar 9, 2015 at 10:39 AM, Satish Patel satish@gmail.com
wrote:
Thanks Razvan,
It is working great!! you guys are awesome!
On Mon, Mar 9, 2015 at 9:43 AM, Satish Patel satish@gmail.com
wrote:
Sorry It was branch
My iPhone is over smart :(
--
Sent from my iPhone
On Mar 9, 2015, at 9:12 AM, Satish Patel satish@gmail.com wrote:
Superb, definitely going to give a try, I have a silly question. Can
I apply that patch manually on my beach because if I try new master then
fires it require any change in mysql?
Reason we have couple of box running 2.1.x so want to keep it same.
But any way let me test code first. And get back to you.
--
Sent from my iPhone
On Mar 9, 2015, at 5:03 AM, Răzvan Crainea raz...@opensips.org wrote:
Hi, Satish!
Indeed, there was an issue with the protocol check. I fixed it in the
master branch. Please give it another try and let me know if you still have
problems.
Best regards,
Răzvan Crainea
OpenSIPS Solutionswww.opensips-solutions.com
On 03/08/2015 09:31 PM, Satish Patel wrote:
I got your point, but our plan is to use 2.1.x and we are already using
it since last 6 month without issue.
But it should work in 2.1.x right?
On Sun, Mar 8, 2015 at 3:20 PM, shah...@voip-demos.com wrote:
OpenSIPS v2.x is much different then v1.x. So, if you are not
familiar with it, you should better use 1.x
Thank you.
On 2015-03-08 19:52, Satish Patel wrote:
I tried same configuration on 1.11 version and it works! so look like
something wrong in 2.1.x version please fix that bug as soon as possible
On Sun, Mar 8, 2015 at 2:04 PM, Satish Patel satish@gmail.com
wrote:
sorry for push but it wired error!
I have configure siptrace to send packet to Homer but getting
following error in logs
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
Opensips 2.1.x
SIP Capture agent
loadmodule siptrace.so
modparam(siptrace, duplicate_uri, sip:192.168.1.200:9060)
modparam(siptrace, duplicate_with_hep, 1)
modparam(siptrace, trace_to_database, 0)
modparam(siptrace, trace_flag, 22)
modparam(siptrace, trace_on, 1)
#HEPv2 == timestamp will be included to HEP header
modparam(siptrace, hep_version, 1)
modparam(siptrace, db_url, mysql://opensips:xx@localhost
/opensips)
...
...
route{
setflag(22);
sip_trace();
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[OpenSIPS-Users] URGENT: ERROR:siptrace:pipport2su: bad protocol udp
sorry for push but it wired error!
I have configure siptrace to send packet to Homer but getting following
error in logs
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
Opensips 2.1.x
SIP Capture agent
loadmodule siptrace.so
modparam(siptrace, duplicate_uri, sip:192.168.1.200:9060)
modparam(siptrace, duplicate_with_hep, 1)
modparam(siptrace, trace_to_database, 0)
modparam(siptrace, trace_flag, 22)
modparam(siptrace, trace_on, 1)
#HEPv2 == timestamp will be included to HEP header
modparam(siptrace, hep_version, 1)
modparam(siptrace, db_url, mysql://opensips:xx@localhost/opensips)
...
...
route{
setflag(22);
sip_trace();
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Re: [OpenSIPS-Users] URGENT: ERROR:siptrace:pipport2su: bad protocol udp
I got your point, but our plan is to use 2.1.x and we are already using it
since last 6 month without issue.
But it should work in 2.1.x right?
On Sun, Mar 8, 2015 at 3:20 PM, shah...@voip-demos.com wrote:
OpenSIPS v2.x is much different then v1.x. So, if you are not familiar
with it, you should better use 1.x
Thank you.
On 2015-03-08 19:52, Satish Patel wrote:
I tried same configuration on 1.11 version and it works! so look like
something wrong in 2.1.x version please fix that bug as soon as possible
On Sun, Mar 8, 2015 at 2:04 PM, Satish Patel satish@gmail.com wrote:
sorry for push but it wired error!
I have configure siptrace to send packet to Homer but getting following
error in logs
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
Opensips 2.1.x
SIP Capture agent
loadmodule siptrace.so
modparam(siptrace, duplicate_uri, sip:192.168.1.200:9060)
modparam(siptrace, duplicate_with_hep, 1)
modparam(siptrace, trace_to_database, 0)
modparam(siptrace, trace_flag, 22)
modparam(siptrace, trace_on, 1)
#HEPv2 == timestamp will be included to HEP header
modparam(siptrace, hep_version, 1)
modparam(siptrace, db_url, mysql://opensips:xx@localhost
/opensips)
...
...
route{
setflag(22);
sip_trace();
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Re: [OpenSIPS-Users] Version question
We have upgraded 1.12.x to 2.1.x and its been 5 month no issue so far, everything works! i am waiting for 2.1.x stable release so i can push it out but i would say so far its good and stable. Just question to Liviu, How do i use latest 2.1.x feature? currently i am using 1.x config. but i would like to use latest feature of 2.1.x On Sun, Mar 8, 2015 at 1:12 PM, Liviu Chircu li...@opensips.org wrote: Syntax-wise, these are the only major changes in 2.1: * for udp/tcp/sctp/tls listeners, you must also do a `loadmodule proto_xxx.so` command after setting your mpath. * the global tcp_xxx and tls_xxx params must now be rewritten as modparam(proto_proto, proto_, ...) Now, from a stability point of view, the network layer has been completely re-organized in order to easily support the addition of new transport protocols. This code is currently in a beta phase - it needs testing. But modules are unchanged and should behave exactly the same. So yes, it should all work in general :) Liviu Chircu OpenSIPS Developerhttp://www.opensips-solutions.com On 08.03.2015 18:43, John Nash wrote: OK. Thank you. I have one more related question. If I use 2.1 version (I understand there may be some bugs) and use it with 1.X scripts (I mean without using async features in script for now) should it all work in general?...Like all modules? On Sun, Mar 8, 2015 at 9:52 PM, Liviu Chircu li...@opensips.org wrote: Hello John, 1.11.3 LTS seems to be what you're looking for [1]. You can get it with: git clone https://github.com/OpenSIPS/opensips.git -b 1.11 [1] http://www.opensips.org/About/AvailableVersions Best regards, Liviu Chircu OpenSIPS Developerhttp://www.opensips-solutions.com On 08.03.2015 17:59, John Nash wrote: I had started testing 1.X series taken from github master branch couple of months ago (It shows version as Server:: OpenSIPS (1.12.0dev-notls (x86_64/linux)) Now I need to install it in one of the production server and before I do that I want to update to the latest version of 1.X series. Which branch should I use to use most upto date 1.X opensips (As master branch now is 2.1) ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Planning OpenSIPS 2.1 release - heads up
Bogdan, I am running 2.1.x and so far great, I had issue with sipteace with homer which I already reported. So please look into it before release. -- Sent from my iPhone On Mar 8, 2015, at 7:02 PM, Terrance Devor ter.de...@gmail.com wrote: Good news, What is rtpengine support. Will the proxy manage RTP directly? Or will we still have to use RTP/Media Proxy Terrance ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
