Re: [OpenSIPS-Users] Sip traces to remote Homer Server
Thanks. It is working now
On Tue, Jun 2, 2020 at 6:44 PM Maxim Sobolev wrote:
> That IP address in the listen directive should be the *address of the
> OpenSIPS machine itself*, since it's going to be the address where the HEP
> client socket binds. The destination address is specified in the modparam(
> "proto_hep",
> "hep_id", ...).
>
> The only scenario when both addresses are the same is when both OpenSIPS
> and Homer are running on the same box, which I suppose is not the case.
>
> -Max
>
> On Tue, Jun 2, 2020 at 7:07 AM Burhan Khan wrote:
>
>> There is some real IP instead of 1.2.3.4. It is pingable and it's port
>> 9060 is accessible from opensips machine.
>>
>> On Tue, Jun 2, 2020 at 3:57 PM Maxim Sobolev
>> wrote:
>>
>>> Well, apparently there is no 1.2.3.4 IP configured on your machine, you
>>> need to replace it with an actual IP address, or possibly 0.0.0.0 if a
>>> particular source address does not matter.
>>>
>>> -Max
>>>
>>> -Max
>>>
>>> On Tue, Jun 2, 2020 at 6:07 AM Burhan Khan
>>> wrote:
>>>
>>>> Hi
>>>>
>>>> I am trying to send sip traces from opensips 3.0 to remote Homer server
>>>> but it is getting error. Following is my configuration
>>>>
>>>>
>>>> loadmodule "proto_hep.so"
>>>>
>>>> loadmodule "tracer.so"
>>>>
>>>>
>>>>
>>>> listen=hep_udp:1.2.3.4:9060
>>>>
>>>>
>>>>
>>>> modparam("tracer", "trace_on", 1)
>>>>
>>>> modparam("proto_hep", "hep_id", "[homer] 1.2.3.4:9060;transport=udp")
>>>>
>>>>
>>>> modparam("tracer", "trace_id", "[tid]uri=hep:homer")
>>>>
>>>>
>>>> In route section
>>>>
>>>>
>>>> $var(trace_id) = "tid";
>>>>
>>>> trace($var(trace_id), , "sip", );
>>>>
>>>> Error is :::
>>>>
>>>> *ERROR:core:udp_init_listener: bind(27, 0x7ff5729214c4, 16) on *1.2.3.4*:
>>>> Cannot assign requested address*
>>>> *ERROR:core:trans_init_all_listeners: failed to init listener [*1.2.3.4*],
>>>> proto hep_udp*
>>>> *ERROR:core:main: failed to init all SIP listeners, aborting*
>>>>
>>>>
>>>>
>>>> *Regards*
>>>> *Burhan Khan*
>>>>
>>>> *+46769568906*
>>>> ___
>>>> Users mailing list
>>>> Users@lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>
>>>
>>> --
>>> Maksym Sobolyev
>>> Sippy Software, Inc.
>>> Internet Telephony (VoIP) Experts
>>> Tel (Canada): +1-778-783-0474
>>> Tel (Toll-Free): +1-855-747-7779
>>> Fax: +1-866-857-6942
>>> Web: http://www.sippysoft.com
>>> MSN: sa...@sippysoft.com
>>> Skype: SippySoft
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>> --
>>
>> *Regards*
>> *Burhan Khan*
>>
>> *+46769568906*
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> --
> Maksym Sobolyev
> Sippy Software, Inc.
> Internet Telephony (VoIP) Experts
> Tel (Canada): +1-778-783-0474
> Tel (Toll-Free): +1-855-747-7779
> Fax: +1-866-857-6942
> Web: http://www.sippysoft.com
> MSN: sa...@sippysoft.com
> Skype: SippySoft
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
--
*Regards*
*Burhan Khan*
*+46769568906*
___
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Re: [OpenSIPS-Users] Sip traces to remote Homer Server
There is some real IP instead of 1.2.3.4. It is pingable and it's port 9060
is accessible from opensips machine.
On Tue, Jun 2, 2020 at 3:57 PM Maxim Sobolev wrote:
> Well, apparently there is no 1.2.3.4 IP configured on your machine, you
> need to replace it with an actual IP address, or possibly 0.0.0.0 if a
> particular source address does not matter.
>
> -Max
>
> -Max
>
> On Tue, Jun 2, 2020 at 6:07 AM Burhan Khan wrote:
>
>> Hi
>>
>> I am trying to send sip traces from opensips 3.0 to remote Homer server
>> but it is getting error. Following is my configuration
>>
>>
>> loadmodule "proto_hep.so"
>>
>> loadmodule "tracer.so"
>>
>>
>>
>> listen=hep_udp:1.2.3.4:9060
>>
>>
>>
>> modparam("tracer", "trace_on", 1)
>>
>> modparam("proto_hep", "hep_id", "[homer] 1.2.3.4:9060;transport=udp")
>>
>>
>> modparam("tracer", "trace_id", "[tid]uri=hep:homer")
>>
>>
>> In route section
>>
>>
>> $var(trace_id) = "tid";
>>
>> trace($var(trace_id), , "sip", );
>>
>> Error is :::
>>
>> *ERROR:core:udp_init_listener: bind(27, 0x7ff5729214c4, 16) on *1.2.3.4*:
>> Cannot assign requested address*
>> *ERROR:core:trans_init_all_listeners: failed to init listener [*1.2.3.4*],
>> proto hep_udp*
>> *ERROR:core:main: failed to init all SIP listeners, aborting*
>>
>>
>>
>> *Regards*
>> *Burhan Khan*
>>
>> *+46769568906*
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> --
> Maksym Sobolyev
> Sippy Software, Inc.
> Internet Telephony (VoIP) Experts
> Tel (Canada): +1-778-783-0474
> Tel (Toll-Free): +1-855-747-7779
> Fax: +1-866-857-6942
> Web: http://www.sippysoft.com
> MSN: sa...@sippysoft.com
> Skype: SippySoft
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
--
*Regards*
*Burhan Khan*
*+46769568906*
___
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Users@lists.opensips.org
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[OpenSIPS-Users] Sip traces to remote Homer Server
Hi
I am trying to send sip traces from opensips 3.0 to remote Homer server but
it is getting error. Following is my configuration
loadmodule "proto_hep.so"
loadmodule "tracer.so"
listen=hep_udp:1.2.3.4:9060
modparam("tracer", "trace_on", 1)
modparam("proto_hep", "hep_id", "[homer] 1.2.3.4:9060;transport=udp")
modparam("tracer", "trace_id", "[tid]uri=hep:homer")
In route section
$var(trace_id) = "tid";
trace($var(trace_id), , "sip", );
Error is :::
*ERROR:core:udp_init_listener: bind(27, 0x7ff5729214c4, 16) on *1.2.3.4*:
Cannot assign requested address*
*ERROR:core:trans_init_all_listeners: failed to init listener [*1.2.3.4*],
proto hep_udp*
*ERROR:core:main: failed to init all SIP listeners, aborting*
*Regards*
*Burhan Khan*
*+46769568906*
___
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Users@lists.opensips.org
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[OpenSIPS-Users] OpenSips Configuration with RTP Proxy
Hi I have complied RTP Proxy and its working also I have added rtpproxy module in opensips.cfg file but it is not working. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Problem with SQL database in Federated User Location Cluster OpenSIPS 3.0
Thanks Liviu, working fine with "load-from-sql" (with dashes).
Regards,
Shahnawaz
On Tue, Sep 17, 2019 at 3:27 PM Liviu Chircu wrote:
> Hi Shanawaj,
>
> By any chance, do you get an ERROR log on startup resembling:
>
> ERROR: unknown 'restart_persistency' value: load_from_sql, using 'none'?
>
> It looks like you should be using "load-from-sql" (with dashes)
>
> Regards,
>
> Liviu Chircu
> OpenSIPS Developer
> http://www.opensips-solutions.com
>
> On 17.09.2019 12:19, Shanawaj Khan wrote:
> > modparam("usrloc", "restart_persistency", "load_from_sql")
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
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[OpenSIPS-Users] Problem with SQL database in Federated User Location Cluster OpenSIPS 3.0
Hi All,
I am trying to save user location data in SQL database for restart
persistency. I am setting parameters as
modparam("usrloc", "db_url", "mysql://root:myuser@localhost/opensips")
modparam("usrloc", "use_domain", 1)
modparam("usrloc", "location_cluster", 1)
modparam("usrloc", "cluster_mode", "federation-cachedb")
modparam("usrloc", "restart_persistency", "load_from_sql")
modparam("usrloc", "sql_write_mode", "write-back")
modparam("usrloc", "cachedb_url", "mongodb://mydomain.com:
myport/opensipsDB.userlocation")
All other OpenSIPS functionalities are working fine but I am unable to find
data in SQL database.
Thanks & Regards,
Shahnawaz
___
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Re: [OpenSIPS-Users] OpenSIPScrashing after specific interval of time
Thanks Bogdan, I am still trying to find out the actual problem. I’ll update you if I get any. On Tue, Jun 24, 2014 at 5:36 PM, Bogdan-Andrei Iancu wrote: > Hi Juned, > > That's an harmless error report - it says the WARNING SIP header (to be > attached to the message) is longer than the static buffer used for building > it. Nothing more, it is not a real error. > > Regards, > Bogdan > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 24.06.2014 14:17, Juned Khan wrote: > > Hi Bogdan, > > PFA > I am using Zenoss system monitoring tool, and it reports below error which > causes opensips to crash. > > ERROR:core:warning_builder:buffer size exceeded > > > > > > > On Thu, Jun 12, 2014 at 4:52 PM, Bogdan-Andrei Iancu > wrote: > >> Hi Juned, >> >> See http://www.opensips.org/Documentation/TroubleShooting-Crash - try >> all the suggestions, maybe you will get a core file at the end >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >> >> On 12.06.2014 08:21, Juned Khan wrote: >> >> double checked the all possible logs but didn't get anything suspected. >> also tried to disable some scripts but still having same problem, >> >> It crashes every four hour almost. >> >> >> On Tue, Jun 10, 2014 at 9:01 PM, Bogdan-Andrei Iancu > > wrote: >> >>> checked all possible log file ? syslog, messages, etc ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >>> >>> On 09.06.2014 08:42, Juned Khan wrote: >>> >>> nope I don't see such messages in Log, also I confirmed about scripts, >>> but i don’t have any scripts which causing this issue. >>> >>> >>> On Thu, Jun 5, 2014 at 9:35 PM, Bogdan-Andrei Iancu >> > wrote: >>> >>>> btw, do you see message from kernel like "opensips[xxx]: segfault >>>> at X ip ." ? >>>> >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >>>> >>>> On 05.06.2014 15:03, Bogdan-Andrei Iancu wrote: >>>> >>>> Those are just memory reports (on usage) and it is done at shutdown - >>>> nothing unusual. >>>> >>>> Are you sure your opensips is not getting terminated (stopped) by >>>> mistake by other scripts on your server ? I see no indication of a crash so >>>> far. >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >>>> >>>> On 05.06.2014 14:59, Juned Khan wrote: >>>> >>>> Hi Bogdan >>>> >>>> >>>> No I don't see any other signal than 15. >>>> >>>> I was little bit curious about this logs >>>> >>>> >>>> Jun 5 06:39:57 proxy02 opensips[24206]: used= 374656, used+overhead= >>>> 453616, free=1722496 >>>> Jun 5 06:39:57 proxy02 opensips[24206]: max used (+overhead)= 453616 >>>> Jun 5 06:39:57 proxy02 opensips[24206]: dumping free list: >>>> Jun 5 06:39:57 proxy02 opensips[24200]: max used (+overhead)= 453616 >>>> Jun 5 06:39:57 proxy02 opensips[24180]: used= 334936, used+overhead= >>>> 417856, free=1762216 >>>> Jun 5 06:39:57 proxy02 opensips[24200]: dumping free list: >>>> Jun 5 06:39:57 proxy02 opensips[24180]: max used (+overhead)= 421648 >>>> Jun 5 06:39:57 proxy02 opensips[24199]: TOTAL: 4 free fragments >>>> = 1781264 free bytes >>>> Jun 5 06:39:57 proxy02 opensips[24179]: max used (+overhead)= 405112 >>>> Jun 5 06:39:57 proxy02 opensips[24197]: max used (+overhead)= 395824 >>>> >>>> This is the only log when OpenSIPS crashes, which quite different then >>>> normal logs. >>>> And it looks like memory problem to me but I am not sure, you better know >>>> about this. >>>> >>>> >>>> >>>> On Thu, Jun 5, 2014 at 5:12 PM, Bogdan-Andrei Iancu < >>>> bog...@opensips.org> wrote: >>>> >>>>> Hi Juned, >>>>> >>>>> Your log (as per IRC discussion ) shows only signal 15 (which is SIG >>>>> TERM) when doing normal shutdown - do you have any other "signal" reported >>>>> in the logs ? >>>>> >>>>> Regards, >>>>> >>>>> Bogdan-Andrei Iancu >>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >>>>> >>>>> >>>>> > -- Thanks, Juned Khan iNextrix Technologies Pvt Ltd. www.inextrix.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPScrashing after specific interval of time
double checked the all possible logs but didn't get anything suspected. also tried to disable some scripts but still having same problem, It crashes every four hour almost. On Tue, Jun 10, 2014 at 9:01 PM, Bogdan-Andrei Iancu wrote: > checked all possible log file ? syslog, messages, etc ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 09.06.2014 08:42, Juned Khan wrote: > > nope I don't see such messages in Log, also I confirmed about scripts, > but i don’t have any scripts which causing this issue. > > > On Thu, Jun 5, 2014 at 9:35 PM, Bogdan-Andrei Iancu > wrote: > >> btw, do you see message from kernel like "opensips[xxx]: segfault >> at X ip ." ? >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >> >> On 05.06.2014 15:03, Bogdan-Andrei Iancu wrote: >> >> Those are just memory reports (on usage) and it is done at shutdown - >> nothing unusual. >> >> Are you sure your opensips is not getting terminated (stopped) by mistake >> by other scripts on your server ? I see no indication of a crash so far. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >> >> On 05.06.2014 14:59, Juned Khan wrote: >> >> Hi Bogdan >> >> >> No I don't see any other signal than 15. >> >> I was little bit curious about this logs >> >> >> Jun 5 06:39:57 proxy02 opensips[24206]: used= 374656, used+overhead= >> 453616, free=1722496 >> Jun 5 06:39:57 proxy02 opensips[24206]: max used (+overhead)= 453616 >> Jun 5 06:39:57 proxy02 opensips[24206]: dumping free list: >> Jun 5 06:39:57 proxy02 opensips[24200]: max used (+overhead)= 453616 >> Jun 5 06:39:57 proxy02 opensips[24180]: used= 334936, used+overhead= >> 417856, free=1762216 >> Jun 5 06:39:57 proxy02 opensips[24200]: dumping free list: >> Jun 5 06:39:57 proxy02 opensips[24180]: max used (+overhead)= 421648 >> Jun 5 06:39:57 proxy02 opensips[24199]: TOTAL: 4 free fragments = >> 1781264 free bytes >> Jun 5 06:39:57 proxy02 opensips[24179]: max used (+overhead)= 405112 >> Jun 5 06:39:57 proxy02 opensips[24197]: max used (+overhead)= 395824 >> >> This is the only log when OpenSIPS crashes, which quite different then >> normal logs. >> And it looks like memory problem to me but I am not sure, you better know >> about this. >> >> >> >> On Thu, Jun 5, 2014 at 5:12 PM, Bogdan-Andrei Iancu >> wrote: >> >>> Hi Juned, >>> >>> Your log (as per IRC discussion ) shows only signal 15 (which is SIG >>> TERM) when doing normal shutdown - do you have any other "signal" reported >>> in the logs ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >>> >>> On 05.06.2014 09:39, Juned Khan wrote: >>> >>> Hi All. >>> >>> Since last few months I am facing very strange issue of OpenSIPS, its >>> crashing after specific interval of time. >>> >>> I have enabled core dump parameters but dump is not generating. it >>> seems some memory related problem. I hope someone expert can give me idea >>> through this. how to solve this issue. >>> >>> I am using >>> >>> OpenSIPS 1.7.1 >>> Debian squeeze 6.0.9 >>> RAM : 16GB >>> >>> >>> Here is the opensips log >>> >>> http://pastebin.com/5YnvF6ub >>> >>> -- >>> Thanks, >>> Juned Khan >>> >>> >>> >>> ___ >>> Users mailing >>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> >> >> -- >> Thanks, >> Juned Khan >> >> >> >> >> ___ >> Users mailing >> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > > > -- > Thanks, > Juned Khan > iNextrix Technologies Pvt Ltd. > www.inextrix.com > > > -- Thanks, Juned Khan iNextrix Technologies Pvt Ltd. www.inextrix.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPScrashing after specific interval of time
nope I don't see such messages in Log, also I confirmed about scripts, but i don’t have any scripts which causing this issue. On Thu, Jun 5, 2014 at 9:35 PM, Bogdan-Andrei Iancu wrote: > btw, do you see message from kernel like "opensips[xxx]: segfault at > X ip ." ? > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 05.06.2014 15:03, Bogdan-Andrei Iancu wrote: > > Those are just memory reports (on usage) and it is done at shutdown - > nothing unusual. > > Are you sure your opensips is not getting terminated (stopped) by mistake > by other scripts on your server ? I see no indication of a crash so far. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 05.06.2014 14:59, Juned Khan wrote: > > Hi Bogdan > > > No I don't see any other signal than 15. > > I was little bit curious about this logs > > > Jun 5 06:39:57 proxy02 opensips[24206]: used= 374656, used+overhead= > 453616, free=1722496 > Jun 5 06:39:57 proxy02 opensips[24206]: max used (+overhead)= 453616 > Jun 5 06:39:57 proxy02 opensips[24206]: dumping free list: > Jun 5 06:39:57 proxy02 opensips[24200]: max used (+overhead)= 453616 > Jun 5 06:39:57 proxy02 opensips[24180]: used= 334936, used+overhead= > 417856, free=1762216 > Jun 5 06:39:57 proxy02 opensips[24200]: dumping free list: > Jun 5 06:39:57 proxy02 opensips[24180]: max used (+overhead)= 421648 > Jun 5 06:39:57 proxy02 opensips[24199]: TOTAL: 4 free fragments = > 1781264 free bytes > Jun 5 06:39:57 proxy02 opensips[24179]: max used (+overhead)= 405112 > Jun 5 06:39:57 proxy02 opensips[24197]: max used (+overhead)= 395824 > > This is the only log when OpenSIPS crashes, which quite different then normal > logs. > And it looks like memory problem to me but I am not sure, you better know > about this. > > > > On Thu, Jun 5, 2014 at 5:12 PM, Bogdan-Andrei Iancu > wrote: > >> Hi Juned, >> >> Your log (as per IRC discussion ) shows only signal 15 (which is SIG >> TERM) when doing normal shutdown - do you have any other "signal" reported >> in the logs ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >> >> On 05.06.2014 09:39, Juned Khan wrote: >> >> Hi All. >> >> Since last few months I am facing very strange issue of OpenSIPS, its >> crashing after specific interval of time. >> >> I have enabled core dump parameters but dump is not generating. it seems >> some memory related problem. I hope someone expert can give me idea through >> this. how to solve this issue. >> >> I am using >> >> OpenSIPS 1.7.1 >> Debian squeeze 6.0.9 >> RAM : 16GB >> >> >> Here is the opensips log >> >> http://pastebin.com/5YnvF6ub >> >> -- >> Thanks, >> Juned Khan >> >> >> >> ___ >> Users mailing >> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > > > -- > Thanks, > Juned Khan > > > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- Thanks, Juned Khan iNextrix Technologies Pvt Ltd. www.inextrix.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPScrashing after specific interval of time
Hi Bogdan No I don't see any other signal than 15. I was little bit curious about this logs Jun 5 06:39:57 proxy02 opensips[24206]: used= 374656, used+overhead=453616, free=1722496 Jun 5 06:39:57 proxy02 opensips[24206]: max used (+overhead)= 453616 Jun 5 06:39:57 proxy02 opensips[24206]: dumping free list: Jun 5 06:39:57 proxy02 opensips[24200]: max used (+overhead)= 453616 Jun 5 06:39:57 proxy02 opensips[24180]: used= 334936, used+overhead=417856, free=1762216 Jun 5 06:39:57 proxy02 opensips[24200]: dumping free list: Jun 5 06:39:57 proxy02 opensips[24180]: max used (+overhead)= 421648 Jun 5 06:39:57 proxy02 opensips[24199]: TOTAL: 4 free fragments = 1781264 free bytes Jun 5 06:39:57 proxy02 opensips[24179]: max used (+overhead)= 405112 Jun 5 06:39:57 proxy02 opensips[24197]: max used (+overhead)= 395824 This is the only log when OpenSIPS crashes, which quite different then normal logs. And it looks like memory problem to me but I am not sure, you better know about this. On Thu, Jun 5, 2014 at 5:12 PM, Bogdan-Andrei Iancu wrote: > Hi Juned, > > Your log (as per IRC discussion ) shows only signal 15 (which is SIG TERM) > when doing normal shutdown - do you have any other "signal" reported in the > logs ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 05.06.2014 09:39, Juned Khan wrote: > >Hi All. > > Since last few months I am facing very strange issue of OpenSIPS, its > crashing after specific interval of time. > > I have enabled core dump parameters but dump is not generating. it seems > some memory related problem. I hope someone expert can give me idea through > this. how to solve this issue. > > I am using > > OpenSIPS 1.7.1 > Debian squeeze 6.0.9 > RAM : 16GB > > > Here is the opensips log > > http://pastebin.com/5YnvF6ub > > -- > Thanks, > Juned Khan > > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- Thanks, Juned Khan <http://www.inextrix.com/> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPScrashing after specific interval of time
Hi All. Since last few months I am facing very strange issue of OpenSIPS, its crashing after specific interval of time. I have enabled core dump parameters but dump is not generating. it seems some memory related problem. I hope someone expert can give me idea through this. how to solve this issue. I am using OpenSIPS 1.7.1 Debian squeeze 6.0.9 RAM : 16GB Here is the opensips log http://pastebin.com/5YnvF6ub -- Thanks, Juned Khan <http://www.inextrix.com/> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] error: 'mysql' tool not found: set MYSQL variable to correct tool path
Thanks for the reply, which mysql shows /usr/bin/mysql but I uninstalled opensips 1.9 and Installed 1.8 following the webinar steps. and its done. On Wed, May 29, 2013 at 1:27 PM, Răzvan Crainea wrote: > Hi, Sunny! > > It seems like the opensipsctl tool cannot find the mysql application. Can > you run 'which mysql' and see if you have it in your PATH? > > Best regards, > > Razvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.**com <http://www.opensips-solutions.com> > > > On 05/28/2013 03:20 PM, Sunny Khan wrote: > >> Hello, >> >> Please help me in this I get the following error >> >> opensipsdbctl create >> >> /usr/local/lib64/opensips/**opensipsctl/opensipsdbctl.**mysql: line 25: >> locate_tool: command not found >> error: 'mysql' tool not found: set MYSQL variable to correct tool path >> >> libmysqlclient-dev, and mysql-client already installed. >> >> >> this is my opensipsctlrc >> >> >> >> DBENGINE=MYSQL >> >> ## database host >> DBHOST=79.143.177.201 >> >> ## database name (for ORACLE this is TNS name) >> DBNAME=opensips >> >> # database path used by dbtext or db_berkeley >> # DB_PATH="/usr/local/etc/**opensips/dbtext" >> >> ## database read/write user >> DBRWUSER=opensips >> >> ## password for database read/write user >> DBRWPW="Jeddah@@1375" >> >> ## database super user (for ORACLE this is 'scheme-creator' user) >> #DBROOTUSER="root" >> >> >> >> __**_ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-**bin/mailman/listinfo/users<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<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] error: 'mysql' tool not found: set MYSQL variable to correct tool path
Hello, Please help me in this I get the following error opensipsdbctl create /usr/local/lib64/opensips/opensipsctl/opensipsdbctl.mysql: line 25: locate_tool: command not found error: 'mysql' tool not found: set MYSQL variable to correct tool path libmysqlclient-dev, and mysql-client already installed. this is my opensipsctlrc DBENGINE=MYSQL ## database host DBHOST=79.143.177.201 ## database name (for ORACLE this is TNS name) DBNAME=opensips # database path used by dbtext or db_berkeley # DB_PATH="/usr/local/etc/opensips/dbtext" ## database read/write user DBRWUSER=opensips ## password for database read/write user DBRWPW="Jeddah@@1375" ## database super user (for ORACLE this is 'scheme-creator' user) #DBROOTUSER="root" ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Problem with Process ID
Hello Faisal, Make these changes, hope this will help you. sudo nano /usr/local/etc/opensips/opensipsctlrc uncomment and change : # PID_FILE=/var/run/opensips.pid to : PID_FILE=/var/run/opensips/opensips.pid http://vidodz.wordpress.com/2009/07/28/install-opensips-on-debian-or-ubuntu/ On Mon, Sep 19, 2011 at 3:35 PM, Faisal Rehman wrote: > *Hi,* > > Sometimes it happens that I make some changes in the opensips.cfg file & > save it but unfortunately I couldn't find the pid file in my processes > directory. What can be the reasons for it? > > > > With Best Regards, > > *Faisal Rehman* > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- *Abid khan Advanced VoIP Cell:+923315391629 *** ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Rejoindre mon réseau sur LinkedIn
LinkedIn Abid khan souhaite se connecter à vous sur LinkedIn : -- Je vous invite à faire partie de mon réseau professionnel en ligne sur le site LinkedIn. Accepter l'invitation de Abid khan http://www.linkedin.com/e/-iucuba-gq4pqp8u-59/XiGay8n_CAD-74lZADGGFpZv1NDQ54l0NBPI0M/blk/I114931795_105/6lColZJrmZznQNdhjRQnOpBtn9QfmhBt71BoSd1p65Lr6lOfPkMclYRejsNcPAQcj59bR1QgkETq4dgbPwRc3sRd3kQdjcLrCBxbOYWrSlI/EML_comm_afe/ Voir l'invitation de Abid khan http://www.linkedin.com/e/-iucuba-gq4pqp8u-59/XiGay8n_CAD-74lZADGGFpZv1NDQ54l0NBPI0M/blk/I114931795_105/dj0NnPkVdP4PejgNckALqnpPbOYWrSlI/svi/ -- SAVEZ-VOUS que vous pouvez être le premier informé quand un membre de votre réseau change de poste ? Les "Nouvelles du réseau" sur votre page d'accueil LinkedIn vous informe des évolutions de carrière dans votre réseau. Soyez informé(e) en premier ! http://www.linkedin.com/ -- (c) 2011, LinkedIn Corporation___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Pjsip dialler with openIMS core
Hello everybody, I want to know main criteria or parameters which i have to fulfill if i want to register* pjsip dialer* with openIMS core. When i try to register pjsip dialler with openims core my server send does not send "Authorization " to pjsip in response and then 504 (request time out) in end. So any one can please help me in this. -- *Abid khan * ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenIMS core (localhost to real IP)
Hello everyone, I need help to configure openIMS core to real IP from local host, plz suggest me some links or documentation. Thanks in Advance. -- *Abid khan * ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Call forwarding to Android using OPenIMS core +Presence Server
Hello , I need some help regarding forwarding of call to to android phone when user is busy or don't want to receive call on their system.How can i do this using presence information. i have configured open ims core and presence server. Thanks in Advance -- *Abid khan * ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Request Timeout (for PUBLISH and SUBSCRIBE) in IMS+Presence Server
*Hi! * I configured the IMS core and opensip presence server on it, but when i am trying to configure the Client(UCT) with presence enabled it gives me following error: * Request Timeout (for PUBLISH) Request Timeout (for SUBSCRIBE)* if any one knows that kindly let me know i am stuck in this since last week. Flow of packets and opensips.cfg file is attached. -- *Abid khan NUST-SEECS BSIT * text Description: Binary data opensips.cfg Description: Binary data ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Request time out for (SUBSCRIBE + PUBLISH) in IMS+presence Server
{
setflag(1); # do accounting
}
if (!uri==myself)
## replace with following line if multi-domain support is used
##if (!is_uri_host_local())
{
append_hf("P-hint: outbound\r\n");
# if you have some interdomain connections via TLS
##if($rd=="tls_domain1.net") {
##t_relay("tls:domain1.net");
##exit;
##} else if($rd=="tls_domain2.net") {
##t_relay("tls:domain2.net");
##exit;
##}
route(1);
}
# requests for my domain
## uncomment this if you want to enable presence server
## and comment the next 'if' block
## NOTE: uncomment also the definition of route[2] from below
if( is_method("PUBLISH|SUBSCRIBE"))
route(2);
#if (is_method("PUBLISH"))
#{
#sl_send_reply("503", "Service Unavailable");
#exit;
#}
if (is_method("REGISTER"))
{
# authenticate the REGISTER requests (uncomment to enable auth)
#if (!www_authorize("open-ims.test", "subscriber"))
#{
#www_challenge("open-ims.test", "0");
#exit;
#}
if (!db_check_to())
{
sl_send_reply("403","Forbidden auth ID");
exit;
}
if (!save("location"))
sl_reply_error();
exit;
}
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# apply DB based aliases (uncomment to enable)
##alias_db_lookup("dbaliases");
# do lookup with method filtering
if (!lookup("location","m")) {
switch ($retcode) {
case -1:
case -3:
t_newtran();
t_reply("404", "Not Found");
exit;
case -2:
sl_send_reply("405", "Method Not Allowed");
exit;
}
}
# when routing via usrloc, log the missed calls also
setflag(2);
route(1);
}
route[1] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
t_on_branch("2");
t_on_reply("2");
t_on_failure("1");
}
if (!t_relay()) {
sl_reply_error();
};
exit;
}
# Presence route
/* uncomment the whole following route for enabling presence
NOTE: do not forget to enable the call of this route from the main
route */
route[2]
{
if (!t_newtran())
{
sl_reply_error();
exit;
};
if(is_method("PUBLISH"))
{
handle_publish();
}
else
if( is_method("SUBSCRIBE"))
{
handle_subscribe();
}
exit;
}
branch_route[2] {
xlog("new branch at $ru\n");
}
onreply_route[2] {
xlog("incoming reply\n");
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
# uncomment the following lines if you want to block client
# redirect based on 3xx replies.
##if (t_check_status("3[0-9][0-9]")) {
##t_reply("404","Not found");
##exit;
##}
# uncomment the following lines if you want to redirect the failed
# calls to a different new destination
##if (t_check_status("486|408")) {
##sethostport("192.168.2.100:5060");
### do not set the missed call flag again
##t_relay();
##}
}
--
*Abid khan
NUST-SEECS BSIT
*
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[OpenSIPS-Users] Request Timeout (for PUBLISH and SUBSCRIBE) in IMS+opensips Presence Server
BE sip:al...@open-ims.test SIP/2.0 Record-Route: Record-Route: Record-Route: Record-Route: Route: , Record-Route: Via: SIP/2.0/UDP 127.0.0.1:5065;branch=z9hG4bKff21.e9bdf697.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKff21.e85e0a12.0 Via: SIP/2.0/UDP 127.0.0.1:5065;branch=z9hG4bKff21.d9bdf697.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKff21.d85e0a12.0 Via: SIP/2.0/UDP 127.0.0.1:5065;branch=z9hG4bKff21.c9bdf697.0 Via: SIP/2.0/UDP 127.0.0.1:6060;branch=z9hG4bKff21.f8017552.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKff21.c85e0a12.0 Via: SIP/2.0/UDP 127.0.0.1:4060;branch=z9hG4bKff21.b073da03.0 Via: SIP/2.0/UDP 192.168.0.213:5061 ;received=127.0.0.1;received=127.0.0.1;rport=5061;branch=z9hG4bK3246144 From: ;tag=2049380917 To: Call-ID: 269819684 CSeq: 20 SUBSCRIBE Contact: Max-Forwards: 9 User-Agent: UCT IMS Client Expires: 3600 Event: presence.winfo Content-Length: 0 P-Asserted-Identity: P-Charging-Vector: icid-value="P-CSCFabcd4d6500d80004";icid-generated-at=127.0.0.1;orig-ioi="open-ims.test" P-hint: I-CSCF rr-enforced P-hint: outbound P-hint: I-CSCF rr-enforced P-hint: outbound P-hint: I-CSCF rr-enforced P-hint: outbound -- *Abid khan NUST-SEECS BSIT* ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] help in Presence module
yes i configured the SIP client and calling and IM is working , but the presence module is not working and i am not getting where is the problem, whether it is in opensip configuration or somthing else... On Mon, Feb 14, 2011 at 6:48 PM, Anca Vamanu wrote: > Hi Abid, > > I suppose you configured your SIP client so that it sends Subscribe and > Publish with a Route header ( a preloaded route) and OpenSIPS does not allow > this. You should watch the network trace to see exactly what happens. > > Regards, > > -- > Anca Vamanu > OpenSIPS Developer > > > On 02/14/2011 11:59 AM, abid khan wrote: > > Hi! > I am getting this error while i configured presence server of Opensips > and try to run UCT client on top of it. Can any one guide me about this. > 14:49:41> REGISTER > 14:49:41> Unauthorized - Challenging the UE (for REGISTER) > 14:49:41> REGISTER with credentials > 14:49:41> OK - SAR succesful and registrar saved (for REGISTER) > 14:49:41> IMPUs: > 14:49:41> SUBSCRIBE (reg event) > 14:49:41> PUBLISH (presence) > 14:49:41> SUBSCRIBE (presence.winfo) > 14:49:41> Subscription to REG saved (for SUBSCRIBE) > 14:49:41> Preload Route denied (for PUBLISH) > 14:49:41> Preload Route denied (for SUBSCRIBE) > 14:49:44> NOTIFY > 14:49:44> OK (NOTIFY) > > > > -- > *Abid khan > NUST-SEECS BSIT > Final Year Student* > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- *Abid khan NUST-SEECS BSIT Final Year Student* ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] help in Presence module
Hi! I am getting this error while i configured presence server of Opensips and try to run UCT client on top of it. Can any one guide me about this. 14:49:41> REGISTER 14:49:41> Unauthorized - Challenging the UE (for REGISTER) 14:49:41> REGISTER with credentials 14:49:41> OK - SAR succesful and registrar saved (for REGISTER) 14:49:41> IMPUs: 14:49:41> SUBSCRIBE (reg event) 14:49:41> PUBLISH (presence) 14:49:41> SUBSCRIBE (presence.winfo) 14:49:41> Subscription to REG saved (for SUBSCRIBE) 14:49:41> Preload Route denied (for PUBLISH) 14:49:41> Preload Route denied (for SUBSCRIBE) 14:49:44> NOTIFY 14:49:44> OK (NOTIFY) -- *Abid khan NUST-SEECS BSIT Final Year Student* ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS 1.6 on Ubuntu
Ricky, If you expand on the problem where you are stuck may be someone can help you... Please be specific that what is the nature of the problem where you are stuck... Khan On Tue, Feb 1, 2011 at 12:21 PM, Robin Malhotra wrote: > > Guys I a newbie to OpenSIPS > > I have installed opensips and mysql on ubuntu following some instructions. > I have also installed x-lite. Now how to register a user in opensips and to > use it with the client ? I am stuck, please let me know > > Regards > Ricky > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- Khan VoIP Rookie Every beginning has an end regardless we believe it or not... ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Transfer issue
Please let me know when you setup the blogspot i'm very interested in seeing how you did it, or if you could provide me your configuration that will be really great. Thanks in advance... Khan On Wed, Oct 28, 2009 at 7:34 AM, Iñaki Baz Castillo wrote: > 2009/10/28 Peter den Hartog : > > Oke i feel so happy right now, i fixed it! it works! i can now create > dials > > over opensips, true asterisk, outside inside i can transfer, everything > > works! damn i'm happy :D! > > the answer was in my opensips.cfg and the routing back to asterisk, i've > > created a routing script that just subscribe and trows the rest in to > > asterisk. > > > > i'm thinking of creating a big straightforward blogpost about this, how > you > > should do this, with what goes where and stuff like that. > > > > I would like to thank everybody who replied on this issue, thanks alot. > > Congratulations ;) > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Khan VoIP Rookie Every beginning has an end regardless we believe it or not... ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.5.x nathelper/rtpproxy configuration
Mani, There is a complete working configuration posted at the following blog link: http://voiprookie.blogspot.com/2009/04/rtpproxy-12x-installation.html You might need to tuneup a little bit based on your needs but last time i have tried it and it was functional. -- Khan VoIP Rookie Every beginning has an end regardless we believe it or not... On Tue, Oct 20, 2009 at 8:20 PM, Manivasagam Sivaraman < smvasagam2...@gmail.com> wrote: > Dear Pros > > The nathelper documentation is good as shown below > http://www.opensips.org/html/docs/modules/1.5.x/nathelper.html#id228280 > > Seems like from 1.5.x many nathelper functions are deprecated and new > functions are introduced like rtpproxy_offer and rtpproxy_answer. > > Actually NAT traversal is still a nightmare for many and beginers it would > be better if some professional post a full working opensips.cfg > nathelper/rtpproxy configuration script for the latest 1.5.3. THe above > documents shows bits and pieces, but a full working config example will > really help. Could any one please post a working example of opensips.cfg > nathelper configuration please. I searched and foind only old configuration > that does not work well. > > Thanks in Advance. I really appreaciate your understanding. > Mani > > ___ > 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] NAT problem, no-audio when calling outside network... Please help
mp; from_uri==myself) /*no multidomain
version*/
##if (!(method=="REGISTER") && is_from_local()) /*multidomain version*/
{
if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
exit;
}
if (!check_from()) {
sl_send_reply("403","Forbidden auth ID");
exit;
}
consume_credentials();
# caller authenticated
}
# preloaded route checking
if (loose_route()) {
xlog("L_ERR",
"Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");
if (!is_method("ACK"))
sl_send_reply("403","Preload Route denied");
exit;
}
# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();
# account only INVITEs
if (is_method("INVITE")) {
setflag(1); # do accounting
}
if (!uri==myself)
## replace with following line if multi-domain support is used
##if (!is_uri_host_local())
{
append_hf("P-hint: outbound\r\n");
# if you have some interdomain connections via TLS
##if($rd=="tls_domain1.net") {
## t_relay("tls:domain1.net");
## exit;
##} else if($rd=="tls_domain2.net") {
## t_relay("tls:domain2.net");
## exit;
##}
route(1);
}
# requests for my domain
if (is_method("PUBLISH")) {
sl_send_reply("503", "Service Unavailable");
exit;
}
if (is_method("REGISTER")) {
# authenticate the REGISTER requests (uncomment to enable auth)
if (!www_authorize("", "subscriber")) {
xlog("L_INFO", "1st Pass - Register authentication - M=$rm
RURI=$ru
F=$fu T=$tu IP=$si ID=$ci\n");
www_challenge("", "0");
exit;
}
if (!check_to()) {
xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu
T=$tu
IP=$si ID=$ci\n");
sl_send_reply("403","Forbidden auth ID");
exit;
}
if (!save("location"))
sl_reply_error();
xlog("L_INFO", "2nd Pass - Registration successful - M=$rm
RURI=$ru
F=$fu T=$tu IP=$si ID=$ci\n");
exit;
}
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
if (!lookup("location")) {
switch ($retcode) {
case -1:
case -3:
t_newtran();
t_reply("404", "Not Found");
exit;
case -2:
sl_send_reply("405", "Method Not Allowed");
exit;
}
}
# when routing via usrloc, log the missed calls also
setflag(2);
route(1);
}
#-->
route[1] {
if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&
!search("^Route:")){
sl_send_reply("479", "We don't forward to private IP
addresses");
exit;
};
if (isbflagset(6)) {
force_rtp_proxy();
};
t_on_reply("1");
#! *** <<
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
t_on_branch("2");
t_on_reply("2");
t_on_failure("1");
}
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
exit;
}
# !! Nathelper
onreply_route[1] {
# NATed transaction ?
if (isbflagset(6) && status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
force_rtp_proxy();
# otherwise, is it a transaction behind a NAT and we did not
# know at time of request processing ? (RFC1918 contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
onreply_route[2] {
xlog("incoming reply\n");
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
}
*
The output capture from WireShark is at the following link.
http://pastebin.com/m1c17484d
Please help me figure out this problem, I appreciate your time.
Thank you,
Khan
VoIP Rookie
Every beginning has an end regardless we believe it or not...
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Re: [OpenSIPS-Users] AVP/ Programing Documentation
Ghaith, That would be violation of copyright law won't it or else we wont buy this book for $40+ Authour has putin hardwork in putting it together, dont you think he should benefit a little ? After all OpenSIPS is free but not everything else lol... On Wed, Aug 19, 2009 at 11:41 AM, Ghaith ALKAYYEM wrote: > Hi,, > I think this book is available through rapidshare in the following link: > > http://rs211.rapidshare.com/files/143531736/Building_Telephony_Systems_with_Openser.pdf > > > > On Wed, 2009-08-19 at 11:26 -0500, osiris123d wrote: >> Also a good idea is to read Flavio E. Goncalves book " >> http://www.packtpub.com/building-telephony-systems-with-openser/book >> Building Telephony systems with OpenSER ". Though the book does deal with >> an old version of OpenSER it still has good documentation and examples. >> >> There is also a section on AVPOPS >> >> >> >> roger wilbert wrote: >> > >> > Can anyone point me to documentation other than the module docs that can >> > explain how to AVP? Not that the module docs don’t provide good >> > information. But it assumes knowledge that is missing from someone who is >> > not readily familiar with Opensips. >> > >> > ___ >> > 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 > -- Khan VoIP Rookie Every beginning has an end regardless we believe it or not... ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Error 408 after installing 1.5.2 - UAC doesn't register
Hello everyone, I have installed 1.5.2 on ubuntue 9.0.4 sever. After installation when i ran OpenSIPS without any database or any other changes in opensips.cfg file I get REGISTRATION timeout constantly. I don't understand why this is happening, can anyone help please. Attached is a wireshark trace... -- Khan VoIP Rookie Every beginning has an end regardless we believe it or not... 080909 Description: Binary data ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ACK timout OpenSIPS MediaProxy Integration using MySQL db
Thanks Adrian for your advise, I really appreciate it, if you look at the mailing list by author you will see in last year, my problem has been same, i have been very specific and at times i have been random. I appologize for my randomness, and trust me i grew in technology quite bit by reading documentations and applying the knowledge. I will keep in mind and stick with specific question, even though it has been asked again and again... Thanks again, Khan On Tue, May 12, 2009 at 11:04 AM, Adrian Georgescu wrote: >> Khan, > > > The quality of an answer seldom exceeds the quality of the asked question. > The more accurate and focused the question is, the more the chance is that > you get a good answer for it. Asking to a busy mailing list a set of wide > questions caused by mere ignorance of the subject will likely yield no > results as you have not taken the time to learn the basics and there are > very few who can afford to spend time in your behalf on such range and > volume of questions. > > You can compensate the frustration by reading the software documentation and > then by asking one question at a time for a particular problem. > > Adrian > > >> This is quite annoying since the same question has been asked here >> million different ways by me but no straight answer or guidence, and i >> dont believe i am slow learner either... >> Frustrated >> Khan >> >> > -- Khan VoIP Rookie Every beginning has an end regardless we believe it or not... ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ACK timout OpenSIPS MediaProxy Integration using MySQL db
*** This is not directed to one person, please feel free to respond ***
I appologize for my ignorace since I am newbee and in process of learning...
I am using standard CFG file with few additions only, rest of
the configuration is standard. I added lines related to record_route()
as follows:
if(!is_method("REGISTER")){
if(nat_uac_test("19")){
setbflag(6);
force_rport();
fix_contact();
record_route(";nat=yes");
} else {
record_route();
};
};
And mediaproxy activation in places where i believe i should be using,
please verify if you could that:
a) did i have all the necessary calls to activate mediaproxy in place?
b) does it activate mediaproxy when 200 OK is sent?
c) what constitutes to UAC start sending "OPTIONS" and not been able
to get ACK for other UAC that has been called?
Please explain to me, how would it effect, and where does mediaproxy
comes in play during call?
I thank all of you in advance...
This is quite annoying since the same question has been asked here
million different ways by me but no straight answer or guidence, and i
dont believe i am slow learner either...
Frustrated
Khan
VoIP Rookie
Every beginning has an end regardless we believe it or not...
On Mon, May 11, 2009 at 5:54 AM, Ruud Klaver wrote:
> Hi,
>
> On 09 May 2009, at 04:59, Khan wrote:
>
>> Sorry forgot the attachment of WS...
>>
>> Please Help...
>>
>> On Fri, May 8, 2009 at 9:56 PM, Khan wrote:
>>
>> Hi everyone,
>>
>> I am having trouble running OpenSIPS, while back I had same problem,
>> this problem is surfaced back again, I am sure there is something I am
>> missing in my configuration.
>> Currently i am trying to do followings:
>> UAC1 >>> MyOpenSIPS Server >>> MySQL auth >>> MediaProxy >>> UAC2
>>
>> At this point, I don't have anything else running except OpenSIPS,
>> MySQL, and MediaProxy.
>>
>> My connections are such...
>> ISP Modem --> *MyRouter ---> MyServerBox
>>
>> * MyRouter has open ports, 80,22,5060, 1-13000, 5-6
>> My Goals is to Integrate OpenSIPS. MySQL, and MediaProxy and make it
>> functional...
>>
>> I started mediaproxy as follows:
>>
>> root# media-dispatcher --no-fork
>> Starting MediaProxy Dispatcher 2.3.4
>> Twisted is using epollreactor
>> mediaproxy.dispatcher.RelayFactory starting on 50100
>> mediaproxy.dispatcher.OpenSIPSControlFactory starting on
>> "'/var/run/mediaproxy/dispatcher.sock'"
>> mediaproxy.dispatcher.ManagementControlFactory starting on 25061
>>
>>
>> Problem is that MediaProxy is not working with my configuration, I
>> don't know what is going on. I can see both MediaProxy and OpenSIPS is
>> running on my server BUT UAC outside my network still giving me the
>> same problem as in the beginning. I constantly receive
>> "OPTIONS"/SUBSCRIBE
>> from softphone outside my network since it doesn't received ACK, thus
>> generate error...
>>
>> I made a call which lasted 35 seconds and got cut off giving UAC of
>> other party an error of network failure.
>>
>> I have produced WS trace, please look at it and guide me what is wrong
>> with this situation, also my configuration is on the following link as
>> of today... http://pastebin.com/m3cf2769e
>>
>> Thanks for all your help,
>>
>>
>> --
>> Khan
>>
>>
>> VoIP Rookie
>> Every beginning has an end regardless we believe it or not...
>>>
>>
>>
>>
>> --
>> Khan
>>
>>
>> VoIP Rookie
>> Every beginning has an end regardless we believe it or not...
>> <0507WStrace.pcap>
>
> I don't see how this has anything to do with mediaproxy, but you should be
> very suspicious about the fact that both the 180 and the 200 OK in response
> to the INVITE contain a Record-Route with a private IP address
> (192.168.1.9), to which the ACK is subsequently sent.
>
> Ruud Klaver
> AG Projects
>
--
Khan
VoIP Rookie
Every beginning has an end regardless we believe it or not...
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Re: [OpenSIPS-Users] [OpenSIPS] sample script that works with cdrtools, freeradius, nat, drouting
[1-9][0-9]{6}") {
> $rU = $(fU{s.substr, 0, 4}) + $rU;
> }
> # 411 Local Directory Assistance
> else if (uri=~"^sip:4...@.*") {
> # the uri with a default call to "local directory
> assistance".
> $rU = $(fU{s.substr, 0, 4}) + "5551212";
> }
>
> # 611 Local Directory Assistance
> else if (uri=~"^sip:6...@.*") {
> # the uri with a default call to "local directory
> assistance".
>
> rewriteuri("sip:17775551...@machinename.somedomain.com");
> }
> #911 is handled by E911 service provider
> else {
> sl_send_reply("404", "Invalid destination");
> exit;
> }
>
> # Set the callerid for the user from an AVP
> #if (avp_db_load("$from/username", "s:callerid")) {
> # subst('/^From: (.*)>(.*)$/From: $avp(callerid)>\2/ig');
> #};
>
> }
>
> if (is_method("INVITE|BYE")) {
> setflag(1); # do accounting ...
> setflag(2); # sip trace
> #call the accounting functions explicitly in local_route for
> #the internally generated BYEs as they do not trigger accounting by just
> #setting the accounting flag
> acc_rad_request("200 ok");
> acc_log_request("200 ok");
> }
>
> #change access point phone number to inbound route for asterisk
> alias_db_lookup("dbaliases");
> #forward asterisk inbound route with dispatcher as load balancer
> if (is_method("INVITE") && $ruri =~ "^sip:17771000...@.*" ) {
> #dispatcher select from set 1 using algorithm 0.
> if(!ds_select_dst("1", "0"))
> {
> sl_send_reply("404", "no destination");
> }
> if(!t_relay()) sl_reply_error();
> exit;
> };
>
> #special relaying to asterisk finished, now we process regular
> requests.
>
> #INVITE, ACK, BYE, OPTIONS to locally registered user.
> if (lookup("location"))
> {
> if (is_method("INVITE")) {
> t_on_branch("1");
> t_on_reply("1");
> t_on_failure("1");
> }
> if (!t_relay()) {
> sl_reply_error();
> };
> exit;
> }
>
> #INVITE to outgoing gateway, route it out.
> if (is_method("INVITE") ) {
> #if (cr_route("default", "machinename.somedomain.com",
> "$rU", "$rU", "call_id")) {
> if (do_routing()) {
> t_on_failure("11");
> if (!t_relay()) {
> sl_reply_error();
> };
> exit;
> };
> exit;
> };
>
> if (!t_relay()) {
> sl_reply_error();
> };
> exit;
> }
>
>
> branch_route[1] {
> xlog("new branch at $ru\n");
> }
>
>
> onreply_route[1] {
> xlog("incoming reply\n");
> }
>
>
> failure_route[1] {
> if (t_was_cancelled()) {
> exit;
> }
>
> # uncomment the following lines if you want to block client
> # redirect based on 3xx replies.
> ##if (t_check_status("3[0-9][0-9]")) {
> ##t_reply("404","Not found");
> ## exit;
> ##}
>
> # uncomment the following lines if you want to redirect the failed
> # calls to a different new destination
> ##if (t_check_status("486|408")) {
> ## sethostport("192.168.2.100:5060");
> ## append_branch();
> ## # do not set the missed call flag again
> ## t_relay();
> ##}
> }
>
>
> ##
> # "default" failover #
> ##
> failure_route[11] {
> xlog("L_INFO", "entering failure_route[11] for reply code
> '$T_reply_code'\n");
>
> if (t_was_cancelled()) {
> exit;
> }
>
> if (t_check_status("408|5[0-9][0-9]")) {
> #xlog("L_INFO","cr_tree_rewrite_uri(\"default\", \"1\");\n");
> #if (cr_route("default", "machinename.somedomain.com", "$rU", "$rU",
> "call_id")) {
> if (do_routing()) {
> t_on_failure("12");
> append_branch();
> route(1);
> };
> exit;
> } else if (t_check_status("3[0-9][0-9]")) {
> t_reply("404","Not found");
> exit;
> }
> }
>
> failure_route[12] {
> xlog("L_INFO", "entering failure_route[12] for reply code
> '$T_reply_code'\n");
> if (t_was_cancelled()) {
> exit;
> }
>
> if (t_check_status("408|5[0-9][0-9]")) {
> xlog("L_INFO","cr_tree_rewrite_uri(\"default\", \"2\");\n");
> #if (cr_route("default", "machinename.somedomain.com", "$rU", "$rU",
> "call_id")) {
> if (do_routing()) {
> t_on_failure("13");
> append_branch();
> route(1);
> };
> exit;
> } else if (t_check_status("3[0-9][0-9]")) {
> t_reply("404","Not found");
> exit;
> }
> }
>
> failure_route[13] {
> xlog("L_INFO", "entering failure_route[13] for reply code
> '$T_reply_code'\n");
> if (t_was_cancelled()) {
> exit;
> }
> }
>
>
>
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
May be if i run into problems, could i also seek your help
--
Khan
VoIP Rookie
Every beginning has an end regardless we believe it or not...
___
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Re: [OpenSIPS-Users] ACK timout OpenSIPS MediaProxy Integration using MySQL db
Sorry forgot the attachment of WS... Please Help... On Fri, May 8, 2009 at 9:56 PM, Khan wrote: Hi everyone, I am having trouble running OpenSIPS, while back I had same problem, this problem is surfaced back again, I am sure there is something I am missing in my configuration. Currently i am trying to do followings: UAC1 >>> MyOpenSIPS Server >>> MySQL auth >>> MediaProxy >>> UAC2 At this point, I don't have anything else running except OpenSIPS, MySQL, and MediaProxy. My connections are such... ISP Modem --> *MyRouter ---> MyServerBox * MyRouter has open ports, 80,22,5060, 1-13000, 5-6 My Goals is to Integrate OpenSIPS. MySQL, and MediaProxy and make it functional... I started mediaproxy as follows: root# media-dispatcher --no-fork Starting MediaProxy Dispatcher 2.3.4 Twisted is using epollreactor mediaproxy.dispatcher.RelayFactory starting on 50100 mediaproxy.dispatcher.OpenSIPSControlFactory starting on "'/var/run/mediaproxy/dispatcher.sock'" mediaproxy.dispatcher.ManagementControlFactory starting on 25061 Problem is that MediaProxy is not working with my configuration, I don't know what is going on. I can see both MediaProxy and OpenSIPS is running on my server BUT UAC outside my network still giving me the same problem as in the beginning. I constantly receive "OPTIONS"/SUBSCRIBE from softphone outside my network since it doesn't received ACK, thus generate error... I made a call which lasted 35 seconds and got cut off giving UAC of other party an error of network failure. I have produced WS trace, please look at it and guide me what is wrong with this situation, also my configuration is on the following link as of today... http://pastebin.com/m3cf2769e Thanks for all your help, -- Khan VoIP Rookie Every beginning has an end regardless we believe it or not... > -- Khan VoIP Rookie Every beginning has an end regardless we believe it or not... 0507WStrace.pcap Description: application/cap ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] ACK timout OpenSIPS MediaProxy Integration using MySQL db
Hi everyone, I am having trouble running OpenSIPS, while back I had same problem, this problem is surfaced back again, I am sure there is something I am missing in my configuration. Currently i am trying to do followings: UAC1 >>> MyOpenSIPS Server >>> MySQL auth >>> MediaProxy >>> UAC2 At this point, I don't have anything else running except OpenSIPS, MySQL, and MediaProxy. My connections are such... ISP Modem --> *MyRouter ---> MyServerBox * MyRouter has open ports, 80,22,5060, 1-13000, 5-6 My Goals is to Integrate OpenSIPS. MySQL, and MediaProxy and make it functional... I started mediaproxy as follows: root# media-dispatcher --no-fork Starting MediaProxy Dispatcher 2.3.4 Twisted is using epollreactor mediaproxy.dispatcher.RelayFactory starting on 50100 mediaproxy.dispatcher.OpenSIPSControlFactory starting on "'/var/run/mediaproxy/dispatcher.sock'" mediaproxy.dispatcher.ManagementControlFactory starting on 25061 Problem is that MediaProxy is not working with my configuration, I don't know what is going on. I can see both MediaProxy and OpenSIPS is running on my server BUT UAC outside my network still giving me the same problem as in the beginning. I constantly receive "OPTIONS"/SUBSCRIBE from softphone outside my network since it doesn't received ACK, thus generate error... I made a call which lasted 35 seconds and got cut off giving UAC of other party an error of network failure. I have produced WS trace, please look at it and guide me what is wrong with this situation, also my configuration is on the following link as of today... http://pastebin.com/m3cf2769e Thanks for all your help, -- Khan VoIP Rookie Every beginning has an end regardless we believe it or not... ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] help me about sql
Khanh, What model of opensips you are using? if you are using later version then 1.3 you will have file name db_mysql.so not "mysql.so" and if it is then your file should be one of the modules directory wherever you created for example /lib/opensips/modules/db_mysql.so if you cant find it try following command find / -name db_mysql.so Drop a line if it doesnt solve your issue. Khan 2009/5/2 khánh nguyễn : > hello > i'm very excited with ser project. > when i try to compile and install i found a pob with "make > prefix=/usr/local install include_modules="mysql" " > yes, i tried it but nothing happen and i can't see mysql.so in > /usr/local/lib/openser/modules. > could you tell me what my prob is and how to fix? > thanks you very much!!! > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Khan VoIP Rookie Every beginning has an end regardless we believe it or not... ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Problem running MediaProxy / FreeRADIS
Dear Users,
I have made bunch of changes in my opensips by upgrading to 1.5.1,
adding FreeRadius to my server, and MediaProxy 2.3.
Before making changes to config files for opensips it worked fine but
after making changes it gives me following errors... I am assuming i
have screwed up in putting proper values for OpenSIPS...
FreeRADIUS failure:
root# freeradius -X
radiusd: Opening IP addresses and Ports
listen {
type = "auth"
ipaddr = *
port = 0
}
listen {
type = "acct"
ipaddr = *
port = 0
}
listen {
type = "control"
listen {
socket = "/var/run/freeradius/freeradius.sock"
}
Failed binding to /var/run/freeradius/freeradius.sock: Permission denied
MediaProxy failure:
root# ./media-dispatcher --no-fork
Starting MediaProxy Dispatcher 2.3.4
Twisted is using epollreactor
fatal error: cannot read the RADIUS configuration file
fatal error: failed to create MediaProxy Dispatcher: 'servers'
Traceback (most recent call last):
--- ---
File "./media-dispatcher", line 56, in
dispatcher = Dispatcher()
File "/etc/mediaproxy/mediaproxy-2.3.4/mediaproxy/dispatcher.py",
line 523, in __init__
self.accounting =
[__import__("mediaproxy.interfaces.accounting.%s" % mod.lower(),
globals(), locals(), [""]).Accounting() for mod in
set(Config.accounting)]
File
"/etc/mediaproxy/mediaproxy-2.3.4/mediaproxy/interfaces/accounting/radius.py",
line 53, in __init__
self.radius = RadiusAccounting()
File
"/etc/mediaproxy/mediaproxy-2.3.4/mediaproxy/interfaces/accounting/radius.py",
line 74, in __init__
secrets = dict(line.rstrip("\n").split(None, 1) for line in
open(config["servers"]) if len(line.split(None, 1)) == 2 and not
line.startswith("#"))
exceptions.KeyError: 'servers'
--
Please HELP...
Khan
VoIP Rookie
Every beginning has an end regardless we believe it or not...
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Re: [OpenSIPS-Users] ACK timout OpenSIPS 1.5 Still not resolved
Deag Bogdan, I have force_rport() in the beginning of script as you can see in the link http://pastebin.com/mcec311 (highlighted section is where i added NAT traversal logic) also the log of failure is at this link <<< call made and ACK timed out >>> http://pastebin.com/m1d11246a I tried to figure out the problem, the highlighted parts might be the problem area if you could please give a quick look at see where in configuration script i went wrong? I know i am asking for too much but please help me, I really appreciate your help ! Khan On Fri, Apr 10, 2009 at 7:44 AM, Bogdan-Andrei Iancu wrote: > Hi Khan, > > The 401 is for a REGISTER (look at the Cseq header). > > anyhow, the lack of an ACK from the caller means the caller didn't received > the reply (200 OK). If the caller is behind a nat, be sure you do > force_rport() in script (at INVITE time) - this will correctly route back > the replies via the NAT. > > > Regards, > Bogdan > > Khan wrote: >> >> Ok, >> >> I guess I sort of see the problem but dont know how to fix it... i >> capture the trafic from the SjPhone UAC which transmit OPTIONS after >> 200 OK, it seems like its getting 401 from server on authentication, >> wonder why? >> >> Here is a link http://pastebin.com/m298ec8c6 >> please let me know if i am on the right track >> >> Thanks for all your time and efforts... >> >> Khan >> >> On Thu, Apr 9, 2009 at 11:00 AM, Khan wrote: >> >>> >>> On Thu, Apr 9, 2009 at 2:13 AM, Uwe Kastens wrote: >>> >>>> >>>> Khan, >>>> >>>> Would it be possible to add a tcpdump/wirshark on the opensips and on >>>> the client in the external network? That make it much easier to debug. >>>> >>> >>> I haven't done this before so, let me try to get the tcpdump for you, >>> I will install wireshark today (like i said im rookie) >>> I will post the tcpdump today :) >>> >>> >>>> >>>> One question: If you use xlite internaly, is the call dropped after >>>> 35secs or not? >>>> >>> >>> No, it only happens outside the network, I believe my NAT traversal >>> works fine, for some reasons my voice reaches them but theirs is lost >>> somewhere in clouds :) >>> >>> >>>> >>>> BR >>>> >>>> Uwe >>>> >>>> Khan schrieb: >>>> >>>>> >>>>> Uwe, >>>>> >>>>> I am using xlite within my network which works fine the problem is >>>>> outside the network, Xlite sends SUBSCRIBE and SJphone Sends OPTIONS >>>>> request... >>>>> >>>>> An example of debug is as follows, >>>>> >>>>> >>>>> Xlite registered fine the dump during the call process is as follows, >>>>> the call last for 35 seconds in which other party could hear me but i >>>>> can see a message on my Sjphone "ACK message awaiting" and then it >>>>> disconnects with the message "Network failure" please review the >>>>> following link... >>>>> http://pastebin.com/dca5bbb0 >>>>> >>>>> Another example is this SJphone which registers fine but after >>>>> registration constantly sends the OPTIONS requsts. The link is as >>>>> follows: >>>>> http://pastebin.com/d3a4fb379 >>>>> >>>>> My opensips.cfg is at this link: >>>>> http://pastebin.com/d6ce3e43d >>>>> >>>>> Thanks for all your help ... >>>>> >>>>> >>>>> Khan >>>>> >>>>> >>>>> >>>>> On Wed, Apr 8, 2009 at 1:46 PM, Uwe Kastens wrote: >>>>> >>>>>> >>>>>> Hi Khan, >>>>>> >>>>>> A easy way to debug this problem is to use a kind of network sniffer >>>>>> on >>>>>> your opensips and directly after your UA. Try to debug this issue with >>>>>> a >>>>>> softphone like xlite, so you can start your network dump on the >>>>>> client. >>>>>> >>>>>> BR >>>>>> >>>>>> Uwe >>>>>> >>>>>> Khan schrieb: >>>>>> >>>>>>> >>>>>>> Hi everyone, >>>
Re: [OpenSIPS-Users] OpenSIPS 1.5 crashes when starting
Thanks Bogdan for your prompt response, following is the version i am using
version: opensips 1.5.0-notls (x86_64/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST,
SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
svnrevision: unknown
@(#) $Id: main.c 5469 2009-03-18 12:43:10Z bogdan_iancu $
main.c compiled on 01:32:02 Mar 29 2009 with gcc 4.2.4
I made few attempts of restarting of opensips last night and
successfully started it back, thus it was running but i did not
perform any tests. I followed your instructions and ran the opensips
again still it didn't crash.
Do you still need back trace even if it doesn't crash?
How do i do it? last time i tried I didn't do it right, anyone like to
help me in this process step by step or direct me where to get info!!!
***
I need to understand why it crashed before?
Do i need to get newer version from SVN repository?
If I do, how would i do it because i never did it before...
I really really appreciate your patience with me and helping me out to
learn and implement this project.
Thank you all,
Khan
On Fri, Apr 10, 2009 at 7:49 AM, Bogdan-Andrei Iancu
wrote:
> Hi Khan,
>
> What version and what revision are you using (use opensips -V) ?
>
> Also, try to get a core file (set "ulimit -c unlimited" before starting
> opensips) and get the backtrace out of it.
>
> Regards,
> Bogdan
>
> Khan wrote:
>>
>> Hello all,
>>
>> I am running ubuntu server with mySQL server, OpenSIPS 1.5, RTPProxy 1.2
>>
>> Having a series of problems with OpenSIPS, my initial problem was
>> getting ACK timed out because of WWW Authentication failure of UAC
>> registration.
>> I am including a link for my traffic sniff and screen dump of debug
>> mode (the highlighted part is my guess of problem area)
>>
>> http://pastebin.com/m788119fb
>>
>> I made few changes in opensips.cfg to resolve the problem of WWW auth
>> failure as follows...
>>
>>
>> # Some systems (like Asterisk) use OPTIONS as a kind of "ping",
>> than we
>> # answer it with 200 OK.
>> if (method=="OPTIONS") {
>> xlog("L_INFO", "*** Method: $rm *** RURI: $ru \n");
>>
>> xlog("L_INFO", "*** Form: $fu To: $tu IP=$si SIP Request Port:
>> $rp \n");
>>
>>
>> if ((uri==myself) && (! uri=~"sip:@]+.*"))
>> {
>> options_reply();
>> } else {
>> sl_send_reply("200", "OK");
>> return;
>> }
>> }
>>
>> *** Also this part which was originally "" i put mydomain in quotes
>>
>> if (!proxy_authorize("mydomain.com", "subscriber")) {
>> xlog("L_INFO", "*** Proxy authentication \n");
>>
>> proxy_challenge("mydomain.com", "0");
>> exit;
>> }
>> if (!check_from()) {
>> xlog("L_INFO", "*** Form URI missing: $fu To: $tu IP=$si \n");
>>
>> sl_send_reply("403","Forbidden auth ID");
>> exit;
>> }
>>
>> ***
>> if (is_method("REGISTER"))
>> {
>> xlog("L_INFO", "*** 08 *** Register Authentication SIP Request
>> Port: $rp \n");
>> xlog("L_INFO", "*** Method: $rm RURI: $ru Form: $fu To: $tu IP=$si
>> \n");
>>
>> # authenticate the REGISTER requests (uncomment to enable
>> auth)
>> if (!www_authorize("mydomain.com", "subscriber"))
>> {
>> www_challenge("mydomain.com", "0");
>> exit;
>> }
>>
>> if (!check_to())
>> {
>> sl_send_reply("403","Forbidden auth ID");
>> exit;
>> }
>>
>> if (!save("location"))
>> sl_reply_error();
>>
>> exit;
>> }
>>
>> *** After making the above changes when i start my opensips it
>> crashes, the detailed out put is at this link
>> http://pastebin.com/m4adecddb
>>
>> Can someone please help me or guide me if i need to update my opensips
>> 1.5 because i see a lot of discussion going on about crashes and bug
>> fixes.
>>
>> Also can someone explain why am i getting ACK timed out and how to
>> resolve auth failure or its suppose to happen???
>>
>>
>> Please help
>>
>>
>> Khan
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
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[OpenSIPS-Users] OpenSIPS 1.5 crashes when starting
Hello all,
I am running ubuntu server with mySQL server, OpenSIPS 1.5, RTPProxy 1.2
Having a series of problems with OpenSIPS, my initial problem was
getting ACK timed out because of WWW Authentication failure of UAC
registration.
I am including a link for my traffic sniff and screen dump of debug
mode (the highlighted part is my guess of problem area)
http://pastebin.com/m788119fb
I made few changes in opensips.cfg to resolve the problem of WWW auth
failure as follows...
# Some systems (like Asterisk) use OPTIONS as a kind of "ping", than we
# answer it with 200 OK.
if (method=="OPTIONS") {
xlog("L_INFO", "*** Method: $rm *** RURI: $ru \n");
xlog("L_INFO", "*** Form: $fu To: $tu IP=$si SIP Request Port: $rp
\n");
if ((uri==myself) && (! uri=~"sip:@]+.*"))
{
options_reply();
} else {
sl_send_reply("200", "OK");
return;
}
}
*** Also this part which was originally "" i put mydomain in quotes
if (!proxy_authorize("mydomain.com", "subscriber")) {
xlog("L_INFO", "*** Proxy authentication \n");
proxy_challenge("mydomain.com", "0");
exit;
}
if (!check_from()) {
xlog("L_INFO", "*** Form URI missing: $fu To: $tu IP=$si \n");
sl_send_reply("403","Forbidden auth ID");
exit;
}
***
if (is_method("REGISTER"))
{
xlog("L_INFO", "*** 08 *** Register Authentication SIP Request
Port: $rp \n");
xlog("L_INFO", "*** Method: $rm RURI: $ru Form: $fu To: $tu IP=$si \n");
# authenticate the REGISTER requests (uncomment to enable auth)
if (!www_authorize("mydomain.com", "subscriber"))
{
www_challenge("mydomain.com", "0");
exit;
}
if (!check_to())
{
sl_send_reply("403","Forbidden auth ID");
exit;
}
if (!save("location"))
sl_reply_error();
exit;
}
*** After making the above changes when i start my opensips it
crashes, the detailed out put is at this link
http://pastebin.com/m4adecddb
Can someone please help me or guide me if i need to update my opensips
1.5 because i see a lot of discussion going on about crashes and bug
fixes.
Also can someone explain why am i getting ACK timed out and how to
resolve auth failure or its suppose to happen???
Please help
Khan
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[OpenSIPS-Users] ACK timout OpenSIPS 1.5 Still not resolved
Ok, I guess I sort of see the problem but dont know how to fix it... i capture the trafic from the SjPhone UAC which transmit OPTIONS after 200 OK, it seems like its getting 401 from server on authentication, wonder why? Here is a link http://pastebin.com/m298ec8c6 please let me know if i am on the right track Thanks for all your time and efforts... Khan On Thu, Apr 9, 2009 at 11:00 AM, Khan wrote: > On Thu, Apr 9, 2009 at 2:13 AM, Uwe Kastens wrote: >> Khan, >> >> Would it be possible to add a tcpdump/wirshark on the opensips and on >> the client in the external network? That make it much easier to debug. > > I haven't done this before so, let me try to get the tcpdump for you, > I will install wireshark today (like i said im rookie) > I will post the tcpdump today :) > >> >> One question: If you use xlite internaly, is the call dropped after >> 35secs or not? > > No, it only happens outside the network, I believe my NAT traversal > works fine, for some reasons my voice reaches them but theirs is lost > somewhere in clouds :) > >> >> BR >> >> Uwe >> >> Khan schrieb: >>> Uwe, >>> >>> I am using xlite within my network which works fine the problem is >>> outside the network, Xlite sends SUBSCRIBE and SJphone Sends OPTIONS >>> request... >>> >>> An example of debug is as follows, >>> >>> >>> Xlite registered fine the dump during the call process is as follows, >>> the call last for 35 seconds in which other party could hear me but i >>> can see a message on my Sjphone "ACK message awaiting" and then it >>> disconnects with the message "Network failure" please review the >>> following link... >>> http://pastebin.com/dca5bbb0 >>> >>> Another example is this SJphone which registers fine but after >>> registration constantly sends the OPTIONS requsts. The link is as >>> follows: >>> http://pastebin.com/d3a4fb379 >>> >>> My opensips.cfg is at this link: >>> http://pastebin.com/d6ce3e43d >>> >>> Thanks for all your help ... >>> >>> >>> Khan >>> >>> >>> >>> On Wed, Apr 8, 2009 at 1:46 PM, Uwe Kastens wrote: >>>> Hi Khan, >>>> >>>> A easy way to debug this problem is to use a kind of network sniffer on >>>> your opensips and directly after your UA. Try to debug this issue with a >>>> softphone like xlite, so you can start your network dump on the client. >>>> >>>> BR >>>> >>>> Uwe >>>> >>>> Khan schrieb: >>>>> Hi everyone, >>>>> >>>>> I'm rookie in SIP technology, strugling with several issues. I am >>>>> having problem with UAC's outside network. I have 3 UAC registered >>>>> within the network (SJ Phone, Xlite) they are working fine, I can talk >>>>> within the network but the problem arrise when I use the UAC outside >>>>> my network. I am seeing two different things from two different UAC's. >>>>> >>>>> 1. Xlite on a network behind NAT try to register, it registers >>>>> successfully after receiving 200 OK it starts senting SUBSCRIBE >>>>> requests, which results in 483 Erro (set up in my config) and when >>>>> call is placed on this it gives ACK time out, person on the other side >>>>> can hear me but i cant hear him. >>>>> >>>>> 2. SjPhone is on another network behind NAT, it regiesters fine, and >>>>> after registration it starts sending OPTIONS request, which I have >>>>> configured to respond as 200 OK. It continiously keep sending the >>>>> requst and my config respond to 200 OK. >>>>> >>>>> My question is several parts, what am I doing wrong, >>>>> a) why don't I get ACK after 200 OK, >>>>> b) how do i handle SUBSCRIBE requests >>>>> c) how do i handle OPTIONS request >>>>> >>>>> The sever is simply being used as SIP server for calls, no IM, Video, >>>>> or other applications are implemented yet. There are OpenSIPS, MySQL >>>>> server, and RTPProxy is running on the box. >>>>> >>>>> Please respond to my request considering my skills in the SIP as >>>>> rookie, guide me on how to resolve problem... >>>>> >>>>> Thanks, >>>>> >>>>> >>>>> Khan >>>>> >>>>> Sorry for such a long email, I am frustrated :( >>>>> >>>>> ___ >>>>> Users mailing list >>>>> Users@lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>> >>>> -- >>>> >>>> kiste lat: 54.322684, lon: 10.13586 >>>> >>> >> >> >> -- >> >> kiste lat: 54.322684, lon: 10.13586 >> > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] ACK timout OpenSIPS 1.5
Hi everyone, I'm rookie in SIP technology, strugling with several issues. I am having problem with UAC's outside network. I have 3 UAC registered within the network (SJ Phone, Xlite) they are working fine, I can talk within the network but the problem arrise when I use the UAC outside my network. I am seeing two different things from two different UAC's. 1. Xlite on a network behind NAT try to register, it registers successfully after receiving 200 OK it starts senting SUBSCRIBE requests, which results in 483 Erro (set up in my config) and when call is placed on this it gives ACK time out, person on the other side can hear me but i cant hear him. 2. SjPhone is on another network behind NAT, it regiesters fine, and after registration it starts sending OPTIONS request, which I have configured to respond as 200 OK. It continiously keep sending the requst and my config respond to 200 OK. My question is several parts, what am I doing wrong, a) why don't I get ACK after 200 OK, b) how do i handle SUBSCRIBE requests c) how do i handle OPTIONS request The sever is simply being used as SIP server for calls, no IM, Video, or other applications are implemented yet. There are OpenSIPS, MySQL server, and RTPProxy is running on the box. Please respond to my request considering my skills in the SIP as rookie, guide me on how to resolve problem... Thanks, Khan Sorry for such a long email, I am frustrated :( ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS 1.5 gives ERROR:core:init_mod: failed to initialize module registrar
OpenSIPS is not running due to errors, these errors are about registrar.so. The error occurs on starting opensips. I have 64bit box running ubuntu server, all the dependencies were installed before installing OpenSIPS, earlier i removed version 1.4.4 due to several crashes. Following is the link to screen output / opensips.cfg file : http://pastebin.com/m738bb3f8 The highlighted portion is the error part, I can not pin point the problem area, so please help me resolve this issue. I am new to this environment so i need all the help i can get. Thank you in advance for your time... Khan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS 1.5 gives error installing building db_mysql.so
l.dbtext ; \
install -m 644 /tmp/opensipsctl.dbtext \
/usr/local/lib64/opensips/opensipsctl/opensipsctl.dbtext ; \
rm -fr /tmp/opensipsctl.dbtext ; \
sed -e
"s#/usr/local/share/opensips#/usr/local/share/opensips/#g" \
< scripts/opensipsdbctl.dbtext >
/tmp/opensipsdbctl.dbtext ; \
touch
/usr/local/lib64/opensips/opensipsctl/opensipsdbctl.dbtext ; \
install -m 644 /tmp/opensipsdbctl.dbtext
/usr/local/lib64/opensips/opensipsctl/ ; \
rm -fr /tmp/opensipsdbctl.dbtext ; \
mkdir -p /usr/local/lib64/opensips/opensipsctl/dbtextdb
; \
touch
/usr/local/lib64/opensips/opensipsctl/dbtextdb/dbtextdb.py ; \
install -m 755 scripts/dbtextdb/dbtextdb.py
/usr/local/lib64/opensips/opensipsctl/dbtextdb/ ; \
mkdir -p /usr/local/share/opensips//dbtext/opensips ; \
for FILE in scripts/dbtext/opensips/acc
scripts/dbtext/opensips/active_watchers
scripts/dbtext/opensips/address scripts/dbtext/opensips/aliases
scripts/dbtext/opensips/carrierfailureroute
scripts/dbtext/opensips/carrierroute scripts/dbtext/opensips/cpl
scripts/dbtext/opensips/dbaliases scripts/dbtext/opensips/dialog
scripts/dbtext/opensips/dialplan scripts/dbtext/opensips/dispatcher
scripts/dbtext/opensips/domain scripts/dbtext/opensips/domainpolicy
scripts/dbtext/opensips/dr_gateways scripts/dbtext/opensips/dr_groups
scripts/dbtext/opensips/dr_rules
scripts/dbtext/opensips/globalblacklist scripts/dbtext/opensips/grp
scripts/dbtext/opensips/gw scripts/dbtext/opensips/imc_members
scripts/dbtext/opensips/imc_rooms scripts/dbtext/opensips/lcr
scripts/dbtext/opensips/load_balancer scripts/dbtext/opensips/location
scripts/dbtext/opensips/missed_calls
scripts/dbtext/opensips/nh_sockets scripts/dbtext/opensips/pdt
scripts/dbtext/opensips/presentity scripts/dbtext/opensips/pua
scripts/dbtext/opensips/re_grp scripts/dbtext/opensips/rls_presentity
scripts/dbtext/opensips/rls_watchers
scripts/dbtext/opensips/route_tree scripts/dbtext/opensips/silo
scripts/dbtext/opensips/sip_trace scripts/dbtext/opensips/speed_dial
scripts/dbtext/opensips/subscriber scripts/dbtext/opensips/trusted
scripts/dbtext/opensips/uri scripts/dbtext/opensips/userblacklist
scripts/dbtext/opensips/usr_preferences
scripts/dbtext/opensips/version scripts/dbtext/opensips/watchers
scripts/dbtext/opensips/xcap ; do \
if [ -f $FILE ] ; then \
touch $FILE \
/usr/local/share/opensips//dbtext/opensips/`basename "$FILE"` ; \
install -m 644 $FILE \
/usr/local/share/opensips//dbtext/opensips/`basename "$FILE"` ; \
fi ;\
done ;\
fi
ERROR: module modules/”db_mysql”/”db_mysql”.so not compiled
make[1]: Entering directory `/usr/src/opensips-1.5.0-tls/modules/acc'
make[1]: Nothing to be done for `install_module_custom'.
make[1]: Leaving directory `/usr/src/opensips-1.5.0-tls/modules/acc'
make[1]: Entering directory `/usr/src/opensips-1.5.0-tls/modules/alias_db'
make[1]: Nothing to be done for `install_module_custom'.
Thanks in advance...
Khan
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[OpenSIPS-Users] OpenSIPS 1.4.4 crashes When making call
Hi everyone, I just moved all my stuff from old box to new Xeon box. When i finished installing everything and applying rtpproxy on my sip proxy I get it crashed. I simply dialed one UAC to another UAC within the network. The call kept ringing on the other end even though i hung-up from my end. I answered the call from other end and had engage tone, secondly it wouldnt hangup. I looked at my log and i see my sip server crashed during this test. Following is the output at pastebin. http://pastebin.com/m7d97b0bd Please help and let me know what did i do wrong, should i go back to 1.4.2 ? Khan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Help proxy authentication/www_authroize
Hello everyone,
Can someone help me or point me in direction to resolve some
authenticaion errors.
Currently I m having problems with UAC outside network, it gives me
authentication problem. I need to understand the followings:
1. www_authorize("", "subscriber") checks in subscriber table but what
fields, what should exist in table, what parameters function passes
for mattching existence
2. proxy_authorize("", "subscriber") checks what in subscriber? what
column should exist in table and match to what
I'm getting 407, I checked subscriber table, it has records with
mydomain, user, pwd but i keep getting error since authentication keep
failing.
##
My xlog outputs:
New request and force_rport - M=REGISTER RURI=sip:mydomain.com
F=sip:10...@mydomain.com T=sip:10...@mydomain.com
My request... M=REGISTER sip:mydomain.com
method REGISTER from R0 ... R6 - M=REGISTER RURI=sip:mydomain.com
F=sip:10...@mydomain.com T=sip:10...@mydomain.com IP=
Register auth failed (subscribe) - M=REGISTER RURI=sip:mydomain.com
F=sip:10...@mydomain.com T=sip:10...@mydomain.com IP=
New request and force_rport - M=REGISTER RURI=sip:mydomain.com
F=sip:10...@mydomain.com T=sip:10...@mydomain.com IP=
My request... M=REGISTER sip:mydomain.com
method REGISTER from R0 ... R6 - M=REGISTER RURI=sip:mydomain.com
F=sip:10...@mydomain.com T=sip:10...@mydomain.com IP=
consume_credentials!!! - M=REGISTER RURI=sip:mydomain.com
F=sip:10...@mydomain.com T=sip:10...@mydomain.com IP=
Request Username in RURI - rU=
Registration successful 6 - M=REGISTER RURI=sip:mydomain.com
F=sip:10...@mydomain.com T=sip:10...@mydomain.com IP=
New request and force_rport - M=SUBSCRIBE RURI=sip:10...@mydomain.com
F=sip:10...@mydomain.com T=sip:10...@mydomain.com IP=
method is NOT REGISTER from R0 ... NAT test - M=SUBSCRIBE
RURI=sip:10...@mydomain.com F=sip:10...@mydomain.com
T=sip:10...@mydomain.c
om IP=
My request... M=SUBSCRIBE sip:10...@mydomain.com
Requested Service Unavailable PUBLISH/SUBSCRIBE/NOTIFY - M=SUBSCRIBE
New request and force_rport - M=INVITE RURI=sip:10...@mydomain.com
F=sip:10...@mydomain.com T=sip:10...@mydomain.com IP=
method is NOT REGISTER from R0 ... NAT test - M=INVITE
RURI=sip:10...@mydomain.com F=sip:10...@mydomain.com
T=sip:10...@mydomain.com
IP=
My request... M=INVITE sip:10...@mydomain.com
method INVITE from R0 ...R8 pre-set 24, 25 - M=INVITE
RURI=sip:10...@mydomain.com F=sip:10...@mydomain.com
T=sip:10...@mydomain.com I
P=
Proxy authentication failed R8- M=INVITE RURI=sip:10...@mydomain.com
F=sip:10...@mydomain.com T=sip:10...@mydomain.com IP=
New request and force_rport - M=INVITE RURI=sip:10...@mydomain.com
F=sip:10...@mydomain.com T=sip:10...@mydomain.com IP=
method is NOT REGISTER from R0 ... NAT test - M=INVITE
RURI=sip:10...@mydomain.com F=sip:10...@mydomain.com
T=sip:10...@mydomain.com
IP=
My request... M=INVITE sip:10...@mydomain.com
method INVITE from R0 ...R8 pre-set 24, 25 - M=INVITE
RURI=sip:10...@mydomain.com F=sip:10...@mydomain.com
T=sip:10...@mydomain.com I
P=
Proxy authentication failed R8- M=INVITE RURI=sip:10...@mydomain.com
F=sip:10...@mydomain.com T=sip:10...@mydomain.com IP=
New request and force_rport - M=INVITE RURI=sip:10...@mydomain.com
F=sip:10...@mydomain.com T=sip:10...@mydomain.com IP=
method is NOT REGISTER from R0 ... NAT test - M=INVITE
RURI=sip:10...@mydomain.com F=sip:10...@mydomain.com
T=sip:10...@mydomain.com
IP=
My request... M=INVITE sip:10...@mydomain.com
method INVITE from R0 ...R8 pre-set 24, 25 - M=INVITE
RURI=sip:10...@mydomain.com F=sip:10...@mydomain.com
T=sip:10...@mydomain.com I
P=
Proxy authentication failed R8- M=INVITE RURI=sip:10...@mydomain.com
F=sip:10...@mydomain.com T=sip:10...@mydomain.com IP=
New request and force_rport - M=SUBSCRIBE RURI=sip:10...@mydomain.com
F=sip:10...@mydomain.com T=sip:10...@mydomain.com IP=
method is NOT REGISTER from R0 ... NAT test - M=SUBSCRIBE
RURI=sip:10...@mydomain.com F=sip:10...@mydomain.com
T=sip:10...@mydomain.c
om IP=
My request... M=SUBSCRIBE sip:10...@mydomain.com
Requested Service Unavailable PUBLISH/SUBSCRIBE/NOTIFY - M=SUBSCRIBE
Khan
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[OpenSIPS-Users] Problem rtpproxy can't connect
Hi everyone, I am going through a series of errors when i added rtpproxy to my box. Currently I have OpenSIPS 1.4.4 installed and running on debian with mysql server 5.x I have 5 UA on various computers which include X-Lite and SJphone. After replacing the sipwize config file ( for OS 1.2) and trouble shoot it to make it upto date for ver 1.4.4 I restarted the openSIPS and start getting following errors: * Feb 8 20:43:12 [7490] ERROR:nathelper:send_rtpp_command: can't connect to RTP proxy Feb 8 20:43:12 [7490] ERROR:nathelper:send_rtpp_command: proxy does not respond, disable it Feb 8 20:43:12 [7490] WARNING:nathelper:rtpp_test: can't get version of the RTP proxy Feb 8 20:43:12 [7490] WARNING:nathelper:rtpp_test: support for RTP proxy has been disabled temporarily Feb 8 20:43:12 [7491] ERROR:nathelper:send_rtpp_command: can't connect to RTP proxy Feb 8 20:43:12 [7491] ERROR:nathelper:send_rtpp_command: proxy does not respond, disable it I have wireless router with the following ports (both tcp/udp) open on it: SSH ---> 22 5060 ---> 5060 8000 ---> 8000 rtp proxy -> 12000 to 13000 media proxy -> 5 to 6 (do i have to open any special port tcp/udp for rtpproxy? ) I'm also attaching the SIP trace and the command line dump for review. I'm unable to understand what did i do wrong. I'm rookie to VoIP so please help me grow in this technology. Any suggestion is appreciated, I'm out of ideas at this point... Thank you in advance... Khan r...@mydomain:/home# opensips restart Listening on udp: mydomain.com [192.168.1.2]:5060 udp: 127.0.0.1 [127.0.0.1]:5060 Aliases: udp: mydomain:5060 udp: localhost:5060 *: mydomain.com:* Feb 8 20:43:12 [7487] INFO:core:init_tcp: using epoll_lt as the TCP io watch method (auto detected) Feb 8 20:43:12 [7489] NOTICE:core:main: version: opensips 1.4.4-notls (i386/linux) Feb 8 20:43:12 [7489] INFO:core:main: using 32 Mb shared memory Feb 8 20:43:12 [7489] INFO:core:main: using 1 Mb private memory per process Feb 8 20:43:12 [7489] INFO:usrloc:ul_init_locks: locks array size 512 Feb 8 20:43:12 [7489] INFO:maxfwd:mod_init: initializing... Feb 8 20:43:12 [7489] INFO:tm:mod_init: TM - initializing... r...@mydomain:/home# Feb 8 20:43:12 [7489] INFO:xlog:mod_init: initializing... Feb 8 20:43:12 [7489] INFO:textops:mod_init: initializing... Feb 8 20:43:12 [7489] INFO:registrar:mod_init: initializing... Feb 8 20:43:12 [7489] INFO:sl:mod_init: Initializing StateLess engine Feb 8 20:43:12 [7489] INFO:auth:mod_init: initializing... Feb 8 20:43:12 [7489] INFO:auth_db:mod_init: initializing... Feb 8 20:43:12 [7489] INFO:avpops:avpops_init: initializing... Feb 8 20:43:12 [7489] INFO:alias_db:mod_init: initializing... Feb 8 20:43:12 [7489] INFO:acc:mod_init: initializing... Feb 8 20:43:12 [7489] INFO:textops:hname_fixup: using hdr type (16) instead of Feb 8 20:43:12 [7489] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 218 kb Feb 8 20:43:12 [7489] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 218 kb Feb 8 20:43:12 [7490] ERROR:nathelper:send_rtpp_command: can't connect to RTP proxy Feb 8 20:43:12 [7490] ERROR:nathelper:send_rtpp_command: proxy does not respond, disable it Feb 8 20:43:12 [7490] WARNING:nathelper:rtpp_test: can't get version of the RTP proxy Feb 8 20:43:12 [7490] WARNING:nathelper:rtpp_test: support for RTP proxy has been disabled temporarily Feb 8 20:43:12 [7491] ERROR:nathelper:send_rtpp_command: can't connect to RTP proxy Feb 8 20:43:12 [7491] ERROR:nathelper:send_rtpp_command: proxy does not respond, disable it Feb 8 20:43:12 [7491] WARNING:nathelper:rtpp_test: can't get version of the RTP proxy Feb 8 20:43:12 [7491] WARNING:nathelper:rtpp_test: support for RTP proxy has been disabled temporarily Feb 8 20:43:12 [7492] ERROR:nathelper:send_rtpp_command: can't connect to RTP proxy Feb 8 20:43:12 [7492] ERROR:nathelper:send_rtpp_command: proxy does not respond, disable it Feb 8 20:43:12 [7492] WARNING:nathelper:rtpp_test: can't get version of the RTP proxy Feb 8 20:43:12 [7492] WARNING:nathelper:rtpp_test: support for RTP proxy has been disabled temporarily Feb 8 20:43:12 [7493] ERROR:nathelper:send_rtpp_command: can't connect to RTP proxy Feb 8 20:43:12 [7493] ERROR:nathelper:send_rtpp_command: proxy does not respond, disable it Feb 8 20:43:12 [7493] WARNING:nathelper:rtpp_test: can't get version of the RTP proxy Feb 8 20:43:12 [7493] WARNING:nathelper:rtpp_test: support for RTP proxy has been disabled temporarily Feb 8 20:43:12 [7494] ERROR:nathelper:send_rtpp_command: can't connect to RTP proxy Feb 8 20:43:12 [7494] ERROR:nathelper:send_rtpp_command: proxy does not respond, disable it Feb 8 20:43:12 [7494] WARNING:nathelper:rtpp_test: can't get version of the RTP
[OpenSIPS-Users] Error no audio devices available
Hi everyone,
I have installed OpenSIPS 1.4.4 recently on a ubuntu 8.04, I have 3 UAC
configured. Two of them are SJphone which are within the same network. One
of them (Xlite) is outside the network and behind NAT. I am facing two
problems.
1. The SJphone within network works fine but user hear his own echo.
2. XLite outside network giving the error "Error no audio devices available"
The config file is as follows:
### Global Parameters #
fork=yes
children=4
/* uncomment the following lines to enable debugging */
debug=6
log_stderror=no
log_facility=LOG_LOCAL0
/* uncomment the next line to disable TCP (default on) */
disable_tcp=yes
/* uncomment the next line to enable the auto temporary blacklisting of
not available destinations (default disabled) */
#disable_dns_blacklist=no
/* uncomment the next line to enable IPv6 lookup after IPv4 dns
lookup failures (default disabled) */
#dns_try_ipv6=yes
port=5060
/* uncomment and configure the following line if you want opensips to
bind on a specific interface/port/proto (default bind on all available)
*/
listen=udp:mydomain.com:5060
listen=udp:999.999.1.2:5060
listen=udp:127.0.0.1:5060
#set module path
mpath="/usr/local/lib/opensips/modules/"
alias=mydomain.com
/* uncomment next line for MySQL DB support */
loadmodule "db_mysql.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "mi_fifo.so"
loadmodule "uri_db.so"
loadmodule "uri.so"
loadmodule "xlog.so"
loadmodule "acc.so"
/* uncomment next lines for MySQL based authentication support
NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule "auth.so"
loadmodule "auth_db.so"
/* uncomment next line for aliases support
NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule "alias_db.so"
/* uncomment next line for multi-domain support
NOTE: a DB (like db_mysql) module must be also loaded
NOTE: be sure and enable multi-domain support in all used modules
(see "multi-module params" section ) */
loadmodule "domain.so"
/* uncomment the next two lines for presence server support
NOTE: a DB (like db_mysql) module must be also loaded */
#loadmodule "presence.so"
#loadmodule "presence_xml.so"
# - setting module-specific parameters ---
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("rr", "enable_full_lr", 1)
modparam("rr", "append_fromtag", 0)
modparam("registrar", "method_filtering", 1)
modparam("uri_db", "use_uri_table", 0)
modparam("uri_db", "db_url", "")
modparam("acc", "early_media", 1)
modparam("acc", "report_ack", 1)
modparam("acc", "report_cancels", 1)
modparam("acc", "detect_direction", 0)
modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "log_flag", 1)
modparam("acc", "log_missed_flag", 2)
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 2)
modparam("usrloc", "db_mode", 2)
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
### Routing Logic
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
}
route(1);
} else {
/* uncomment the following lines if you want to enable presence
*/
##if (is_method("SUBSCRIBE") && $rd == "your.server.ip.address")
{
### in-dialog subscribe requests
##route(2);
##exit;
##}
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
t_relay();
exit;
} else {
# ACK without matching transaction ... ignore and
discard.\n");
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
#initial requests
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();
# account only INVITEs
if (is_method("INVITE")) {
setflag(1); # do accounting
}
if (!uri==myself)
/* replace with following line if multi-domain support is used */
##if (!is_uri_host_local())
{
append_hf("P-hint: outbound\r\n");
route(1);
}
# requests for my domain
/* uncomment this if you want to enable presence
[OpenSIPS-Users] Needs help OpenSIPS-cp
I am in process of installation of OCP, I read the installation instructions. Im running OpenSIPS 1.4.2. Installation instruction does not say anything about changing SQL or .sh file. The sample files refer to database OpenSIPS_1_4, I assumed that i need to rename them to my db which is opensips thus I created cdr table in my opensips db. The table acc does not have caller_id, callee_id, destination, and some other fields. I added those columns in opensips database table acc. I was wondering am i doing right so far? Why installation instructions doesn't talk about it? OR Do I have to create new database with the name opensips_1_4? Please help. Khan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Help please - OpenSIPS Runtime Errors
Hello All, I am in need of help, I did not receive any response from anyone. Please take a look at this problem and guide me, seriously I am frustrated :( May be there is a secret handshake for this list to get responses :) Khan On Mon, Jan 5, 2009 at 9:19 PM, Khan Friend wrote: > Hello everyone, > > I am having runtime errors in OpenSIPS, I have tried to locate the root of > the problem but being newbee I'm confused where does the problem lies. I > have Debian running with OpenSIPS, Asterisks, RTPproxy, MySQL database > running. > > I have debian machine running all those servers, when i do ifconfig i get > following response: > > eth0 Link encap:UNSPEC HWaddr > 00-60-1D-00-00-00-05-B2-00-00-00-00-00-00-00-00 > inet addr:192.168.1.2 Bcast:192.168.1.255 Mask:255.255.255.0 > UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 > RX packets:0 errors:0 dropped:0 overruns:0 frame:0 > TX packets:230 errors:0 dropped:0 overruns:0 carrier:0 > collisions:0 txqueuelen:1000 > RX bytes:0 (0.0 B) TX bytes:18984 (18.5 KiB) > > eth1 Link encap:Ethernet HWaddr 00:c0:49:63:64:a0 > inet addr:192.168.1.2 Bcast:192.168.1.255 Mask:255.255.255.0 > inet6 addr: fe80::2c0:49ff:fe63:64a0/64 Scope:Link > UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 > RX packets:22591 errors:0 dropped:0 overruns:0 frame:0 > TX packets:22471 errors:0 dropped:0 overruns:0 carrier:0 > collisions:0 txqueuelen:1000 > RX bytes:8050374 (7.6 MiB) TX bytes:3810685 (3.6 MiB) > Interrupt:169 Base address:0xd800 > > loLink encap:Local Loopback > inet addr:127.0.0.1 Mask:255.0.0.0 > inet6 addr: ::1/128 Scope:Host > UP LOOPBACK RUNNING MTU:16436 Metric:1 > RX packets:7184 errors:0 dropped:0 overruns:0 frame:0 > TX packets:7184 errors:0 dropped:0 overruns:0 carrier:0 > collisions:0 txqueuelen:0 > RX bytes:3347190 (3.1 MiB) TX bytes:3347190 (3.1 MiB) > > > when i register my X-lite within the network it registers fine. when i look > in log i see error > > Jan 5 20:38:15myosips[5021]: ERROR:core:forward_reply: no 2nd via found in > reply > > I tried changing script many ways, finally i put a simple script but still > facing same problem. Can somone tell me where is the problem. > > some of the things i see as problem is listed below but complet log is > attached. > > ### > > Jan 5 20:37:17myosips[5017]: DBG:core:db_do_init: connection 0x819ed30 not > found in pool > Jan 5 20:37:17myosips[5017]: DBG:db_mysql:db_mysql_new_connection: opening > connection: mysql://:x...@localhost/opensips > Jan 5 20:37:17myosips[5017]: DBG:db_mysql:db_mysql_new_connection: > connection type is Localhost via UNIX socket > Jan 5 20:37:17myosips[5017]: DBG:db_mysql:db_mysql_new_connection: > protocol version is 10 > Jan 5 20:37:17myosips[5017]: DBG:db_mysql:db_mysql_new_connection: server > version is 5.0.75-1 > Jan 5 20:37:17myosips[5017]: DBG:core:db_new_result: allocate 28 bytes for > result set at 0x819edf0 > Jan 5 20:37:17myosips[5017]: DBG:db_mysql:db_mysql_get_columns: 1 columns > returned from the query > Jan 5 20:37:17myosips[5017]: DBG:core:db_allocate_columns: allocate 4 > bytes for result names at 0x819ee18 > Jan 5 20:37:17myosips[5017]: DBG:core:db_allocate_columns: allocate 4 > bytes for result types at 0x819ee28 > Jan 5 20:37:17myosips[5017]: DBG:db_mysql:db_mysql_get_columns: allocate 8 > bytes for RES_NAMES[0] at 0x819ee38 > Jan 5 20:37:17myosips[5017]: DBG:db_mysql:db_mysql_get_columns: > RES_NAMES(0x819ee38)[0]=[table_version] > Jan 5 20:37:17myosips[5017]: DBG:db_mysql:db_mysql_get_columns: use DB_INT > result type > Jan 5 20:37:17myosips[5017]: DBG:db_mysql:db_mysql_convert_rows: allocate > 8 bytes for rows at 0x819ee48 > Jan 5 20:37:17myosips[5017]: DBG:db_mysql:db_mysql_convert_row: allocate > 20 bytes for row values at 0x819ee58 > Jan 5 20:37:17myosips[5017]: DBG:db_mysql:db_mysql_str2val: converting INT > [6] > > > Jan 5 20:38:15myosips[5021]: DBG:core:parse_msg: SIP Request: > Jan 5 20:38:15myosips[5021]: DBG:core:parse_msg: method: > Jan 5 20:38:15myosips[5021]: DBG:core:parse_msg: uri: domain.com> > Jan 5 20:38:15myosips[5021]: DBG:core:parse_msg: version: > Jan 5 20:38:15myosips[5021]: DBG:core:parse_headers: flags=2 > Jan 5 20:38:15myosips[5021]: DBG:core:parse_via_param: found param type > 232, = ; state=6 > Jan 5 20:38:15myosips[5021]: DBG:core:parse_via_param: found param type > 235, = ; state=17 > Jan 5 20:38:15myosips[5021]: DBG:core:parse_vi
[OpenSIPS-Users] Help please - OpenSIPS Runtime Errors
]: DBG:core:get_hdr_field: [64]; uri=[sip: domain.com] Jan 5 20:38:15myosips[5021]: DBG:core:get_hdr_field: to body [] Jan 5 20:38:15myosips[5021]: DBG:core:get_hdr_field: cseq : <102> Jan 5 20:38:15myosips[5021]: DBG:tm:t_reply_matching: failure to match a transaction Jan 5 20:38:15myosips[5021]: DBG:tm:t_check: end=(nil) Jan 5 20:38:15myosips[5021]: DBG:core:parse_headers: flags=4 Jan 5 20:38:15myosips[5021]: DBG:core:get_hdr_field: content_length=0 Jan 5 20:38:15myosips[5021]: DBG:core:get_hdr_field: found end of header Jan 5 20:38:15myosips[5021]: ERROR:core:forward_reply: no 2nd via found in reply Jan 5 20:38:15myosips[5021]: DBG:core:destroy_avp_list: destroying list (nil) Jan 5 20:38:15myosips[5021]: DBG:core:receive_msg: cleaning up ### Thank you, Khan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] moving openser 1.2 to Opensips
I would be interested in tutorial too, please keep me updated. -- Thank you, Khan On Mon, Jan 5, 2009 at 5:22 AM, Asim Riaz wrote: > Hi Bogdan, > could you please explain me what is involve in the migration from openser > 1.2 to opensips or send me link. > > Thanks, > Asim > > On Mon, Jan 5, 2009 at 11:16 AM, Bogdan-Andrei Iancu < > bog...@voice-system.ro> wrote: > >> Hi Asim, >> >> Sure it is possible - I was planing to write down a migration >> tutorial. >> >> Regards, >> Bogdan >> >> Asim Riaz wrote: >> >>> >>> >>> >>> Hi, >>> could anyone advice me, if its possible to move from openser 1.2 to >>> opensips ? >>> >>> >>> Thanks in Advance. >>> >>> -Asim >>> >>> >>> >>> ___ >>> 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] OpenSIPS is not running, Erorr
Bogdan,
The problem is that I don't know much about SIP server and VoIP. This is
experimental project, I studied and successfully ran simple OpenSIPS server.
When I try to add Asterisk or NAT Traversal, I ran into many problems. One
of them is this (Asterisk config), I traced the log file but not much luck
understanding what part needs fixing.
Please help me identify the root of the problem and how to fix. How do i
find SIP replies, what do i do to see them and capture them.
Thanks in advance,
Khan
On Sun, Dec 28, 2008 at 4:33 AM, Bogdan-Andrei Iancu wrote:
> Hi Khan,
>
> your OpenSIPS runs ok - what you see are runtime errors, not startup
> errors.
>
> The errors you see are indicating processing of SIP reply messages that
> could not be routed - they were received with only one VIA and they were not
> matching any local transaction.
>
> Can you identify the SIP replies triggering this error?
>
> Regards,
> Bogdan
>
> Khan Friend wrote:
>
>> Hi guys,
>>
>> I am trying to troubleshoot errors in my OpenSIPS config file but unable
>> to understand what am i doing wrong.
>>
>> The log file shows as follows:
>> Dec 26 21:38:02 [22302] INFO:usrloc:ul_init_locks: locks array size 512
>> Dec 26 21:38:02 [22302] INFO:registrar:mod_init: initializing...
>> Dec 26 21:38:02 [22302] INFO:textops:mod_init: initializing...
>> Dec 26 21:38:02 [22302] INFO:avpops:avpops_init: initializing...
>> Dec 26 21:38:02 [22302] INFO:auth:mod_init: initializing...
>> Dec 26 21:38:02 [22302] INFO:auth_db:mod_init: initializing...
>> Dec 26 21:38:02 [22302] INFO:core:probe_max_receive_buffer: using a UDP
>> receive buffer of 214 kb
>> Dec 26 21:38:56 [22303] ERROR:core:forward_reply: no 2nd via found in
>> reply
>> Dec 26 21:38:57 [22308] ERROR:core:forward_reply: no 2nd via found in
>> reply
>> Dec 26 21:38:58 [22306] ERROR:core:forward_reply: no 2nd via found in
>> reply
>> Dec 26 21:38:59 [22304] ERROR:core:forward_reply: no 2nd via found in
>> reply
>> Dec 26 21:39:00 [22303] ERROR:core:forward_reply: no 2nd via found in
>> reply
>> Dec 26 21:39:10 [22308] ERROR:core:forward_reply: no 2nd via found in
>> reply
>> Dec 26 21:39:11 [22306] ERROR:core:forward_reply: no 2nd via found in
>> reply
>> D
>>
>> --
>>
>>
>> My opensips.cfg is as follows:
>>
>> route{
>>
>># initial sanity checks -- messages with
>># max_forwards==0, or excessively long requests
>>
>>if (!mf_process_maxfwd_header("10")) {
>>sl_send_reply("483","Too Many Hops");
>>exit;
>>};
>>
>>if (msg:len >= 2048 ) {
>>sl_send_reply("513", "Message too big");
>>exit;
>>};
>>
>># we record-route all messages -- to make sure that
>># subsequent messages will go through our proxy; that's
>># particularly good if upstream and downstream entities
>># use different transport protocol
>>
>>if (!method=="REGISTER")
>>record_route();
>>
>># subsequent messages withing a dialog should take the
>># path determined by record-routing
>>
>>if (loose_route()) {
>># mark routing logic in request
>>append_hf("P-hint: rr-enforced\r\n");
>>route(1);
>>};
>>
>>if (!uri==myself) {
>># mark routing logic in request
>>append_hf("P-hint: outbound\r\n");
>>route(1);
>>};
>>
>># if the request is for other domain use UsrLoc
>># (in case, it does not work, use the following command
>># with proper names and addresses in it)
>>if (uri==myself) {
>>
>>if (method=="REGISTER") {
>>if (!www_authorize("", "subscriber")) {
>>www_challenge("", "0");
>>exit;
>>};
>>
>>save("location");
>>exit;
>>};
>>
>># requests for Media server
>>if(is_method("INVITE") && !has_totag() && uri=~"sip:\*9") {
>>route(3);
>>exit;
>>}
>>
>># mark transaction if user is in voicemail group
>>if(is_method("INVITE") && !has_totag()
>>&& is_user_in("Request-URI","voicemail"))
>>{
>>xdbg("user [
[OpenSIPS-Users] OpenSIPS is not running, Erorr
Hi guys,
I am trying to troubleshoot errors in my OpenSIPS config file but unable to
understand what am i doing wrong.
The log file shows as follows:
Dec 26 21:38:02 [22302] INFO:usrloc:ul_init_locks: locks array size 512
Dec 26 21:38:02 [22302] INFO:registrar:mod_init: initializing...
Dec 26 21:38:02 [22302] INFO:textops:mod_init: initializing...
Dec 26 21:38:02 [22302] INFO:avpops:avpops_init: initializing...
Dec 26 21:38:02 [22302] INFO:auth:mod_init: initializing...
Dec 26 21:38:02 [22302] INFO:auth_db:mod_init: initializing...
Dec 26 21:38:02 [22302] INFO:core:probe_max_receive_buffer: using a UDP
receive buffer of 214 kb
Dec 26 21:38:56 [22303] ERROR:core:forward_reply: no 2nd via found in reply
Dec 26 21:38:57 [22308] ERROR:core:forward_reply: no 2nd via found in reply
Dec 26 21:38:58 [22306] ERROR:core:forward_reply: no 2nd via found in reply
Dec 26 21:38:59 [22304] ERROR:core:forward_reply: no 2nd via found in reply
Dec 26 21:39:00 [22303] ERROR:core:forward_reply: no 2nd via found in reply
Dec 26 21:39:10 [22308] ERROR:core:forward_reply: no 2nd via found in reply
Dec 26 21:39:11 [22306] ERROR:core:forward_reply: no 2nd via found in reply
D
--
My opensips.cfg is as follows:
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
exit;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (!method=="REGISTER")
record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
if (!www_authorize("", "subscriber")) {
www_challenge("", "0");
exit;
};
save("location");
exit;
};
# requests for Media server
if(is_method("INVITE") && !has_totag() && uri=~"sip:\*9") {
route(3);
exit;
}
# mark transaction if user is in voicemail group
if(is_method("INVITE") && !has_totag()
&& is_user_in("Request-URI","voicemail"))
{
xdbg("user [$ru] has voicemail redirection enabled\n");
# backup R-URI
avp_pushto("$ru","$avp(i:10)");
#avp_write("$ruri","$avp(i:10)");
setflag(2);
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
if(isflagset(2)) {
# route to Asterisk Media Server
prefix("1");
rewritehostport("192.168.1.11:5060");
route(1);
} else {
sl_send_reply("404", "Not Found");
exit;
}
};
append_hf("P-hint: usrloc applied\r\n");
};
route(1);
}
route[1] {
if(isflagset(2))
t_on_failure("1");
if (!t_relay()) {
sl_reply_error();
};
exit;
}
# voicemail access
# - *98 - listen caller's voice messages, being prompted for pin
# - *981 - listen voice messages, being promted for mailbox and pin
# - *98 - leave voice message to
#
route[3] {
# direct voicemail
if (uri =~ "sip:\*98@" ) {
rewriteuser("1");
xdbg("voicemail access\n");
} else if (uri =~ "sip:\*981@" ) {
strip(4);
rewriteuser("11");
} else if (uri =~ "sip:\*98.+@" ) {
strip(3);
prefix("1");
} else {
xlog("unknown media extension $rU\n");
sl_send_reply("404", "Unknown media service");
exit;
}
# route to Asterisk Media Server
rewritehostport("192.168.1.11:5060");
route(1);
}
failure_route[1] {
if (t_was_cancelled()) {
xdbg("transaction was cancelled by UAC\n");
return;
}
# restore initial uri
avp_pushto("$ru","$avp(i:10)");
#avp_pushto("$ru", "i:10");
prefix("1");
# route to Asterisk Media Server
rewritehostport("192.168.1.11:5060");
resetflag(2);
route(1);
}
Thank you,
Khan
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Needs help OpenSIPS NAT Traversal Using RTPProxy
I have OpenSIPS Server - Installed and functional on Debian, Problem presist in NAT traversal area (RTPProxy, Asterisk) needs fine tunning... Authentication is done with MySQL (digest authentication with qop=auth), Managing multiple domains, alternative routes, inbound-to-outbond, outbond-to-inbound but no inter-domain communications, and domain authentication is through database. User Portal - No GUI user portal, using OPENSIPSCTL at command line at this point (please suggest any free opensource sysem) Asterisk - Connectivity to the PSTN Gateway. Use of LCR/CR module, securing re-invites, blacklists, Voice mail (Media Server Component AVM) Call forwarding AVP. For SIP NAT Traversal (RTPProxy). Accounting and Billing, MySQL server, RADIUS server, and rate calls CDRTool (not installed at present but will be installed before testing configuration script). OpenSIPS configuration file needed for a SIP Server (OpenSIPS 1.4) on Debian running with Asterisk behind NAT, RTPproxy 1.1, and MySQL server 5.x. Asterisk used to monitor, voice mail, agent's caller id should be changed to DID number. If anyone calls, we should be able to see their DID/User ID as a caller's number/name. It should be changed via OpenSIPS, not via Asterisk. I would like to authenticate all users and load balancing via FreeRADIUS . Currently I can make successful calls within the network but I have not checked the loadbalancing and Asterisk part, which should run as mentioned above with an accounting system. Problem: Having trouble configuring NAT traversal part, FreeRADIUS is not installed or included in configuration, and also the Asterisk part is not configured or checked. Since, I'm stuck in NAT Traversal thus above mentioned software is not added in configuration but required in the requested configuration. Following is the output when i try to run OpenSIPS, curently it stops running after few seconds and spits out these messages. Nov 29 17:32:16 [3542] WARNING:core:main: no fork mode Nov 29 17:32:16 [3542] INFO:core:init_tcp: using epoll_lt as the TCP io watch method (auto detected) Nov 29 17:32:16 [3542] NOTICE:core:main: version: opensips 1.4.2-notls (i386/linux) Nov 29 17:32:16 [3542] INFO:core:main: using 128 Mb shared memory Nov 29 17:32:16 [3542] INFO:core:main: using 1 Mb private memory per process Nov 29 17:32:16 [3542] INFO:usrloc:ul_init_locks: locks array size 512 Nov 29 17:32:16 [3542] INFO:tm:mod_init: TM - initializing... Nov 29 17:32:16 [3542] INFO:xlog:mod_init: initializing... Nov 29 17:32:16 [3542] INFO:dialog:mod_init: Dialog module - initializing Nov 29 17:32:16 [3542] INFO:textops:mod_init: initializing... Nov 29 17:32:16 [3542] INFO:sl:mod_init: Initializing StateLess engine Nov 29 17:32:16 [3542] INFO:registrar:mod_init: initializing... Nov 29 17:32:16 [3542] INFO:maxfwd:mod_init: initializing... Nov 29 17:32:16 [3542] INFO:auth:mod_init: initializing... Nov 29 17:32:16 [3542] INFO:auth_db:mod_init: initializing... Nov 29 17:32:16 [3542] INFO:avpops:avpops_init: initializing... Nov 29 17:32:16 [3542] INFO:alias_db:mod_init: initializing... Nov 29 17:32:16 [3542] INFO:acc:mod_init: initializing... Nov 29 17:32:16 [3542] INFO:textops:hname_fixup: using hdr type (16) instead of Proxy-Authorization Nov 29 17:32:16 [3542] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 214 kb AFTER FEW SECONDS FOLLOWING ERRORS POPS UP: New request - M=OPTIONS RURI=sip:mydomain.com F=sip:aster...@192.168.x.xt=sip: mydomain.com IP=192.168.x.x id=50fc878a344d621149e5cb2d4dba0...@192.168.x.x Method not supported - M=OPTIONS RURI=sip:mydomain.comF=sip:aster...@192.168.x.xt=sip: mydomain.com IP=192.168.x.x id=50fc878a344d621149e5cb2d4dba0...@192.168.x.x Nov 29 17:32:16 [3424] ERROR:core:forward_reply: no 2nd via found in reply New request - M=OPTIONS RURI=sip:mydomain.com F=sip:aster...@192.168.x.xt=sip: mydomain.com IP=192.168.x.x id=50fc878a344d621149e5cb2d4dba0...@192.168.x.x Method not supported - M=OPTIONS RURI=sip:mydomain.comF=sip:aster...@192.168.x.x T=sip:mydomain.com IP=192.168.x.x id=50fc878a344d621149e5cb2d4dba0...@192.168.x.x Nov 29 17:32:16 [3424] ERROR:core:forward_reply: no 2nd via found in reply ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
