[OpenSIPS-Users] alias_db failing to look up alias for IP using multi_domain
Using the alias_db module, if I look up an alias by the IP address as
the domain, it doesn't work. The alias table however does not let me
add the alias as an IP address as well as a domain.
Error message is: DBG:alias_db:alias_db_lookup: no alias found for R-URI
Example settings:
Server: a.domain.com
IP of server: 1.2.3.4
User: 1...@a.domain.com
Alias: +18085551...@a.domain.com
A call to +18085551...@1.2.3.4 fails when using alias_db_lookup("dbaliases");.
I also can't add both aliases to the dbaliases table:
# opensipsctl alias_db add +18085551...@1.2.3.4 1...@a.domain.com
INFO: +18085551234 alias already in dbaliases table
Thanks,
Julian
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Re: [OpenSIPS-Users] alias_db failing to look up alias for IP using multi_domain
So my only solution is then pass through +18085551...@a.domain.com
(which isn't feasible) or to disable multi-domain?
Is there a way I can accept both +18085551...@a.domain.com and
+18085551...@1.2.3.4?
- Julian
On Sun, Jan 4, 2009 at 5:50 PM, Alex Balashov wrote:
> Alias_db has a literal - not a virtual (i.e. uri == myself style, aka all
> DNS aliases and locally homed IP interfaces) - approach to domains, as does
> auth_db and others.
>
> One domain, and it must literally match the one in the RURI.
>
> Julian Yap wrote:
>
>> Using the alias_db module, if I look up an alias by the IP address as
>> the domain, it doesn't work. The alias table however does not let me
>> add the alias as an IP address as well as a domain.
>>
>> Error message is: DBG:alias_db:alias_db_lookup: no alias found for R-URI
>>
>> Example settings:
>> Server: a.domain.com
>> IP of server: 1.2.3.4
>>
>> User: 1...@a.domain.com
>>
>> Alias: +18085551...@a.domain.com
>>
>> A call to +18085551...@1.2.3.4 fails when using
>> alias_db_lookup("dbaliases");.
>>
>> I also can't add both aliases to the dbaliases table:
>> # opensipsctl alias_db add +18085551...@1.2.3.4 1...@a.domain.com
>> INFO: +18085551234 alias already in dbaliases table
>>
>> Thanks,
>> Julian
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (678) 237-1775
>
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Re: [OpenSIPS-Users] alias_db failing to look up alias for IP using multi_domain
I just tried manually inputting straight in to the DB and it works for me.
I guess that solves my issue.
Thanks,
Julian
On Sun, Jan 4, 2009 at 5:58 PM, Alex Balashov wrote:
> Have you tried adding both combinations to the DB manually without using
> opensipsctl?
>
> On Jan 4, 2009, at 10:55 PM, "Julian Yap" wrote:
>
>> So my only solution is then pass through +18085551...@a.domain.com
>> (which isn't feasible) or to disable multi-domain?
>>
>> Is there a way I can accept both +18085551...@a.domain.com and
>> +18085551...@1.2.3.4?
>>
>> - Julian
>>
>> On Sun, Jan 4, 2009 at 5:50 PM, Alex Balashov
>> wrote:
>>>
>>> Alias_db has a literal - not a virtual (i.e. uri == myself style, aka all
>>> DNS aliases and locally homed IP interfaces) - approach to domains, as
>>> does
>>> auth_db and others.
>>>
>>> One domain, and it must literally match the one in the RURI.
>>>
>>> Julian Yap wrote:
>>>
>>>> Using the alias_db module, if I look up an alias by the IP address as
>>>> the domain, it doesn't work. The alias table however does not let me
>>>> add the alias as an IP address as well as a domain.
>>>>
>>>> Error message is: DBG:alias_db:alias_db_lookup: no alias found for R-URI
>>>>
>>>> Example settings:
>>>> Server: a.domain.com
>>>> IP of server: 1.2.3.4
>>>>
>>>> User: 1...@a.domain.com
>>>>
>>>> Alias: +18085551...@a.domain.com
>>>>
>>>> A call to +18085551...@1.2.3.4 fails when using
>>>> alias_db_lookup("dbaliases");.
>>>>
>>>> I also can't add both aliases to the dbaliases table:
>>>> # opensipsctl alias_db add +18085551...@1.2.3.4 1...@a.domain.com
>>>> INFO: +18085551234 alias already in dbaliases table
>>>>
>>>> Thanks,
>>>> Julian
>>>>
>>>> ___
>>>> Users mailing list
>>>> Users@lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>> --
>>> Alex Balashov
>>> Evariste Systems
>>> Web: http://www.evaristesys.com/
>>> Tel: (+1) (678) 954-0670
>>> Direct : (+1) (678) 954-0671
>>> Mobile : (+1) (678) 237-1775
>>>
>
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Re: [OpenSIPS-Users] alias_db failing to look up alias for IP using multi_domain
There are no config options in opensipsctlrc to be multi-domain aware
so I would say that opensipsctl is broken in this respect.
The relevant IF statement in opensipsctl only looks up the 'user'
portion to check that it is unique:
if is_value_in_db $DA_TABLE $DA_ALIAS_USER_COLUMN $TMP_OSIPSUSER; then
minfo "$TMP_OSIPSUSER alias already in $DA_TABLE table"
exit 0
fi
On Sun, Jan 4, 2009 at 6:11 PM, Alex Balashov wrote:
> I'm not sure if opensipsctl is broken in this respect or if you have to make
> it aware that you're doing multi-domain support via its config file. But
> yes, when in doubt, manipulate the raw database.
>
> Julian Yap wrote:
>
>> I just tried manually inputting straight in to the DB and it works for me.
>>
>> I guess that solves my issue.
>>
>> Thanks,
>> Julian
>>
>> On Sun, Jan 4, 2009 at 5:58 PM, Alex Balashov
>> wrote:
>>>
>>> Have you tried adding both combinations to the DB manually without using
>>> opensipsctl?
>>>
>>> On Jan 4, 2009, at 10:55 PM, "Julian Yap" wrote:
>>>
>>>> So my only solution is then pass through +18085551...@a.domain.com
>>>> (which isn't feasible) or to disable multi-domain?
>>>>
>>>> Is there a way I can accept both +18085551...@a.domain.com and
>>>> +18085551...@1.2.3.4?
>>>>
>>>> - Julian
>>>>
>>>> On Sun, Jan 4, 2009 at 5:50 PM, Alex Balashov
>>>>
>>>> wrote:
>>>>>
>>>>> Alias_db has a literal - not a virtual (i.e. uri == myself style, aka
>>>>> all
>>>>> DNS aliases and locally homed IP interfaces) - approach to domains, as
>>>>> does
>>>>> auth_db and others.
>>>>>
>>>>> One domain, and it must literally match the one in the RURI.
>>>>>
>>>>> Julian Yap wrote:
>>>>>
>>>>>> Using the alias_db module, if I look up an alias by the IP address as
>>>>>> the domain, it doesn't work. The alias table however does not let me
>>>>>> add the alias as an IP address as well as a domain.
>>>>>>
>>>>>> Error message is: DBG:alias_db:alias_db_lookup: no alias found for
>>>>>> R-URI
>>>>>>
>>>>>> Example settings:
>>>>>> Server: a.domain.com
>>>>>> IP of server: 1.2.3.4
>>>>>>
>>>>>> User: 1...@a.domain.com
>>>>>>
>>>>>> Alias: +18085551...@a.domain.com
>>>>>>
>>>>>> A call to +18085551...@1.2.3.4 fails when using
>>>>>> alias_db_lookup("dbaliases");.
>>>>>>
>>>>>> I also can't add both aliases to the dbaliases table:
>>>>>> # opensipsctl alias_db add +18085551...@1.2.3.4 1...@a.domain.com
>>>>>> INFO: +18085551234 alias already in dbaliases table
>>>>>>
>>>>>> Thanks,
>>>>>> Julian
>>>>>>
>>>>>> ___
>>>>>> Users mailing list
>>>>>> Users@lists.opensips.org
>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>> --
>>>>> Alex Balashov
>>>>> Evariste Systems
>>>>> Web: http://www.evaristesys.com/
>>>>> Tel: (+1) (678) 954-0670
>>>>> Direct : (+1) (678) 954-0671
>>>>> Mobile : (+1) (678) 237-1775
>>>>>
>
>
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (678) 237-1775
>
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Re: [OpenSIPS-Users] alias_db failing to look up alias for IP using multi_domain
FYI, I have filed Bug ID: 2487485.
Direct link:
http://sourceforge.net/tracker/index.php?func=detail&aid=2487485&group_id=232389&atid=1086410
- Julian
On Sun, Jan 4, 2009 at 6:40 PM, Sergio Gutierrez wrote:
> Hi Julian.
>
> Please post the bug report on the tracker, so that we can fix it.
>
> Thanks and regards.
>
> Sergio G.
>
> On Sun, Jan 4, 2009 at 11:31 PM, Alex Balashov
> wrote:
>>
>> Aha. Well, I guess it could make for an edifying bug report.
>>
>> Julian Yap wrote:
>>
>> > There are no config options in opensipsctlrc to be multi-domain aware
>> > so I would say that opensipsctl is broken in this respect.
>> >
>> > The relevant IF statement in opensipsctl only looks up the 'user'
>> > portion to check that it is unique:
>> > if is_value_in_db $DA_TABLE $DA_ALIAS_USER_COLUMN $TMP_OSIPSUSER; then
>> > minfo "$TMP_OSIPSUSER alias already in $DA_TABLE table"
>> > exit 0
>> > fi
>> >
>> > On Sun, Jan 4, 2009 at 6:11 PM, Alex Balashov
>> > wrote:
>> >> I'm not sure if opensipsctl is broken in this respect or if you have to
>> >> make
>> >> it aware that you're doing multi-domain support via its config file.
>> >> But
>> >> yes, when in doubt, manipulate the raw database.
>> >>
>> >> Julian Yap wrote:
>> >>
>> >>> I just tried manually inputting straight in to the DB and it works for
>> >>> me.
>> >>>
>> >>> I guess that solves my issue.
>> >>>
>> >>> Thanks,
>> >>> Julian
>> >>>
>> >>> On Sun, Jan 4, 2009 at 5:58 PM, Alex Balashov
>> >>>
>> >>> wrote:
>> >>>> Have you tried adding both combinations to the DB manually without
>> >>>> using
>> >>>> opensipsctl?
>> >>>>
>> >>>> On Jan 4, 2009, at 10:55 PM, "Julian Yap"
>> >>>> wrote:
>> >>>>
>> >>>>> So my only solution is then pass through +18085551...@a.domain.com
>> >>>>> (which isn't feasible) or to disable multi-domain?
>> >>>>>
>> >>>>> Is there a way I can accept both +18085551...@a.domain.com and
>> >>>>> +18085551...@1.2.3.4?
>> >>>>>
>> >>>>> - Julian
>> >>>>>
>> >>>>> On Sun, Jan 4, 2009 at 5:50 PM, Alex Balashov
>> >>>>>
>> >>>>> wrote:
>> >>>>>> Alias_db has a literal - not a virtual (i.e. uri == myself style,
>> >>>>>> aka
>> >>>>>> all
>> >>>>>> DNS aliases and locally homed IP interfaces) - approach to domains,
>> >>>>>> as
>> >>>>>> does
>> >>>>>> auth_db and others.
>> >>>>>>
>> >>>>>> One domain, and it must literally match the one in the RURI.
>> >>>>>>
>> >>>>>> Julian Yap wrote:
>> >>>>>>
>> >>>>>>> Using the alias_db module, if I look up an alias by the IP address
>> >>>>>>> as
>> >>>>>>> the domain, it doesn't work. The alias table however does not let
>> >>>>>>> me
>> >>>>>>> add the alias as an IP address as well as a domain.
>> >>>>>>>
>> >>>>>>> Error message is: DBG:alias_db:alias_db_lookup: no alias found for
>> >>>>>>> R-URI
>> >>>>>>>
>> >>>>>>> Example settings:
>> >>>>>>> Server: a.domain.com
>> >>>>>>> IP of server: 1.2.3.4
>> >>>>>>>
>> >>>>>>> User: 1...@a.domain.com
>> >>>>>>>
>> >>>>>>> Alias: +18085551...@a.domain.com
>> >>>>>>>
>> >>>>>>> A call to +18085551...@1.2.3.4 fails when using
>> >>>>>>> alias_db_lookup("dbaliases");.
>> >>>>>>>
>> >>>>>>> I also can't add both aliases to the dbaliases table:
>> >>>>>>> # opensipsctl alias_db add +18085551...@1.2.3.4 1...@a.domain.com
>> >>>>>>> INFO: +18085551234 alias already in dbaliases table
>> >>>>>>>
>> >>>>>>> Thanks,
>> >>>>>>> Julian
>> >>>>>>>
>> >>>>>>> ___
>> >>>>>>> Users mailing list
>> >>>>>>> Users@lists.opensips.org
>> >>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> >>>>>> --
>> >>>>>> Alex Balashov
>> >>>>>> Evariste Systems
>> >>>>>> Web: http://www.evaristesys.com/
>> >>>>>> Tel: (+1) (678) 954-0670
>> >>>>>> Direct : (+1) (678) 954-0671
>> >>>>>> Mobile : (+1) (678) 237-1775
>> >>>>>>
>> >>
>> >> --
>> >> Alex Balashov
>> >> Evariste Systems
>> >> Web: http://www.evaristesys.com/
>> >> Tel: (+1) (678) 954-0670
>> >> Direct : (+1) (678) 954-0671
>> >> Mobile : (+1) (678) 237-1775
>> >>
>>
>>
>> --
>> Alex Balashov
>> Evariste Systems
>> Web: http://www.evaristesys.com/
>> Tel: (+1) (678) 954-0670
>> Direct : (+1) (678) 954-0671
>> Mobile : (+1) (678) 237-1775
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> --
> Sergio Gutiérrez
>
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[OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT
Hi All, I was looking to get some assistance with debugging an issue with an AudioCodes Mediant 2000 and OpenSIPS 1.4.4. I think it's a simple config issue. Basically, I'm getting one way audio :( Not sure if I need to be running RTPProxy to make this all work for me. The basic flow is: UA --> OpenSIPS --> AudioCodes G/W --> PSTN Here's a Pastebin of my OpenSIPS config: http://pastebin.com/m41d16a22 Here's a Pastebin of an Ngrep trace on the OpenSIPS server: http://pastebin.com/m7944a8d8 Private info changed in Pastebin's :). Any help appreciated! - Julian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT
Thanks all. I'll check to see if the AudioCodes gateway does have comedia support. That clarifies some half baked NAT/RTP knowledge in my head. - Julian On 2/10/09, Bogdan-Andrei Iancu wrote: > Hi Olle, > > Johansson Olle E wrote: >> >> 10 feb 2009 kl. 12.25 skrev Iñaki Baz Castillo: >> >>> 2009/2/10 : > You don't know if RtpProxy should be running, does it mean you are > trying to use it or not? I don't want to spend time inspecting what > you want to do by reading your config, sorry. Yeah, I'm trying not to run RTPProxy. After more testing, I'm thinking I may need to. >>> >>> You cannot decide if you need RtpProxy or not based on testing, it's >>> pure theory: >>> >>> A RTP proxy is NOT needed when (assuming the proxy has in the public >>> internet): >>> >>> - Both caller and callee have public IP or use STUN. >>> - Both caller and callee are in the *SAME* private LAN. >>> - The caller is in a private LAN and the callee has public IP and >>> supports Comedia mode (typical in some media servers and gateways). >>> - The callee is in a private LAN and the caller has public IP and >>> supports Comedia mode. >>> >>> >>> A RTP proxy is needed when: >>> >>> - Caller is in private LAN (with no STUN) and callee in public >>> internet (and not supporting Comedia). >>> - Caller and callee are in different private LAN's with no STUN. >> >> I would like to add that it's the device that can't receive audio that >> needs the RTP proxy to get incoming audio. >> >> If both devices are on private IP's, there's going to be two >> RTP proxys involved if they're on different SIP networks. >> >> Each SIP service needs an RTP proxy for supporting their >> local users. >> >> To simplify: >> >> - If my user is on a private IP and sends an INVITE, add RTP proxy >> handling to the INVITE >> >> - If my user receives a call and sends a 200 OK, add RTP proxy >> handling to the 200 OK >> > This logic is simple but not efficientTheoretically, if a call has > already a leg in public net, there is not need for a media relay for > traversing the nat. > > The only requirement is that all the devices to support symmetric media > (comedia support). > > So, after the caller proxy forced RTPproxy, the callee should not do the > same because the SDP already have a public IP, the nat traversal works > even if the callee is behind a nat. > > Regards, > Bogdan > > > > > ___ > 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] One way audio with AudioCodes Mediant 2000 and NAT
Hi all,
I eventually played around with the Audiocodes box and enabled some
settings so it worked with Comedia support.
Thanks,
Julian
On 2/10/09, Bogdan-Andrei Iancu wrote:
> HI Julian,
>
> If it has, you can actually force it by adding "direction=active" into
> SDP as indication. See "fix_nated_sdp("1") :
> http://www.opensips.org/html/docs/modules/1.4.x/nathelper.html#id270439
>
> Regards,
> Bogdan
>
> Julian Yap wrote:
>> Thanks all. I'll check to see if the AudioCodes gateway does have
>> comedia support.
>>
>> That clarifies some half baked NAT/RTP knowledge in my head.
>>
>> - Julian
>>
>>
>> On 2/10/09, Bogdan-Andrei Iancu wrote:
>>
>>> Hi Olle,
>>>
>>> Johansson Olle E wrote:
>>>
>>>> 10 feb 2009 kl. 12.25 skrev Iñaki Baz Castillo:
>>>>
>>>>
>>>>> 2009/2/10 :
>>>>>
>>>>>>> You don't know if RtpProxy should be running, does it mean you are
>>>>>>> trying to use it or not? I don't want to spend time inspecting what
>>>>>>> you want to do by reading your config, sorry.
>>>>>>>
>>>>>> Yeah, I'm trying not to run RTPProxy. After more testing, I'm
>>>>>> thinking I may
>>>>>> need to.
>>>>>>
>>>>> You cannot decide if you need RtpProxy or not based on testing, it's
>>>>> pure theory:
>>>>>
>>>>> A RTP proxy is NOT needed when (assuming the proxy has in the public
>>>>> internet):
>>>>>
>>>>> - Both caller and callee have public IP or use STUN.
>>>>> - Both caller and callee are in the *SAME* private LAN.
>>>>> - The caller is in a private LAN and the callee has public IP and
>>>>> supports Comedia mode (typical in some media servers and gateways).
>>>>> - The callee is in a private LAN and the caller has public IP and
>>>>> supports Comedia mode.
>>>>>
>>>>>
>>>>> A RTP proxy is needed when:
>>>>>
>>>>> - Caller is in private LAN (with no STUN) and callee in public
>>>>> internet (and not supporting Comedia).
>>>>> - Caller and callee are in different private LAN's with no STUN.
>>>>>
>>>> I would like to add that it's the device that can't receive audio that
>>>> needs the RTP proxy to get incoming audio.
>>>>
>>>> If both devices are on private IP's, there's going to be two
>>>> RTP proxys involved if they're on different SIP networks.
>>>>
>>>> Each SIP service needs an RTP proxy for supporting their
>>>> local users.
>>>>
>>>> To simplify:
>>>>
>>>> - If my user is on a private IP and sends an INVITE, add RTP proxy
>>>> handling to the INVITE
>>>>
>>>> - If my user receives a call and sends a 200 OK, add RTP proxy
>>>> handling to the 200 OK
>>>>
>>>>
>>> This logic is simple but not efficientTheoretically, if a call has
>>> already a leg in public net, there is not need for a media relay for
>>> traversing the nat.
>>>
>>> The only requirement is that all the devices to support symmetric media
>>> (comedia support).
>>>
>>> So, after the caller proxy forced RTPproxy, the callee should not do the
>>> same because the SDP already have a public IP, the nat traversal works
>>> even if the callee is behind a nat.
>>>
>>> Regards,
>>> Bogdan
>>>
>>>
>>>
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>
>
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Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT
The full story is that I was looking to get T.38 working behind NAT.
Unfortunately, no matter what I tried, it wouldn't work behind NAT. I
had the initial INVITE (G.711) working fine but when there was the
T.38 re-INVITE, the RTP media would connect up fine but just wouldn't
negotiate properly with T.38. Very strange as it worked fine with the
UA behind a public IP.
In the end, I implemented RTPProxy and T.38 works fine behind NAT.
- Julian
On Sun, Feb 15, 2009 at 1:25 AM, Bogdan-Andrei Iancu
wrote:
> Hi Julian,
>
> That is cool - in this way you save a lot of bandwidth and processing power
> with media relaying.
>
> Regards,
> Bogdan
>
> Julian Yap wrote:
>>
>> Hi all,
>>
>> I eventually played around with the Audiocodes box and enabled some
>> settings so it worked with Comedia support.
>>
>> Thanks,
>> Julian
>>
>>
>> On 2/10/09, Bogdan-Andrei Iancu wrote:
>>
>>>
>>> HI Julian,
>>>
>>> If it has, you can actually force it by adding "direction=active" into
>>> SDP as indication. See "fix_nated_sdp("1") :
>>>
>>> http://www.opensips.org/html/docs/modules/1.4.x/nathelper.html#id270439
>>>
>>> Regards,
>>> Bogdan
>>>
>>> Julian Yap wrote:
>>>
>>>>
>>>> Thanks all. I'll check to see if the AudioCodes gateway does have
>>>> comedia support.
>>>>
>>>> That clarifies some half baked NAT/RTP knowledge in my head.
>>>>
>>>> - Julian
>>>>
>>>>
>>>> On 2/10/09, Bogdan-Andrei Iancu wrote:
>>>>
>>>>
>>>>>
>>>>> Hi Olle,
>>>>>
>>>>> Johansson Olle E wrote:
>>>>>
>>>>>
>>>>>>
>>>>>> 10 feb 2009 kl. 12.25 skrev Iñaki Baz Castillo:
>>>>>>
>>>>>>
>>>>>>
>>>>>>>
>>>>>>> 2009/2/10 :
>>>>>>>
>>>>>>>
>>>>>>>>>
>>>>>>>>> You don't know if RtpProxy should be running, does it mean you are
>>>>>>>>> trying to use it or not? I don't want to spend time inspecting what
>>>>>>>>> you want to do by reading your config, sorry.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> Yeah, I'm trying not to run RTPProxy. After more testing, I'm
>>>>>>>> thinking I may
>>>>>>>> need to.
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> You cannot decide if you need RtpProxy or not based on testing, it's
>>>>>>> pure theory:
>>>>>>>
>>>>>>> A RTP proxy is NOT needed when (assuming the proxy has in the public
>>>>>>> internet):
>>>>>>>
>>>>>>> - Both caller and callee have public IP or use STUN.
>>>>>>> - Both caller and callee are in the *SAME* private LAN.
>>>>>>> - The caller is in a private LAN and the callee has public IP and
>>>>>>> supports Comedia mode (typical in some media servers and gateways).
>>>>>>> - The callee is in a private LAN and the caller has public IP and
>>>>>>> supports Comedia mode.
>>>>>>>
>>>>>>>
>>>>>>> A RTP proxy is needed when:
>>>>>>>
>>>>>>> - Caller is in private LAN (with no STUN) and callee in public
>>>>>>> internet (and not supporting Comedia).
>>>>>>> - Caller and callee are in different private LAN's with no STUN.
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> I would like to add that it's the device that can't receive audio that
>>>>>> needs the RTP proxy to get incoming audio.
>>>>>>
>>>>>> If both devices are on private IP's, there's going to be two
>>>>>> RTP proxys involved if they're on different SIP networks.
>>>>>>
>>>>>> Each SIP service needs an RTP proxy for supporting their
>>>>>> local users.
>>>>>>
>>>>>> To simplify:
>>>>>>
>>>>>> - If my user is on a private IP and sends an INVITE, add RTP proxy
>>>>>> handling to the INVITE
>>>>>>
>>>>>> - If my user receives a call and sends a 200 OK, add RTP proxy
>>>>>> handling to the 200 OK
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>> This logic is simple but not efficientTheoretically, if a call has
>>>>> already a leg in public net, there is not need for a media relay for
>>>>> traversing the nat.
>>>>>
>>>>> The only requirement is that all the devices to support symmetric media
>>>>> (comedia support).
>>>>>
>>>>> So, after the caller proxy forced RTPproxy, the callee should not do
>>>>> the
>>>>> same because the SDP already have a public IP, the nat traversal works
>>>>> even if the callee is behind a nat.
>>>>>
>>>>> Regards,
>>>>> Bogdan
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ___
>>>>> Users mailing list
>>>>> Users@lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>
>>>
>>
>>
>
>
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Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT
In an example scenario, the re-INVITE is handled by the end device.
So:
PSTN Fax --> GW --> OpenSIPS --> UA (ATA attached to Fax machine)
UA answers the call and then sends the re-INVITE which is correct as
that is the terminating side.
I read this RFC
http://tools.ietf.org/html/draft-mule-sip-t38callflows-02 which was
quite handy. :P
The re-INVITE get accepted and RTP communication starts... However,
for some reason, the T.38 part fails. In theory it should work but
doesn't for me. Perhaps it's something wrong with my config at the
time and the handling of the re-INVITE and NAT. Or perhaps it was
some obscure issue with the GW and T.38 communications and timing,
etc... Eventually I re-implemented it all with RTPProxy and that
worked for me first time, inbound and outbound.
Perhaps if someone has a clean working config with re-INVITE without
using RTPProxy or MediaProxy, I can try that. Seems like all the
example configs out there are used with a RTP proxy.
- Julian
On Mon, Feb 16, 2009 at 1:04 PM, Bogdan-Andrei Iancu
wrote:
> Hi Julian,
>
> You can still handle the NAT wih COMEDIA even for T.38, but you have to
> handle the re-INVITE also . In your scenario, who is generating the
> re-INVITE?
>
> Regards,
> Bogdan
>
> Julian Yap wrote:
>>
>> The full story is that I was looking to get T.38 working behind NAT.
>>
>> Unfortunately, no matter what I tried, it wouldn't work behind NAT. I
>> had the initial INVITE (G.711) working fine but when there was the
>> T.38 re-INVITE, the RTP media would connect up fine but just wouldn't
>> negotiate properly with T.38. Very strange as it worked fine with the
>> UA behind a public IP.
>>
>> In the end, I implemented RTPProxy and T.38 works fine behind NAT.
>>
>> - Julian
>>
>> On Sun, Feb 15, 2009 at 1:25 AM, Bogdan-Andrei Iancu
>> wrote:
>>
>>>
>>> Hi Julian,
>>>
>>> That is cool - in this way you save a lot of bandwidth and processing
>>> power
>>> with media relaying.
>>>
>>> Regards,
>>> Bogdan
>>>
>>> Julian Yap wrote:
>>>
>>>>
>>>> Hi all,
>>>>
>>>> I eventually played around with the Audiocodes box and enabled some
>>>> settings so it worked with Comedia support.
>>>>
>>>> Thanks,
>>>> Julian
>>>>
>>>>
>>>> On 2/10/09, Bogdan-Andrei Iancu wrote:
>>>>
>>>>
>>>>>
>>>>> HI Julian,
>>>>>
>>>>> If it has, you can actually force it by adding "direction=active" into
>>>>> SDP as indication. See "fix_nated_sdp("1") :
>>>>>
>>>>>
>>>>> http://www.opensips.org/html/docs/modules/1.4.x/nathelper.html#id270439
>>>>>
>>>>> Regards,
>>>>> Bogdan
>>>>>
>>>>> Julian Yap wrote:
>>>>>
>>>>>
>>>>>>
>>>>>> Thanks all. I'll check to see if the AudioCodes gateway does have
>>>>>> comedia support.
>>>>>>
>>>>>> That clarifies some half baked NAT/RTP knowledge in my head.
>>>>>>
>>>>>> - Julian
>>>>>>
>>>>>>
>>>>>> On 2/10/09, Bogdan-Andrei Iancu wrote:
>>>>>>
>>>>>>
>>>>>>
>>>>>>>
>>>>>>> Hi Olle,
>>>>>>>
>>>>>>> Johansson Olle E wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>> 10 feb 2009 kl. 12.25 skrev Iñaki Baz Castillo:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>>
>>>>>>>>> 2009/2/10 :
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> You don't know if RtpProxy should be running, does it mean you
>>>>>>>>>>> are
>>>>>>>>>>> trying to use it or not? I don't want to spend time inspecting
>>>>>>>>>>> what
>>>>>>>>>>> you want to do by reading your config, sorry.
>>>>>>>>>>>
>>>>>>>>>&
[OpenSIPS-Users] Loading rpid_avp from auth
Hi all,
I'm intending on loading rpid_avp from auth. This will be used with a
P-Asserted-Identity header. I am storing the RPID in the 'rpid' of
the subscriber column of my database.
I believe that s:rpid should then get loaded during authentication but
I think I'm missing something.
Here is my module load line:
modparam("^auth$|lcr", "rpid_avp", "$avp(i:302)")
Then later on, setting the $avp(i:302) manually works:
$avp(i:302) = "";
append_hf("P-Asserted-Identity: $avp(i:302)\r\n");
Any clues here? Basically I want to eliminate the line where I'm
hard-coding in the $avp(i:302) value. I'm kinda confused with the
documentation and haven't found any examples.
Thanks,
Julian
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Re: [OpenSIPS-Users] Loading rpid_avp from auth
Sergio and OpenSIPS users,
Damn, I feel so stupid. Changing that modparam worked for me.
For those wondering, here is how I load it up:
if ($avp(s:rpid)) {
append_hf("P-Asserted-Identity: \r\n");
};
One general question on AVP's. When I select AVP to have an integer
value 302 in the modparam statement, what is 302? Is that just an
arbitrary value? Or is value '302' preassigned somewhere? I'm
working from examples and AVP's are still kind of confusing.
- Julian
On Sun, Feb 22, 2009 at 8:08 AM, Sergio Gutierrez wrote:
>
> Hi Julian.
>
> Are you using the syntax you pasted for modparam?
>
> Try the following syntax:
>
> modparam("auth|lcr", "rpid_avp", "$avp(i:302)")
>
> Regards.
>
> Sergio G.
>
>
> On Sun, Feb 22, 2009 at 7:36 AM, Julian Yap wrote:
>>
>> Hi all,
>>
>> I'm intending on loading rpid_avp from auth. This will be used with a
>> P-Asserted-Identity header. I am storing the RPID in the 'rpid' of
>> the subscriber column of my database.
>>
>> I believe that s:rpid should then get loaded during authentication but
>> I think I'm missing something.
>>
>> Here is my module load line:
>> modparam("^auth$|lcr", "rpid_avp", "$avp(i:302)")
>>
>> Then later on, setting the $avp(i:302) manually works:
>> $avp(i:302) = "";
>> append_hf("P-Asserted-Identity: $avp(i:302)\r\n");
>>
>> Any clues here? Basically I want to eliminate the line where I'm
>> hard-coding in the $avp(i:302) value. I'm kinda confused with the
>> documentation and haven't found any examples.
>>
>> Thanks,
>> Julian
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> --
> Sergio Gutiérrez
>
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Re: [OpenSIPS-Users] OpenSIPs 1.4.4 rpm on CentOS
Hi Marc,
I took a look at fixing the spec file.
Try the attached patch file. Builds fine for me on CentOS 5.x 32-bit.
Developers, please test and accept the patch!
Thanks,
Julian
On Wed, Feb 25, 2009 at 6:04 AM, Marc Leurent
wrote:
> Thank you for your answer,
> I have seen that you wrote this spec and I thank you for this, but,
> in the code retrieved from the opensips website:
> http://www.opensips.org/pub/opensips/1.4.4/src/opensips-1.4.4-tls_src.tar.gz
> the packaging/rpm/asterisk.spec-4.1 defines a 1.4.0 version
>
> here is the problem I had and how I solve them:
> 1) the version parameter was 1.4.0, I have replaced it with 1.4.4
> 2) the download URL has changed from
> http://opensips.org/pub/opensips/stable/%{name}-%{ver}_src.tar.gz
> to
> http://www.opensips.org/pub/opensips/%{ver}/src/%{name}-%{ver}-tls_src.tar.gz
> 3) The name of the mysql module has change to db_mysql, so I had to change
> some lines into the spec file,...
> 4) the name of some build dependencies have changed on centOS, like
> unixODBC-devel, not unixodbc-dev,...
>
> Even without modifying the code to retrieve the 1.4.0 version, it wasn't
> working on centOS because of the download URL and the name of mysql module
>
> Unfortunatly, I cannot use Debian like I used to and its great packaging
> system, so I'm a little bit stuck...
>
> Thank you for your time.
> Have a nice afternoon!
>
>
> Le Wednesday 25 February 2009 15.05:22 Bogdan-Andrei Iancu, vous avez écrit :
>> Hi Marc,
>>
>> even if the name of the spec is misleading, it will build the 4.4 (the
>> code you downloaded)so, have you tried without changing the script
>> itself?
>>
>> Regards,
>> Bogdan
>>
>> Marc Leurent wrote:
>> > Hello everybody,
>> > I had to migrate from debian to centOS and I would like to build rpm
>> > packages of opensips on a CentOS dev server.
>> > I have retrieved the opensips.spec-4.1 file from the packaging folder in
>> > opensips source code and I'm trying to adapt it to opensips-4.4.
>> > I would be very grateful if someone that have already done that could
>> > help me...
>> >
>> > Best Regards,
>> >
>> > When compiling, I have some path error I'm trying to fix.
>> >
>> > rpmbuild -bb opensips.spec-4.4
>> >
>> >
>> >
>> > RPM build errors:
>> > File not found:
>> > /var/tmp/opensips-1.4.4-root/usr/sbin/opensips/opensipsctl File not
>> > found: /var/tmp/opensips-1.4.4-root/usr/share/doc/opensips/README.mysql
>> > File not
>> > found: /var/tmp/opensips-1.4.4-root/usr/lib/opensips/modules/mysql.so
>> > File not found:
>> > /var/tmp/opensips-1.4.4-root/usr/sbin/opensips_mysql.sh File not
>> > found: /var/tmp/opensips-1.4.4-root/usr/lib/opensipsctl/opensipsctl.mysql
>> > File not
>> > found:
>> > /var/tmp/opensips-1.4.4-root/usr/share/doc/opensips/README.postgres File
>> > not
>> > found: /var/tmp/opensips-1.4.4-root/usr/lib/opensips/modules/postgres.so
>> > File not found:
>> > /var/tmp/opensips-1.4.4-root/usr/sbin/opensips_postgres.sh File not
>> > found: /var/tmp/opensips-1.4.4-root/usr/lib/opensipsctl/opensipsctl.pgsql
>> > File not
>> > found:
>> > /var/tmp/opensips-1.4.4-root/usr/share/doc/opensips/README.unixodbc File
>> > not
>> > found: /var/tmp/opensips-1.4.4-root/usr/lib/opensips/modules/unixodbc.so
>> > File not
>> > found: /var/tmp/opensips-1.4.4-root/usr/share/doc/opensips/README.pa
>> > File not
>> > found: /var/tmp/opensips-1.4.4-root/usr/lib/opensips/modules/pa.so
>
>
>
> --
> -- --
> Marc LEURENT
> Ingénieur VoIP
>
> DECKPOINT SA
> Une société du groupe VTX Telecom
>
> Rue Eugène-Marziano 15 - 1227 Les Acacias
> http://www.vtx.ch - marc.leur...@vtx-telecom.ch
>
> VTX, votre partenaire telecom proche de vous !
>
>
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
opensips.spec-4.1.patch
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Re: [OpenSIPS-Users] OpenSIPs 1.4.4 rpm on CentOS
Sorry, I realized my commands were back to front when I created the patch.
The corrected on is attached.
- Julian
On Wed, Feb 25, 2009 at 11:33 AM, Julian Yap wrote:
> Hi Marc,
>
> I took a look at fixing the spec file.
>
> Try the attached patch file. Builds fine for me on CentOS 5.x 32-bit.
>
> Developers, please test and accept the patch!
>
> Thanks,
> Julian
>
> On Wed, Feb 25, 2009 at 6:04 AM, Marc Leurent
> wrote:
>> Thank you for your answer,
>> I have seen that you wrote this spec and I thank you for this, but,
>> in the code retrieved from the opensips website:
>> http://www.opensips.org/pub/opensips/1.4.4/src/opensips-1.4.4-tls_src.tar.gz
>> the packaging/rpm/asterisk.spec-4.1 defines a 1.4.0 version
>>
>> here is the problem I had and how I solve them:
>> 1) the version parameter was 1.4.0, I have replaced it with 1.4.4
>> 2) the download URL has changed from
>> http://opensips.org/pub/opensips/stable/%{name}-%{ver}_src.tar.gz
>> to
>> http://www.opensips.org/pub/opensips/%{ver}/src/%{name}-%{ver}-tls_src.tar.gz
>> 3) The name of the mysql module has change to db_mysql, so I had to change
>> some lines into the spec file,...
>> 4) the name of some build dependencies have changed on centOS, like
>> unixODBC-devel, not unixodbc-dev,...
>>
>> Even without modifying the code to retrieve the 1.4.0 version, it wasn't
>> working on centOS because of the download URL and the name of mysql module
>>
>> Unfortunatly, I cannot use Debian like I used to and its great packaging
>> system, so I'm a little bit stuck...
>>
>> Thank you for your time.
>> Have a nice afternoon!
>>
>>
>> Le Wednesday 25 February 2009 15.05:22 Bogdan-Andrei Iancu, vous avez écrit :
>>> Hi Marc,
>>>
>>> even if the name of the spec is misleading, it will build the 4.4 (the
>>> code you downloaded)so, have you tried without changing the script
>>> itself?
>>>
>>> Regards,
>>> Bogdan
>>>
>>> Marc Leurent wrote:
>>> > Hello everybody,
>>> > I had to migrate from debian to centOS and I would like to build rpm
>>> > packages of opensips on a CentOS dev server.
>>> > I have retrieved the opensips.spec-4.1 file from the packaging folder in
>>> > opensips source code and I'm trying to adapt it to opensips-4.4.
>>> > I would be very grateful if someone that have already done that could
>>> > help me...
>>> >
>>> > Best Regards,
>>> >
>>> > When compiling, I have some path error I'm trying to fix.
>>> >
>>> > rpmbuild -bb opensips.spec-4.4
>>> >
>>> >
>>> >
>>> > RPM build errors:
>>> > File not found:
>>> > /var/tmp/opensips-1.4.4-root/usr/sbin/opensips/opensipsctl File not
>>> > found: /var/tmp/opensips-1.4.4-root/usr/share/doc/opensips/README.mysql
>>> > File not
>>> > found: /var/tmp/opensips-1.4.4-root/usr/lib/opensips/modules/mysql.so
>>> > File not found:
>>> > /var/tmp/opensips-1.4.4-root/usr/sbin/opensips_mysql.sh File not
>>> > found: /var/tmp/opensips-1.4.4-root/usr/lib/opensipsctl/opensipsctl.mysql
>>> > File not
>>> > found:
>>> > /var/tmp/opensips-1.4.4-root/usr/share/doc/opensips/README.postgres File
>>> > not
>>> > found: /var/tmp/opensips-1.4.4-root/usr/lib/opensips/modules/postgres.so
>>> > File not found:
>>> > /var/tmp/opensips-1.4.4-root/usr/sbin/opensips_postgres.sh File not
>>> > found: /var/tmp/opensips-1.4.4-root/usr/lib/opensipsctl/opensipsctl.pgsql
>>> > File not
>>> > found:
>>> > /var/tmp/opensips-1.4.4-root/usr/share/doc/opensips/README.unixodbc File
>>> > not
>>> > found: /var/tmp/opensips-1.4.4-root/usr/lib/opensips/modules/unixodbc.so
>>> > File not
>>> > found: /var/tmp/opensips-1.4.4-root/usr/share/doc/opensips/README.pa
>>> > File not
>>> > found: /var/tmp/opensips-1.4.4-root/usr/lib/opensips/modules/pa.so
>>
>>
>>
>> --
>> -- --
>> Marc LEURENT
>> Ingénieur VoIP
>>
>> DECKPOINT SA
>> Une société du groupe VTX Telecom
>>
>> Rue Eugène-Marziano 15 - 1227 Les Acacias
>> http://www.vtx.ch - marc.leur...@vtx-telecom.ch
>>
>> VTX, votre partenaire telecom proche de vous !
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
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Re: [OpenSIPS-Users] OpenSIPs 1.4.4 rpm on CentOS
Marc,
Try this updated patch file.
- Julian
On Wed, Feb 25, 2009 at 11:41 PM, Marc Leurent
wrote:
> Hello,
> Thank you for the patch, I was closed to it :-)
> I have applied the patch and launched the following compilation command:
> rpmbuild -bb opensips.spec-4.4-julian
>> /root/opensips-compilation-spec-4.4-julian.log
> 2> /root/opensips-compilation-spec-4.4-julian-error.log
>
> and I have some strange error,
> I'm compiling it on
> CentOS 5.2
> Linux 2.6.18-92.1.22.el5PAE
> gcc version 4.1.2 20071124 (Red Hat 4.1.2-42)
>
> Please find enclosed the error log
> Any clue? I do not understand!
>
> Best Regards!
>
> Le Wednesday 25 February 2009 22.33:34 Julian Yap, vous avez écrit :
>> Hi Marc,
>>
>> I took a look at fixing the spec file.
>>
>> Try the attached patch file. Builds fine for me on CentOS 5.x 32-bit.
>>
>> Developers, please test and accept the patch!
>>
>> Thanks,
>> Julian
>>
>> On Wed, Feb 25, 2009 at 6:04 AM, Marc Leurent
>>
>> wrote:
>> > Thank you for your answer,
>> > I have seen that you wrote this spec and I thank you for this, but,
>> > in the code retrieved from the opensips website:
>> > http://www.opensips.org/pub/opensips/1.4.4/src/opensips-1.4.4-tls_src.tar
>> >.gz the packaging/rpm/asterisk.spec-4.1 defines a 1.4.0 version
>> >
>> > here is the problem I had and how I solve them:
>> > 1) the version parameter was 1.4.0, I have replaced it with 1.4.4
>> > 2) the download URL has changed from
>> > http://opensips.org/pub/opensips/stable/%{name}-%{ver}_src.tar.gz
>> > to
>> > http://www.opensips.org/pub/opensips/%{ver}/src/%{name}-%{ver}-tls_src.ta
>> >r.gz 3) The name of the mysql module has change to db_mysql, so I had to
>> > change some lines into the spec file,...
>> > 4) the name of some build dependencies have changed on centOS, like
>> > unixODBC-devel, not unixodbc-dev,...
>> >
>> > Even without modifying the code to retrieve the 1.4.0 version, it wasn't
>> > working on centOS because of the download URL and the name of mysql
>> > module
>> >
>> > Unfortunatly, I cannot use Debian like I used to and its great packaging
>> > system, so I'm a little bit stuck...
>> >
>> > Thank you for your time.
>> > Have a nice afternoon!
>> >
>> > Le Wednesday 25 February 2009 15.05:22 Bogdan-Andrei Iancu, vous avez
> écrit :
>> >> Hi Marc,
>> >>
>> >> even if the name of the spec is misleading, it will build the 4.4 (the
>> >> code you downloaded)so, have you tried without changing the script
>> >> itself?
>> >>
>> >> Regards,
>> >> Bogdan
>> >>
>> >> Marc Leurent wrote:
>> >> > Hello everybody,
>> >> > I had to migrate from debian to centOS and I would like to build rpm
>> >> > packages of opensips on a CentOS dev server.
>> >> > I have retrieved the opensips.spec-4.1 file from the packaging folder
>> >> > in opensips source code and I'm trying to adapt it to opensips-4.4. I
>> >> > would be very grateful if someone that have already done that could
>> >> > help me...
>> >> >
>> >> > Best Regards,
>> >> >
>> >> > When compiling, I have some path error I'm trying to fix.
>> >> >
>> >> > rpmbuild -bb opensips.spec-4.4
>> >> >
>> >> >
>> >> >
>> >> > RPM build errors:
>> >> > File not found:
>> >> > /var/tmp/opensips-1.4.4-root/usr/sbin/opensips/opensipsctl File not
>> >> > found:
>> >> > /var/tmp/opensips-1.4.4-root/usr/share/doc/opensips/README.mysql File
>> >> > not
>> >> > found: /var/tmp/opensips-1.4.4-root/usr/lib/opensips/modules/mysql.so
>> >> > File not found:
>> >> > /var/tmp/opensips-1.4.4-root/usr/sbin/opensips_mysql.sh File not
>> >> > found:
>> >> > /var/tmp/opensips-1.4.4-root/usr/lib/opensipsctl/opensipsctl.mysql
>> >> > File not
>> >> > found:
>> >> > /var/tmp/opensips-1.4.4-root/usr/share/doc/opensips/README.postgres
>> >> > File not
>> >> > found:
>> >> > /var/tmp/opensips-1.4.4-root/us
Re: [OpenSIPS-Users] OpenSIPs 1.4.4 rpm on CentOS - patch OK - should be uploaded in SVN repo!
Hi Marc,
That's great news. Just wanted to give something back to the
community :P I know a lot of people use CentOS such as myself.
Committers, please test and add with Changelog comment:
Thu Feb 26 2009 Julian Yap
* Update spec to OpenSIPS version 1.4.4. Rename database module
names. Rename PUA module. Fix dependencies. Minor build fixes.
Additional files and new modules.
Thanks,
Julian
On Thu, Feb 26, 2009 at 1:19 AM, Marc Leurent
wrote:
> It's working! I have done exactly the same thing but without sorting sections
> in alphabetical order...
> For me this patch is great and should be added in svn repository!
> No error, no warning!!
>
> Julian, You should add a comment in the changelog!
>
>
> Checking for unpackaged
> file(s): /usr/lib/rpm/check-files /var/tmp/opensips-1.4.4-buildroot
> Wrote: /usr/src/redhat/RPMS/i386/opensips-1.4.4-0.i386.rpm
> Wrote: /usr/src/redhat/RPMS/i386/opensips-mysql-1.4.4-0.i386.rpm
> Wrote: /usr/src/redhat/RPMS/i386/opensips-postgres-1.4.4-0.i386.rpm
> Wrote: /usr/src/redhat/RPMS/i386/opensips-unixodbc-1.4.4-0.i386.rpm
> Wrote: /usr/src/redhat/RPMS/i386/opensips-jabber-1.4.4-0.i386.rpm
> Wrote: /usr/src/redhat/RPMS/i386/opensips-cpl-1.4.4-0.i386.rpm
> Wrote: /usr/src/redhat/RPMS/i386/opensips-pua-1.4.4-0.i386.rpm
> Wrote: /usr/src/redhat/RPMS/i386/opensips-radius-1.4.4-0.i386.rpm
> Executing(%clean): /bin/sh -e /var/tmp/rpm-tmp.33568
>
> Have a nice day
>
> Le Thursday 26 February 2009 11.47:07 Julian Yap, vous avez écrit :
>> Marc,
>>
>> Try this updated patch file.
>>
>> - Julian
>>
>> On Wed, Feb 25, 2009 at 11:41 PM, Marc Leurent
>>
>> wrote:
>> > Hello,
>> > Thank you for the patch, I was closed to it :-)
>> > I have applied the patch and launched the following compilation command:
>> > rpmbuild -bb opensips.spec-4.4-julian
>> >
>> >> /root/opensips-compilation-spec-4.4-julian.log
>> >
>> > 2> /root/opensips-compilation-spec-4.4-julian-error.log
>> >
>> > and I have some strange error,
>> > I'm compiling it on
>> > CentOS 5.2
>> > Linux 2.6.18-92.1.22.el5PAE
>> > gcc version 4.1.2 20071124 (Red Hat 4.1.2-42)
>> >
>> > Please find enclosed the error log
>> > Any clue? I do not understand!
>> >
>> > Best Regards!
>> >
>> > Le Wednesday 25 February 2009 22.33:34 Julian Yap, vous avez écrit :
>> >> Hi Marc,
>> >>
>> >> I took a look at fixing the spec file.
>> >>
>> >> Try the attached patch file. Builds fine for me on CentOS 5.x 32-bit.
>> >>
>> >> Developers, please test and accept the patch!
>> >>
>> >> Thanks,
>> >> Julian
>> >>
>> >> On Wed, Feb 25, 2009 at 6:04 AM, Marc Leurent
>> >>
>> >> wrote:
>> >> > Thank you for your answer,
>> >> > I have seen that you wrote this spec and I thank you for this, but,
>> >> > in the code retrieved from the opensips website:
>> >> > http://www.opensips.org/pub/opensips/1.4.4/src/opensips-1.4.4-tls_src.
>> >> >tar .gz the packaging/rpm/asterisk.spec-4.1 defines a 1.4.0 version
>> >> >
>> >> > here is the problem I had and how I solve them:
>> >> > 1) the version parameter was 1.4.0, I have replaced it with 1.4.4
>> >> > 2) the download URL has changed from
>> >> > http://opensips.org/pub/opensips/stable/%{name}-%{ver}_src.tar.gz
>> >> > to
>> >> > http://www.opensips.org/pub/opensips/%{ver}/src/%{name}-%{ver}-tls_src
>> >> >.ta r.gz 3) The name of the mysql module has change to db_mysql, so I
>> >> > had to change some lines into the spec file,...
>> >> > 4) the name of some build dependencies have changed on centOS, like
>> >> > unixODBC-devel, not unixodbc-dev,...
>> >> >
>> >> > Even without modifying the code to retrieve the 1.4.0 version, it
>> >> > wasn't working on centOS because of the download URL and the name of
>> >> > mysql module
>> >> >
>> >> > Unfortunatly, I cannot use Debian like I used to and its great
>> >> > packaging system, so I'm a little bit stuck...
>> >> >
>> >> > Thank you for your time.
>> >> > Have a nice afternoon!
>> >> >
>> >> > Le Wednesday 25 February 2009 15.05:22 Bogdan-Andrei Iancu, vous avez
>> >
>> > écrit :
>> >> >> Hi Marc,
&g
Re: [OpenSIPS-Users] T38 with rtp proxy
Michel, Your trace looks kinda strange. Why does it initially send 4 INVITE's? I think you need to check on that first. Because later on, it sends multiple T.38 INVITE's as well. Perhaps try and disable you firewall as well when testing. - Julian On Fri, Mar 6, 2009 at 10:23 AM, michel freiha wrote: > Dear Sir, > > Kindly note that Outgoing FAX is working fine...The issue appears in > incominf FAX from PSTN GW to OpenSIPS > > Regards > > On Fri, Mar 6, 2009 at 10:20 PM, michel freiha wrote: >> >> Hi all, >> I have a problem when trying to send t38 packets to OpenSIPSThe >> Endpoint is sending T38 packets and no response from OpenSIPS server as you >> can see in the attached SIP trace >> >> Regards > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Rewriting From header using Contact information
I have a scenario where the PSTN to SIP gateway (AudioCodes) I am
using sets the From header to 'anonymous' when it does not receive a
Calling Name from the PSTN side.
The modified INVITE from the gateway then looks like this (changed
some numbers and IP's):
From: "anonymous" ;tag=1c49690767.
To: .
CSeq: 1 INVITE.
Contact: .
When the gateway does receive the Calling Name from the PSTN, it looks
like this:
From: "HONOLULU HI" ;tag=1c1248847826.
To: .
CSeq: 1 INVITE.
Contact: .
In the first instance, I want to re-write the From header because I do
in fact have the calling number from the Contact header. In theory,
the PSTN gateway should sort this out for me and not send me the
'anonymous' From header but I've searched the manuals and it doesn't.
Anyone else encountered this?
This is the IF statement that satisfies the criteria:
if($fu=='sip:anonym...@anonymous.invalid' &&
$ct=~"^ That is Contact header variable;From tag
I had a look at the UAC module but using the function
uac_replace_from(), I don't know how to strip the '<' and '>' of the
Contact header to satisfy the arguments of the function. Is it
possible? I'm thinking that using the UAC is preferable to what I'm
proposing below.
Does this method sound sane? Or is it dangerous?:
remove_hf("From");
append_hf("From: $ct;$ft\r\n");
Thanks,
Julian
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Re: [OpenSIPS-Users] Rewriting From header using Contact information
I just tested and this does not work:
remove_hf("From");
append_hf("From: $ct;$ft\r\n");
So basically I want to rewrite the From header by using the details
from the Contact header.
Any suggestions?
- Julian
On Wed, Mar 18, 2009 at 1:15 AM, Julian Yap wrote:
> I have a scenario where the PSTN to SIP gateway (AudioCodes) I am
> using sets the From header to 'anonymous' when it does not receive a
> Calling Name from the PSTN side.
>
> The modified INVITE from the gateway then looks like this (changed
> some numbers and IP's):
> From: "anonymous" ;tag=1c49690767.
> To: .
> CSeq: 1 INVITE.
> Contact: .
>
> When the gateway does receive the Calling Name from the PSTN, it looks
> like this:
> From: "HONOLULU HI" ;tag=1c1248847826.
> To: .
> CSeq: 1 INVITE.
> Contact: .
>
> In the first instance, I want to re-write the From header because I do
> in fact have the calling number from the Contact header. In theory,
> the PSTN gateway should sort this out for me and not send me the
> 'anonymous' From header but I've searched the manuals and it doesn't.
> Anyone else encountered this?
>
> This is the IF statement that satisfies the criteria:
> if($fu=='sip:anonym...@anonymous.invalid' &&
> $ct=~"^ {
> xlog("L_INFO", "fix anonymous\n");
> }
>
> The final From would be:
> From: $ct;$ft
> --> That is Contact header variable;From tag
>
> I had a look at the UAC module but using the function
> uac_replace_from(), I don't know how to strip the '<' and '>' of the
> Contact header to satisfy the arguments of the function. Is it
> possible? I'm thinking that using the UAC is preferable to what I'm
> proposing below.
>
> Does this method sound sane? Or is it dangerous?:
> remove_hf("From");
> append_hf("From: $ct;$ft\r\n");
>
> Thanks,
> Julian
>
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Re: [OpenSIPS-Users] Rewriting From header using Contact information
Yeah, I've tried modifying the RPID header...
It looks like that's working for me. I had to make up a hard coded
display name...
This is slightly off topic but in my current T1's I need to add in an
Asterisk 'wait' command for the facility IE with the Calling name to
be sent through.
ie. --> http://www.voipinfo.org/wiki/view/CallerID
Is there something similar for OpenSIPS?
- Julian
On Wed, Mar 18, 2009 at 9:49 AM, Brett Nemeroff wrote:
> It's not really standards compliant, but you can do it. I'm not sure why you
> want to take stuff out of the contact header and stick it in from. From
> shouldn't ever be changed, if you can help it. If your really trying to
> change the resultant display name (ie: caller id) then go for manipulating /
> adding RPID headers instead.
> http://www.opensips.org/html/docs/modules/1.4.x/auth.html#append-rpid-hf-no-params
>
> I've had to rewrite "From" headers because non-compliant carriers have
> insisted on using data in the from header for E911 call routing.. idiots..
> :P
> If you insist on changing the From header; this may help.. you won't be able
> to do it the way you are presently trying..
> take a look here:
> http://www.opensips.org/index.php?n=Resources.DocsTipsFaqs
> and here:
> http://www.opensips.org/html/docs/modules/1.4.x/uac.html#id227417
>
> On Wed, Mar 18, 2009 at 2:37 PM, Julian Yap wrote:
>>
>> I just tested and this does not work:
>> remove_hf("From");
>> append_hf("From: $ct;$ft\r\n");
>>
>> So basically I want to rewrite the From header by using the details
>> from the Contact header.
>>
>> Any suggestions?
>>
>> - Julian
>>
>> On Wed, Mar 18, 2009 at 1:15 AM, Julian Yap wrote:
>> > I have a scenario where the PSTN to SIP gateway (AudioCodes) I am
>> > using sets the From header to 'anonymous' when it does not receive a
>> > Calling Name from the PSTN side.
>> >
>> > The modified INVITE from the gateway then looks like this (changed
>> > some numbers and IP's):
>> > From: "anonymous" ;tag=1c49690767.
>> > To: .
>> > CSeq: 1 INVITE.
>> > Contact: .
>> >
>> > When the gateway does receive the Calling Name from the PSTN, it looks
>> > like this:
>> > From: "HONOLULU HI"
>> > ;tag=1c1248847826.
>> > To: .
>> > CSeq: 1 INVITE.
>> > Contact: .
>> >
>> > In the first instance, I want to re-write the From header because I do
>> > in fact have the calling number from the Contact header. In theory,
>> > the PSTN gateway should sort this out for me and not send me the
>> > 'anonymous' From header but I've searched the manuals and it doesn't.
>> > Anyone else encountered this?
>> >
>> > This is the IF statement that satisfies the criteria:
>> > if($fu=='sip:anonym...@anonymous.invalid' &&
>> > $ct=~"^> > {
>> > xlog("L_INFO", "fix anonymous\n");
>> > }
>> >
>> > The final From would be:
>> > From: $ct;$ft
>> > --> That is Contact header variable;From tag
>> >
>> > I had a look at the UAC module but using the function
>> > uac_replace_from(), I don't know how to strip the '<' and '>' of the
>> > Contact header to satisfy the arguments of the function. Is it
>> > possible? I'm thinking that using the UAC is preferable to what I'm
>> > proposing below.
>> >
>> > Does this method sound sane? Or is it dangerous?:
>> > remove_hf("From");
>> > append_hf("From: $ct;$ft\r\n");
>> >
>> > Thanks,
>> > Julian
>> >
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] OpenSIPS with T38
How are you handling your T.38 media? 2009/3/28 John kader : > Dear All, > > I'm new to opensips...I have installed it on my server and everything looks > fine except that I'm not be be able to send FAX using T38...It seems that > Ihave some problems in ACK and REINVITE... > > Can you please provide me a working configuration example where FAX is going > fine in order to test it on my servers? > > Regards > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Invalid_version_of_table_rls_presentity
Did you upgrade from 1.4 and hence did you run the upgrade scripts? - Julian On 4/4/09, JayaPrakash wrote: > Hi All, > Opensips 1.5 is installed in Debian 5. > Presence related module are loaded in the following order. > > loadmodule "presence.so" > loadmodule "presence_xml.so" > loadmodule "presence_mwi.so" > loadmodule "pua.so" > loadmodule "pua_mi.so" > loadmodule "presence_xcapdiff.so" > loadmodule "rls.so" > loadmodule "xcap_client.so" > loadmodule "mi_xmlrpc.so" > loadmodule "avp_radius.so" > > When Opensips is started, it is crashing throwing the following error about > invalid rls_presentity table. > > ERROR:core:db_check_table_version: invalid version 0 for table > rls_presentity found, expected 1 > Apr 4 20:53:51 [27955] ERROR:rls:mod_init: error during table version > check. > Apr 4 20:53:51 [27955] ERROR:core:init_mod: failed to initialize module rls > Apr 4 20:53:51 [27955] ERROR:core:main: error while initializing modules > Apr 4 20:53:51 [27955] NOTICE:presence:destroy: destroy module ... > > > Will you please review this error. > > Thanks > JayaPrakash > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Modifying INVITE header to add phone-context
I have a PSTN gateway which requires a Phone-Context value in the outgoing SIP INVITE message to further apply ISDN NPI/TON details. Here's an example of what I currently have going out to the PSTN gateway: INVITE sip:1...@sip.server.com:5060;user=phone SIP/2.0. This is what I require: INVITE sip:1...@sip.server.com:5060;phone-context=sip.server.com;user=phone SIP/2.0. Any clues on how to add the Phone-Context value? Thanks, Julian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Modifying INVITE header to add phone-context
I pretty much solved the issue.
This is what I used:
subst_uri('/^sip:([0-9]+)@(.*)$/sip:\...@\2;phone-context=sip.server.com/i
On Tue, Apr 7, 2009 at 4:25 PM, Julian Yap wrote:
> I have a PSTN gateway which requires a Phone-Context value in the
> outgoing SIP INVITE message to further apply ISDN NPI/TON details.
>
> Here's an example of what I currently have going out to the PSTN gateway:
> INVITE sip:1...@sip.server.com:5060;user=phone SIP/2.0.
>
> This is what I require:
> INVITE sip:1...@sip.server.com:5060;phone-context=sip.server.com;user=phone
> SIP/2.0.
>
> Any clues on how to add the Phone-Context value?
>
> Thanks,
> Julian
>
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Re: [OpenSIPS-Users] DRouting Sample Config
Echoing the sentiments of the original poster, I'm starting to look into Drouting as well... Having wasted my time with and implementing my current system with LCR :). Having to patch lcr_mod.c to increase MAX_NO_OF_LCRS is very very sad. But yeah, I was wondering if anyone had a LCR --> Drouting tutorial. OpenSIPs is going the way of Drouting (vs. Carrierroute) so that would be very handy. Also, I'm interested in the inbound aspects of Drouting as well... Right now I'm using the alias_db module to route inbound calls but that's limiting. Checking out an example would be handy. Thanks, Julian On Tue, Apr 28, 2009 at 8:04 AM, Brett Nemeroff wrote: > Hi Tim, > Actually, the documentation is quite good IMO. LCR and CarrierRoute are all > but replaced by Drouting.. so don't bother reading up on LCR or > CarrierRoute. > For drouting, setup your termination points in dr_gateways and your routing > rules in dr_rules. It's really pretty simple. > Are you having a specific problem we can help you out with? > -Brett > > On Tue, Apr 28, 2009 at 1:00 PM, Tim Donahue > wrote: >> >> Hi all, >> >> I have been using OpenSER for a couple years now for routing our DIDs >> from our carriers to the proper customer trunk on our SBC. We have been >> using a static configuration with a routing entry for each number DID >> number we have. As you can imagine, even with scripts building the >> openser.cfg, this has become a little cumbersome. >> >> We are looking at replacing the OpenSER servers with something that is >> database driven, and the drouting module looks like it is probably the >> best fit between the LCR, carrierroute and drouting modules. >> Unfortunately the documentation is quite thin and I have not been able >> to get a working config at this point. >> >> Does anyone have a config they would be willing to share with a couple >> example DB entries as an example for how to configure the drouting module? >> >> Tim >> >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Add Contact header to ACK
Hi, I have an issue where a T38 fax UA which is calling outbound does not add a contact header to an ACK when replying to the receiving party before starting the initial RTP conversation. As a result, the Re-INVITE from the receiving UA is misdirected and never gets sent to the calling UA. I'm wondering how I can add the Contact header to the ACK so that the Re-INVITE then sends to the right place. I tried writing the initial Contact header to an AVP variable but the AVP does not persist by the time I get to the ACK. Below are the relevant SIP packets (changed details for privacy). === INITIAL INVITE from calling UA has a Contact header === INVITE sip:5551234@mysipserver SIP/2.0 Via: SIP/2.0/UDP 8.8.8.10:5891;branch=z9hG4bK17BE From: IPFax ;tag=IPF_PORT_0001_17BD To: Call-ID: 9fb5d1a5-fbf7-43ee-8eed-d96a2853d91c@10.10.0.178 CSeq: 1 INVITE Max-Forwards: 70 Contact: User-Agent: Net Fax/IP_FAX-9.0.6039.885 Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY Content-Type: application/sdp Content-Length: 168 v=0 o=IPFax 0 0 IN IP4 8.8.8.10 s=SIP Fax Call i=IPFax c=IN IP4 8.8.8.10 t=0 0 m=audio 10946 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv === ACK from calling UA does NOT have a Contact header === ACK sip:8085551000@8.8.8.20:5060 SIP/2.0 Via: SIP/2.0/UDP 8.8.8.10:5891;branch=z9hG4bK17C0 Route: From: IPFax ;tag=IPF_PORT_0001_17BD To: ;tag=IPF_PORT_0024_DE2C Call-ID: 9fb5d1a5-fbf7-43ee-8eed-d96a2853d91c@10.10.0.178 CSeq: 2 ACK Max-Forwards: 70 User-Agent: Net Fax/IP_FAX-9.0.6039.885 Proxy-Authorization: Digest username="1006",realm="mysipserver",nonce="4f17fa65012ed6c5fca98a3b6522fa041bfb558d4833",opaque="",uri="sip:5551234@mysipserver",response="b9910d8e4b5994109ec671728365b2f6" Content-Length: 0 === REINVITE from receiving UA is misdirected === INVITE sip:1006@mysipserver SIP/2.0 Via: SIP/2.0/UDP 8.8.8.20:5060;branch=z9hG4bKDE2D Route: From: ;tag=IPF_PORT_0024_DE2C To: IPFax ;tag=IPF_PORT_0001_17BD Call-ID: 9fb5d1a5-fbf7-43ee-8eed-d96a2853d91c@10.10.0.178 CSeq: 3 INVITE Max-Forwards: 70 Contact: User-Agent: Net Fax/IP_FAX-8.6.5262.504 Supported: timer,replaces,billing,presence,* Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY Content-Type: application/sdp Content-Length: 359 v=0 o=IPFax 0 1 IN IP4 8.8.8.20 s=SIP Fax Call i=IPFax c=IN IP4 8.8.8.20 t=0 0 m=image 49198 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxUdpEC:t38UDPRedundancy ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Using RTPProxy to proxy all non-NAT'ed calls as well
I have a working set up where OpenSIPS is my registrar and SIP Proxy. NAT'ed UA calls are handled with RTPProxy. I want RTPProxy to handle non-NAT'ed UA's (for example, UA's on a public internet IP). I tried a quick test of running the force_rtp_proxy() function on non-NAT'ed INVITE packets but I get this error: Oct 18 00:18:17 sip1 /sbin/opensips[18749]: ERROR:nathelper:extract_body: message body has length zero Oct 18 00:18:17 sip1 /sbin/opensips[18749]: ERROR:nathelper:force_rtp_proxy2_f: can't extract body from the message Is it possible to run RTPProxy in this fashion? I'm looking for the easiest fix. Thanks, Julian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Using RTPProxy to proxy all non-NAT'ed calls as well
Hmm, so in theory RTPProxy should work in my case?
I just need to check the SDP? My NAT section has something like this:
if(!isflagset(22) && !search("^Content-Length:[ ]*0"))
That Content-Length check should suffice?
On Sun, Oct 18, 2009 at 12:34 AM, Alex Balashov
wrote:
> You are attempting to invoke rtpproxy on a request or reply which does
> not contain an SDP body, and therefore force_rtp_proxy() cannot
> operate on it.
>
> Julian Yap wrote:
>
>> I have a working set up where OpenSIPS is my registrar and SIP Proxy.
>>
>> NAT'ed UA calls are handled with RTPProxy.
>>
>> I want RTPProxy to handle non-NAT'ed UA's (for example, UA's on a
>> public internet IP).
>>
>> I tried a quick test of running the force_rtp_proxy() function on
>> non-NAT'ed INVITE packets but I get this error:
>> Oct 18 00:18:17 sip1 /sbin/opensips[18749]:
>> ERROR:nathelper:extract_body: message body has length zero
>> Oct 18 00:18:17 sip1 /sbin/opensips[18749]:
>> ERROR:nathelper:force_rtp_proxy2_f: can't extract body from the
>> message
>>
>> Is it possible to run RTPProxy in this fashion? I'm looking for the
>> easiest fix.
>>
>> Thanks,
>> Julian
>>
>> ___
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>
>
> --
> Alex Balashov - Principal
> Evariste Systems
> Web : http://www.evaristesys.com/
> Tel : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
>
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Re: [OpenSIPS-Users] Using RTPProxy to proxy all non-NAT'ed calls as well
Alex,
Thanks for your help. Works great now.
To answer my own question, it's possible to RTPProxy all calls.
Julian
On 10/18/09, Alex Balashov wrote:
> Julian Yap wrote:
>
>> Hmm, so in theory RTPProxy should work in my case?
>>
>> I just need to check the SDP? My NAT section has something like this:
>> if(!isflagset(22) && !search("^Content-Length:[ ]*0"))
>>
>> That Content-Length check should suffice?
>
> The payload may be something other than an SDP body.
>
> I suggest:
>
>if(search("Content-Type: application/sdp"))
>
> --
> Alex Balashov - Principal
> Evariste Systems
> Web : http://www.evaristesys.com/
> Tel : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
>
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[OpenSIPS-Users] PUA MI and pua_publish
I'm trying to hack together my own presence server. I have PUA USRLOC working fine. So then I just want to be able to generate the 'phone in use' and 'phone not in use' PUBLISH messages via PUA MI. Here's what I have in my pua_publish FIFO example: --- :pua_publish:test_fifo sip:10001...@sipdev1.me.com 3600 presence application/pidf+xml . . inuse --- The actual NOTIFY goes out as: inuseopen ... So the status gets overwritten as 'open'. Any tips? What would a working 'phone is in use' and 'phone is not in use' FIFO look like? Thanks! Julian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Different name for 'location' table?
If there's no module parameter, you'd probably need to hack the source code.
On Sat, Apr 24, 2010 at 12:27 AM, Chris Maciejewski wrote:
> Hi,
>
> Is it possible to use different name of 'location' table? For example
> if I wanted to use 'prefix1_location' instead of standard 'location',
> I understand in my config file I would use:
>
> save("prefix1_location");
> lookup("prefix1_location");
>
> but how do I tell 'userloc' module to use 'prefix1_location' instead
> of 'location' (in order to purge expired contacts etc.) ? There
> doesn't seem to be 'table_name(...)' module parameter?
>
> Regards,
> Chris
>
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Re: [OpenSIPS-Users] PUA MI and pua_publish
Hmm, I have this working now using SIP SIMPLE standard (by testing with X-Lite user agent)... Unfortunately this doesn't look to be supported by Polycom phones. Is there a way to translate the PUBLISH so the NOTIFY will go out in an appropriate format depending on the UA? eg. PUBLISH is generated server side using MI: :pua_publish:test_fifo sip:10001...@sipdev1.test.com 3600 presence application/pidf+xml . . openOn the Phone SIP packet: U 2010/04/26 00:49:32.590822 64.xxx.yyy.41:5060 -> 64.xxx.yyy.41:5060 PUBLISH sip:10001...@sipdev1.test.com SIP/2.0. Via: SIP/2.0/UDP 64.xxx.yyy.41;branch=z9hG4bK4d6c.8543c9f1.0. To: sip:10001...@sipdev1.test.com. From: ;tag=cdedd5270a3714d043d2a63f64a8b9e2-bc47. CSeq: 10 PUBLISH. Call-ID: 39497fd430730179-10...@64.xxx.yyy.41. Content-Length: 476. UA: Test SIP Proxy. Max-Forwards: 70. Event: presence. Expires: 3601. SIP-If-Match: a.1272199485.30908.19.4. Content-Type: application/pidf+xml. . openOn the Phone And the NOTIFY goes out: U 2010/04/26 00:49:32.594132 64.xxx.yyy.41:5060 -> 66.aaa.bbb.23:50330 NOTIFY sip:10001...@66.aaa.bbb.23:50330 SIP/2.0. Via: SIP/2.0/UDP 64.xxx.yyy.41;branch=z9hG4bKe0ef.b2a3e8c1.0. To: ;tag=49DA9FFC-42875603. From: ;tag=164906b045e4966ce5bf3f0df6f222f4-ba51. CSeq: 4 NOTIFY. Call-ID: 7f817a77-30df0af6-2e84e...@192.168.1.47. Content-Length: 583. UA: Test SIP Proxy. Max-Forwards: 70. Event: presence. Contact: . Subscription-State: active;expires=3136. Content-Type: application/pidf+xml. . openOn the Phoneopen On Sun, Apr 25, 2010 at 11:14 PM, Anca Vamanu wrote: > Hi Julian, > > Putting inuse as basic status is not really legal - since the pidf RFC > says that you can have only open and close > (http://www.faqs.org/rfcs/rfc3863.html). > > Regards, > > -- > Anca Vamanu > www.voice-system.ro > > > Julian Yap wrote: >> I'm trying to hack together my own presence server. I have PUA USRLOC >> working fine. >> >> So then I just want to be able to generate the 'phone in use' and >> 'phone not in use' PUBLISH messages via PUA MI. >> >> Here's what I have in my pua_publish FIFO example: >> --- >> :pua_publish:test_fifo >> sip:10001...@sipdev1.me.com >> 3600 >> presence >> application/pidf+xml >> . >> . >> > xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' >> xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' >> xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' >> entity='10001...@sipdev1.me.com'>inuse >> >> --- >> >> The actual NOTIFY goes out as: >> >> > xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" >> xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" >> xmlns:c="urn:ietf:params:xml:ns:pidf:cipid" >> entity="10001...@sipdev1.me.com">> id="0x7fff9d908330">inuse> xmlns="urn:ietf:params:xml:ns:pidf" >> id="0x7fff1b75a760">open >> >> >> >> ... So the status gets overwritten as 'open'. >> >> Any tips? What would a working 'phone is in use' and 'phone is not in >> use' FIFO look like? >> >> Thanks! >> >> Julian >> >> ___ >> 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] PUA MI and pua_publish
I guess Polycom phones only accept application/xpidf+xml. On Mon, Apr 26, 2010 at 1:18 AM, Anca Vamanu wrote: > Hi Julian, > > Julian Yap wrote: >> >> Hmm, I have this working now using SIP SIMPLE standard (by testing >> with X-Lite user agent)... Unfortunately this doesn't look to be >> supported by Polycom phones. Is there a way to translate the PUBLISH >> so the NOTIFY will go out in an appropriate format depending on the >> UA? >> >> > > I don't understand what you mean by translate the Publish in an appropriate > format.. What other format do you want to have? > > Regards, > > -- > Anca Vamanu > www.voice-system.ro > > >> eg. PUBLISH is generated server side using MI: >> :pua_publish:test_fifo >> sip:10001...@sipdev1.test.com >> 3600 >> presence >> application/pidf+xml >> . >> . >> > xmlns='urn:ietf:params:xml:ns:pidf' >> xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' >> xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' >> xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' >> >> entity='10001...@sipdev1.test.com'>openOn >> the Phone >> >> >> SIP packet: >> U 2010/04/26 00:49:32.590822 64.xxx.yyy.41:5060 -> 64.xxx.yyy.41:5060 >> PUBLISH sip:10001...@sipdev1.test.com SIP/2.0. >> Via: SIP/2.0/UDP 64.xxx.yyy.41;branch=z9hG4bK4d6c.8543c9f1.0. >> To: sip:10001...@sipdev1.test.com. >> From: >> ;tag=cdedd5270a3714d043d2a63f64a8b9e2-bc47. >> CSeq: 10 PUBLISH. >> Call-ID: 39497fd430730179-10...@64.xxx.yyy.41. >> Content-Length: 476. >> UA: Test SIP Proxy. >> Max-Forwards: 70. >> Event: presence. >> Expires: 3601. >> SIP-If-Match: a.1272199485.30908.19.4. >> Content-Type: application/pidf+xml. >> . >> >> > xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" >> xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" >> xmlns:c="urn:ietf:params:xml:ns:pidf:cipid" >> entity="10001...@sipdev1.test.com">> id="0x7fff82d20c50">open> >> id="0x7fff82d20c50">On >> the Phone >> >> >> And the NOTIFY goes out: >> U 2010/04/26 00:49:32.594132 64.xxx.yyy.41:5060 -> 66.aaa.bbb.23:50330 >> NOTIFY sip:10001...@66.aaa.bbb.23:50330 SIP/2.0. >> Via: SIP/2.0/UDP 64.xxx.yyy.41;branch=z9hG4bKe0ef.b2a3e8c1.0. >> To: ;tag=49DA9FFC-42875603. >> From: >> ;tag=164906b045e4966ce5bf3f0df6f222f4-ba51. >> CSeq: 4 NOTIFY. >> Call-ID: 7f817a77-30df0af6-2e84e...@192.168.1.47. >> Content-Length: 583. >> UA: Test SIP Proxy. >> Max-Forwards: 70. >> Event: presence. >> Contact: . >> Subscription-State: active;expires=3136. >> Content-Type: application/pidf+xml. >> . >> >> > xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" >> xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" >> xmlns:c="urn:ietf:params:xml:ns:pidf:cipid" >> entity="10001...@sipdev1.test.com">> id="0x7fff82d20c50">open> >> id="0x7fff82d20c50">On >> the Phone> xmlns="urn:ietf:params:xml:ns:pidf" >> >> id="0x7fff1b75a760">open >> >> >> >> >> On Sun, Apr 25, 2010 at 11:14 PM, Anca Vamanu wrote: >> >>> >>> Hi Julian, >>> >>> Putting inuse as basic status is not really legal - since the pidf RFC >>> says that you can have only open and close >>> (http://www.faqs.org/rfcs/rfc3863.html). >>> >>> Regards, >>> >>> -- >>> Anca Vamanu >>> www.voice-system.ro >>> >>> >>> Julian Yap wrote: >>> >>>> >>>> I'm trying to hack together my own presence server. I have PUA USRLOC >>>> working fine. >>>> >>>> So then I just want to be able to generate the 'phone in use' and >>>> 'phone not in use' PUBLISH messages via PUA MI. >>>> >>>> Here's what I have in my pua_publish FIFO example: >>>> --- >>>> :pua_publish:test_fifo >>>> sip:10001...@sipdev1.me.com >>>> 3600 >>>> presence >>>> application/pidf+xml >>>> . >>>> . >>>> >>> xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' >>>> xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' >>>> xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' >>>> >>>> entity='10001...@sipdev1.me.com'>inuse >>>> >>>> --- >>>> >>>> The actual NOTIFY goes out as: >>>> >>>> >>> xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" >>>> xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" >>>> xmlns:c="urn:ietf:params:xml:ns:pidf:cipid" >>>> entity="10001...@sipdev1.me.com">>>> id="0x7fff9d908330">inuse>>> xmlns="urn:ietf:params:xml:ns:pidf" >>>> >>>> id="0x7fff1b75a760">open >>>> >>>> >>>> >>>> ... So the status gets overwritten as 'open'. >>>> >>>> Any tips? What would a working 'phone is in use' and 'phone is not in >>>> use' FIFO look like? >>>> >>>> Thanks! >>>> >>>> Julian >>>> >>>> ___ >>>> 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] PUA MI and pua_publish
Hi Anca, Yeah I upgraded to the latest Polycom firmware 3.2.3 (see User Agent below). It seems Polycom only accepts application/xpidf+xml and text/xml+msrtc.pidf. If I send if "application/pidf+xml" it doesn't work, even it I forward on a ""application/pidf+xml" NOTIFY from valid UA like X-Lite. If you check out: http://fisheye.freeswitch.org/browse/~raw,r=17178/FreeSWITCH/src/mod/endpoints/mod_sofia/sofia_presence.c The search for 'polycom'. You'll see that it sends out "application/xpidf+xml" specifically for Polycom phones. This is the SUBSCRIBE that a Polycom sends out: SUBSCRIBE sip:10001...@sipdev1.test.com SIP/2.0. Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK4f059433100DF28E. From: "10001007" ;tag=3965CFF7-15338CF2. To: . CSeq: 1 SUBSCRIBE. Call-ID: b35e2521-9206c4ac-8639c...@192.168.1.47. Contact: . Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. Event: presence. User-Agent: PolycomSoundPointIP-SPIP_450-UA/3.2.3.1734. Accept-Language: en. Accept: application/xpidf+xml,text/xml+msrtc.pidf. Max-Forwards: 70. Expires: 3600. Content-Length: 0. - Julian On Mon, Apr 26, 2010 at 2:31 AM, Anca Vamanu wrote: > Julian Yap wrote: >> >> I guess Polycom phones only accept application/xpidf+xml. >> >> >> > > Hi Julian, > > I kept searching what "application/xpidf+xml" is and it was quite hard to > find since it seems to be something not used any more. I found in the ietf > draft preceding the PIDF RFC > (http://www.jdrosen.net/papers/draft-rosenberg-impp-pidf-00.txt) the > following statement: > > PIDF documents have the MIME type "application/xpidf+xml". > > > So it seems to be a proposition present in the draft but that was latter > dropped when passing to the RFC ( now the mime type is application/pidf+xml > ) > > Strange that Polycom have such an old implementation of presence maybe you > should update your firmare? ). > > And related to what you asked, you can not configure presence module to > change the Content Type of Notify depending on the destination. But maybe > you can hack it in a local route, catch the generated Notify, remove the > Content-Type header and add a new one. But I don't know if the this is > sufficient and the document itself will be accepted by the phone. > > Regards, > > -- > Anca Vamanu > www.voice-system.ro > > > >> On Mon, Apr 26, 2010 at 1:18 AM, Anca Vamanu wrote: >> >>> >>> Hi Julian, >>> >>> Julian Yap wrote: >>> >>>> >>>> Hmm, I have this working now using SIP SIMPLE standard (by testing >>>> with X-Lite user agent)... Unfortunately this doesn't look to be >>>> supported by Polycom phones. Is there a way to translate the PUBLISH >>>> so the NOTIFY will go out in an appropriate format depending on the >>>> UA? >>>> >>>> >>>> >>> >>> I don't understand what you mean by translate the Publish in an >>> appropriate >>> format.. What other format do you want to have? >>> >>> Regards, >>> >>> -- >>> Anca Vamanu >>> www.voice-system.ro >>> >>> >>> >>>> >>>> eg. PUBLISH is generated server side using MI: >>>> :pua_publish:test_fifo >>>> sip:10001...@sipdev1.test.com >>>> 3600 >>>> presence >>>> application/pidf+xml >>>> . >>>> . >>>> >>> xmlns='urn:ietf:params:xml:ns:pidf' >>>> xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' >>>> xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' >>>> xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' >>>> >>>> >>>> entity='10001...@sipdev1.test.com'>openOn >>>> the Phone >>>> >>>> >>>> SIP packet: >>>> U 2010/04/26 00:49:32.590822 64.xxx.yyy.41:5060 -> 64.xxx.yyy.41:5060 >>>> PUBLISH sip:10001...@sipdev1.test.com SIP/2.0. >>>> Via: SIP/2.0/UDP 64.xxx.yyy.41;branch=z9hG4bK4d6c.8543c9f1.0. >>>> To: sip:10001...@sipdev1.test.com. >>>> From: >>>> >>>> ;tag=cdedd5270a3714d043d2a63f64a8b9e2-bc47. >>>> CSeq: 10 PUBLISH. >>>> Call-ID: 39497fd430730179-10...@64.xxx.yyy.41. >>>> Content-Length: 476. >>>> UA: Test SIP Proxy. >>>> Max-Forwards: 70. >>>> Event: presence. >>>> Expires: 3601. >>>> SIP-If-Match: a.12
Re: [OpenSIPS-Users] PUA MI and pua_publish
Hi Anca,
Thanks for your help. Unfortunately it doesn't work. Let me know how
I can help in testing.
Here's the local route I added which rewrites the NOTIFY appropriately:
local_route
{
if(is_method("NOTIFY"))
{
if(is_present_hf("Content-Type"))
{
xlog("L_INFO", "rewriting Content-Type \n");
remove_hf("Content-Type");
append_hf("Content-Type: application/xpidf+xml\r\n");
};
};
}
On Mon, Apr 26, 2010 at 9:42 PM, Anca Vamanu wrote:
> Hi Julian,
>
> Can you please try what I indicated in the previous mail - replacing the
> Content-Type header for Notify to Polycom in local route? And then let me
> know if it works. This is to check if it is just a matter of content type
> name or if the body itself should also be different. Then I might also
> include this case in the code as sofia does.
>
> Regards,
>
> --
> Anca Vamanu
> www.voice-system.ro
>
>
> Julian Yap wrote:
>>
>> Hi Anca,
>>
>> Yeah I upgraded to the latest Polycom firmware 3.2.3 (see User Agent
>> below).
>>
>> It seems Polycom only accepts application/xpidf+xml and
>> text/xml+msrtc.pidf. If I send if "application/pidf+xml" it doesn't
>> work, even it I forward on a ""application/pidf+xml" NOTIFY from valid
>> UA like X-Lite.
>>
>> If you check out:
>>
>> http://fisheye.freeswitch.org/browse/~raw,r=17178/FreeSWITCH/src/mod/endpoints/mod_sofia/sofia_presence.c
>>
>> The search for 'polycom'. You'll see that it sends out
>> "application/xpidf+xml" specifically for Polycom phones.
>>
>> This is the SUBSCRIBE that a Polycom sends out:
>> SUBSCRIBE sip:10001...@sipdev1.test.com SIP/2.0.
>> Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK4f059433100DF28E.
>> From: "10001007" ;tag=3965CFF7-15338CF2.
>> To: .
>> CSeq: 1 SUBSCRIBE.
>> Call-ID: b35e2521-9206c4ac-8639c...@192.168.1.47.
>> Contact: .
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
>> NOTIFY, PRACK, UPDATE, REFER.
>> Event: presence.
>> User-Agent: PolycomSoundPointIP-SPIP_450-UA/3.2.3.1734.
>> Accept-Language: en.
>> Accept: application/xpidf+xml,text/xml+msrtc.pidf.
>> Max-Forwards: 70.
>> Expires: 3600.
>> Content-Length: 0.
>>
>> - Julian
>>
>> On Mon, Apr 26, 2010 at 2:31 AM, Anca Vamanu wrote:
>>
>>>
>>> Julian Yap wrote:
>>>
>>>>
>>>> I guess Polycom phones only accept application/xpidf+xml.
>>>>
>>>>
>>>>
>>>>
>>>
>>> Hi Julian,
>>>
>>> I kept searching what "application/xpidf+xml" is and it was quite hard
>>> to
>>> find since it seems to be something not used any more. I found in the
>>> ietf
>>> draft preceding the PIDF RFC
>>> (http://www.jdrosen.net/papers/draft-rosenberg-impp-pidf-00.txt) the
>>> following statement:
>>>
>>> PIDF documents have the MIME type "application/xpidf+xml".
>>>
>>>
>>> So it seems to be a proposition present in the draft but that was latter
>>> dropped when passing to the RFC ( now the mime type is
>>> application/pidf+xml
>>> )
>>>
>>> Strange that Polycom have such an old implementation of presence maybe
>>> you
>>> should update your firmare? ).
>>>
>>> And related to what you asked, you can not configure presence module to
>>> change the Content Type of Notify depending on the destination. But maybe
>>> you can hack it in a local route, catch the generated Notify, remove the
>>> Content-Type header and add a new one. But I don't know if the this is
>>> sufficient and the document itself will be accepted by the phone.
>>>
>>> Regards,
>>>
>>> --
>>> Anca Vamanu
>>> www.voice-system.ro
>>>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] T38 and changing media port for T38 SDP
Any help greatly appreciated! I'm having problems with a T38 UA which changes port when negotiating T38 media. All the other UA's I've encountered thus far use the same RTP port throughout. I'm also using RTPProxy. In the final 200 OK SDP, the UA changes media port from 49200 to 49152 but this changeover isn't detected and the media is sent back to port 49200 so the call then fails to negotiate T38 properly. Not sure how to log the port changes to further debug this issue as well. Here is the flow: | UA | OpenSIPS | T38 GW | | INVITE SDP ( g711U) | | |(5060) --> (5060) | | | 100 Trying| | | |(5060) <-- (5060) | | | | INVITE SDP ( g711U) | | |(5060) --> (5060) | | | 100 Trying| | | |(5060) <-- (5060) | | | 180 Ringing SDP ( g711U) | | |(5060) <-- (5060) | | | 200 OK SDP ( g711U) | | |(5060) <-- (5060) | | | RTP (g711U) | | |(11392) <-- (14110) | | RTP (g711U) | | |(49200) <-- (10878) | | | 180 Ringing SDP ( g711U) | | |(5060) <-- (5060) | | | RTP (g711U) | | |(49200) --> (10878) | | | | RTP (g711U) | | |(11392) --> (14110) | | 200 OK SDP ( g711U) | | |(5060) <-- (5060) | | | ACK | | | |(5060) --> (5060) | | | RTP (g711U) | | |(49200) --> (10878) | | | RTP (g711U) | | |(49200) <-- (10878) | | | | 200 OK SDP ( g711U) | | |(5060) <-- (5060) | | | RTP (g711U) | | |(11392) --> (14110) | | | ACK | | | |(5060) --> (5060) | | 200 OK SDP ( g711U) | | |(5060) <-- (5060) | | | ACK | | | |(5060) --> (5060) | | | RTP (g711U) | | |(49200) --> (10878) | | | RTP (g711U) | | |(49200) <-- (10878) | | | | ACK | | | |(5060) --> (5060) | | | INVITE SDP ( t38) | | |(5060) <-- (5060) | | INVITE SDP ( t38) | | |(5060) <-- (5060) | | | 200 OK SDP ( t38) | | |(5060) --> (5060) | | This is where it sends the 200 OK with a different media port. - Julian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] T38 and changing media port for T38 SDP
Hi Pavel, Yeah, in my example, the media port is changed on the T38 reinvite. Isn't RTPProxy meant to handle this port change? - Julian 2010/7/3 Pavel Eremin : > you may change and control content of SDP body by module "textops". So, if > port is change you may change it back or log it or smth else... > i use it when Broadworks change media port for G711 to 0... > > Julian Yap писал(а) в своём письме Sat, 03 Jul > 2010 03:37:48 +0600: > >> Any help greatly appreciated! >> >> I'm having problems with a T38 UA which changes port when negotiating >> T38 media. All the other UA's I've encountered thus far use the same >> RTP port throughout. >> >> I'm also using RTPProxy. >> >> In the final 200 OK SDP, the UA changes media port from 49200 to 49152 >> but this changeover isn't detected and the media is sent back to port >> 49200 so the call then fails to negotiate T38 properly. Not sure how >> to log the port changes to further debug this issue as well. >> >> Here is the flow: >> | UA | OpenSIPS | T38 GW | >> | INVITE SDP ( g711U) | | >> |(5060) --> (5060) | | >> | 100 Trying| | | >> |(5060) <-- (5060) | | >> | | INVITE SDP ( g711U) | >> | |(5060) --> (5060) | >> | | 100 Trying| | >> | |(5060) <-- (5060) | >> | | 180 Ringing SDP ( g711U) | >> | |(5060) <-- (5060) | >> | | 200 OK SDP ( g711U) | >> | |(5060) <-- (5060) | >> | | RTP (g711U) | >> | |(11392) <-- (14110) | >> | RTP (g711U) | | >> |(49200) <-- (10878) | | >> | 180 Ringing SDP ( g711U) | | >> |(5060) <-- (5060) | | >> | RTP (g711U) | | >> |(49200) --> (10878) | | >> | | RTP (g711U) | >> | |(11392) --> (14110) | >> | 200 OK SDP ( g711U) | | >> |(5060) <-- (5060) | | >> | ACK | | | >> |(5060) --> (5060) | | >> | RTP (g711U) | | >> |(49200) --> (10878) | | >> | RTP (g711U) | | >> |(49200) <-- (10878) | | >> | | 200 OK SDP ( g711U) | >> | |(5060) <-- (5060) | >> | | RTP (g711U) | >> | |(11392) --> (14110) | >> | | ACK | | >> | |(5060) --> (5060) | >> | 200 OK SDP ( g711U) | | >> |(5060) <-- (5060) | | >> | ACK | | | >> |(5060) --> (5060) | | >> | RTP (g711U) | | >> |(49200) --> (10878) | | >> | RTP (g711U) | | >> |(49200) <-- (10878) | | >> | | ACK | | >> | |(5060) --> (5060) | >> | | INVITE SDP ( t38) | >> | |(5060) <-- (5060) | >> | INVITE SDP ( t38) | | >> |(5060) <-- (5060) | | >> | 200 OK SDP ( t38) | | >> |(5060) --> (5060) | | >>
Re: [OpenSIPS-Users] T38 and changing media port for T38 SDP
Hi All, I managed to get it working by adding a whole lot of debugging and a whole lot of rtpproxy_offer() and rtpproxy_answer(). Now I need to clean up my config because it's a total mess. - Julian On Tue, Jul 6, 2010 at 10:07 AM, Bogdan-Andrei Iancu wrote: > Hi Julian, > > Do you trigger again the rtpproxy at re-INVITE time ? basically you need > to do the insertion of rtpproxy again, as for the initial INVITE. > > Regards, > Bogdan > > Julian Yap wrote: >> Any help greatly appreciated! >> >> I'm having problems with a T38 UA which changes port when negotiating >> T38 media. All the other UA's I've encountered thus far use the same >> RTP port throughout. >> >> I'm also using RTPProxy. >> >> In the final 200 OK SDP, the UA changes media port from 49200 to 49152 >> but this changeover isn't detected and the media is sent back to port >> 49200 so the call then fails to negotiate T38 properly. Not sure how >> to log the port changes to further debug this issue as well. >> >> Here is the flow: >> | UA | OpenSIPS | T38 GW | >> | INVITE SDP ( g711U) | | >> |(5060) --> (5060) | | >> | 100 Trying| | | >> |(5060) <-- (5060) | | >> | | INVITE SDP ( g711U) | >> | |(5060) --> (5060) | >> | | 100 Trying| | >> | |(5060) <-- (5060) | >> | | 180 Ringing SDP ( g711U) | >> | |(5060) <-- (5060) | >> | | 200 OK SDP ( g711U) | >> | |(5060) <-- (5060) | >> | | RTP (g711U) | >> | |(11392) <-- (14110) | >> | RTP (g711U) | | >> |(49200) <-- (10878) | | >> | 180 Ringing SDP ( g711U) | | >> |(5060) <-- (5060) | | >> | RTP (g711U) | | >> |(49200) --> (10878) | | >> | | RTP (g711U) | >> | |(11392) --> (14110) | >> | 200 OK SDP ( g711U) | | >> |(5060) <-- (5060) | | >> | ACK | | | >> |(5060) --> (5060) | | >> | RTP (g711U) | | >> |(49200) --> (10878) | | >> | RTP (g711U) | | >> |(49200) <-- (10878) | | >> | | 200 OK SDP ( g711U) | >> | |(5060) <-- (5060) | >> | | RTP (g711U) | >> | |(11392) --> (14110) | >> | | ACK | | >> | |(5060) --> (5060) | >> | 200 OK SDP ( g711U) | | >> |(5060) <-- (5060) | | >> | ACK | | | >> |(5060) --> (5060) | | >> | RTP (g711U) | | >> |(49200) --> (10878) | | >> | RTP (g711U) | | >> |(49200) <-- (10878) | | >> | | ACK | | >> | |(5060) --> (5060) | >> | | INVITE SDP ( t38) | >> | |(5060) <-- (5060) | >> | INVITE SDP ( t38) | | >> |(5060) <-- (5060) | | >> | 200 OK SDP ( t38) | | >> |(5060) --> (5060) |
[OpenSIPS-Users] Delay in relaying SIP message to UA
I'm having a problem where is looks like there is a delay in OpenSIPS relaying a SIP packet back to the originating UA. Here is a stripped Ngrep trace which shows the time stamps. In the example: 9.9.9.101 = IP of the Calling UA 8.8.8.41 = IP of the Proxy 9.9.9.15 = IP of the Called UA The 180 Ringing is received by the Proxy at 16:00:58.628464 but is only relayed on at 16:00:58.918988. Trace: U 2010/07/13 16:00:58.560395 9.9.9.101:10673 -> 8.8.8.41:5060 INVITE sip:5551...@my.proxy.com;user=phone SIP/2.0. -- U 2010/07/13 16:00:58.560599 8.8.8.41:5060 -> 9.9.9.101:10673 SIP/2.0 100 Trying. -- U 2010/07/13 16:00:58.560658 8.8.8.41:5060 -> 9.9.9.101:10673 SIP/2.0 407 Proxy Authentication Required. -- U 2010/07/13 16:00:58.588648 9.9.9.101:10673 -> 8.8.8.41:5060 ACK sip:5551...@my.proxy.com SIP/2.0. -- U 2010/07/13 16:00:58.592480 9.9.9.101:10673 -> 8.8.8.41:5060 INVITE sip:5551...@my.proxy.com;user=phone SIP/2.0. -- U 2010/07/13 16:00:58.592592 8.8.8.41:5060 -> 9.9.9.101:10673 SIP/2.0 100 Trying. -- U 2010/07/13 16:00:58.594010 8.8.8.41:5060 -> 9.9.9.15:5060 INVITE sip:+18085551...@9.9.9.15 SIP/2.0. -- U 2010/07/13 16:00:58.598573 9.9.9.15:5060 -> 8.8.8.41:5060 SIP/2.0 100 Trying. -- U 2010/07/13 16:00:58.628464 9.9.9.15:5060 -> 8.8.8.41:5060 SIP/2.0 180 Ringing. -- U 2010/07/13 16:00:58.628780 9.9.9.15:5060 -> 8.8.8.41:5060 SIP/2.0 200 OK. -- U 2010/07/13 16:00:58.918988 8.8.8.41:5060 -> 9.9.9.101:10673 SIP/2.0 180 Ringing. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Delay in relaying SIP message to UA
This is strange. I only get the issues if I use rtpproxy_offer(). If I use force_rtp_proxy() then there are no delays. - Julian On Tue, Jul 13, 2010 at 4:43 PM, Julian Yap wrote: > I'm having a problem where is looks like there is a delay in OpenSIPS > relaying a SIP packet back to the originating UA. > > Here is a stripped Ngrep trace which shows the time stamps. > > In the example: > 9.9.9.101 = IP of the Calling UA > 8.8.8.41 = IP of the Proxy > 9.9.9.15 = IP of the Called UA > > The 180 Ringing is received by the Proxy at 16:00:58.628464 but is > only relayed on at 16:00:58.918988. > > Trace: > > U 2010/07/13 16:00:58.560395 9.9.9.101:10673 -> 8.8.8.41:5060 > INVITE sip:5551...@my.proxy.com;user=phone SIP/2.0. > -- > U 2010/07/13 16:00:58.560599 8.8.8.41:5060 -> 9.9.9.101:10673 > SIP/2.0 100 Trying. > -- > U 2010/07/13 16:00:58.560658 8.8.8.41:5060 -> 9.9.9.101:10673 > SIP/2.0 407 Proxy Authentication Required. > -- > U 2010/07/13 16:00:58.588648 9.9.9.101:10673 -> 8.8.8.41:5060 > ACK sip:5551...@my.proxy.com SIP/2.0. > -- > U 2010/07/13 16:00:58.592480 9.9.9.101:10673 -> 8.8.8.41:5060 > INVITE sip:5551...@my.proxy.com;user=phone SIP/2.0. > -- > U 2010/07/13 16:00:58.592592 8.8.8.41:5060 -> 9.9.9.101:10673 > SIP/2.0 100 Trying. > -- > U 2010/07/13 16:00:58.594010 8.8.8.41:5060 -> 9.9.9.15:5060 > INVITE sip:+18085551...@9.9.9.15 SIP/2.0. > -- > U 2010/07/13 16:00:58.598573 9.9.9.15:5060 -> 8.8.8.41:5060 > SIP/2.0 100 Trying. > -- > U 2010/07/13 16:00:58.628464 9.9.9.15:5060 -> 8.8.8.41:5060 > SIP/2.0 180 Ringing. > -- > U 2010/07/13 16:00:58.628780 9.9.9.15:5060 -> 8.8.8.41:5060 > SIP/2.0 200 OK. > -- > U 2010/07/13 16:00:58.918988 8.8.8.41:5060 -> 9.9.9.101:10673 > SIP/2.0 180 Ringing. > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] T38 and changing media port for T38 SDP
Yeah, I am having way too many issues with RTPProxy and the re-invite port changing. It's an absolute nightmare. Everything works fine when the UA doesn't change media port but when it does, it's really inconsistent. Sometimes it works, sometimes it doesn't work. I think I'm going to try out MediaProxy. MediaProxy is able to detect the changing media port? - Julian On Tue, Jul 6, 2010 at 9:19 PM, Adrian Georgescu wrote: > > On Jul 7, 2010, at 12:27 AM, Julian Yap wrote: > >> Hi All, >> >> I managed to get it working by adding a whole lot of debugging and a >> whole lot of rtpproxy_offer() and rtpproxy_answer(). >> >> Now I need to clean up my config because it's a total mess. >> > > You may want to try use mediaproxy if you do not like the mess. You > need a single line of code in your opensips configuration to achieve > this. > > Adrian > > >> - Julian >> >> On Tue, Jul 6, 2010 at 10:07 AM, Bogdan-Andrei Iancu >> wrote: >>> Hi Julian, >>> >>> Do you trigger again the rtpproxy at re-INVITE time ? basically you >>> need >>> to do the insertion of rtpproxy again, as for the initial INVITE. >>> >>> Regards, >>> Bogdan >>> >>> Julian Yap wrote: >>>> Any help greatly appreciated! >>>> >>>> I'm having problems with a T38 UA which changes port when >>>> negotiating >>>> T38 media. All the other UA's I've encountered thus far use the >>>> same >>>> RTP port throughout. >>>> >>>> I'm also using RTPProxy. >>>> >>>> In the final 200 OK SDP, the UA changes media port from 49200 to >>>> 49152 >>>> but this changeover isn't detected and the media is sent back to >>>> port >>>> 49200 so the call then fails to negotiate T38 properly. Not sure >>>> how >>>> to log the port changes to further debug this issue as well. >>>> >>>> Here is the flow: >>>> | UA | OpenSIPS | T38 GW | >>>> | INVITE SDP ( g711U) | | >>>> |(5060) --> (5060) | | >>>> | 100 Trying| | | >>>> |(5060) <-- (5060) | | >>>> | | INVITE SDP ( g711U) | >>>> | |(5060) --> (5060) | >>>> | | 100 Trying| | >>>> | |(5060) <-- (5060) | >>>> | | 180 Ringing SDP ( g711U) | >>>> | |(5060) <-- (5060) | >>>> | | 200 OK SDP ( g711U) | >>>> | |(5060) <-- (5060) | >>>> | | RTP (g711U) | >>>> | |(11392) <-- (14110) | >>>> | RTP (g711U) | | >>>> |(49200) <-- (10878) | | >>>> | 180 Ringing SDP ( g711U) | | >>>> |(5060) <-- (5060) | | >>>> | RTP (g711U) | | >>>> |(49200) --> (10878) | | >>>> | | RTP (g711U) | >>>> | |(11392) --> (14110) | >>>> | 200 OK SDP ( g711U) | | >>>> |(5060) <-- (5060) | | >>>> | ACK | | | >>>> |(5060) --> (5060) | | >>>> | RTP (g711U) | | >>>> |(49200) --> (10878) | | >>>> | RTP (g711U) | | >>>> |(49200) <-- (10878) | | >>>> | | 200 OK SDP ( g711U) | >>>> | |(5060) <-- (5060) | >>>> | | RTP (g711U)
Re: [OpenSIPS-Users] T38 and changing media port for T38 SDP
On Wed, Jul 14, 2010 at 11:14 PM, Adrian Georgescu wrote: > > On Jul 15, 2010, at 11:59 AM, Thomas Gelf wrote: > >> Am 15.07.2010 09:26, schrieb Adrian Georgescu: MediaProxy is able to detect the changing media port? >>> >>> If the port change is present in the signaling MP will act upon it >>> accordingly. >> >> I can confirm this, I'm intensively T.38 with and without Mediaproxy. >> And YES, there are such ReINVITEs - changing IPs and Ports... >> > > Thanks for the confirmation RTPProxy has been working nicely for me thus far but I've introduced a new Fax Server which doesn't play nicely with RTPProxy for some reason :(. >>> You only need to use a single line in your config for enabling mediaproxy >>> so is very easy to test it. >> >> Please note that once you are using multiple parallel / serial branches >> this could no longer be true, you could be forced to manually "use/stop" >> Mediaproxy on your branch routes instead of just calling "engage". > > Right, no all corner cases are captured by the logic of engage method. You mean 'Now all corner cases'? So I would use engage_media_proxy() and also need to implement the dialog module? Does MediaProxy require 2 servers. One for OpenSIPS and one for the relay? Dispatcher can run on the same box as OpenSIPS? Or can it all go on 1 server. >> However, I'd in most cases still prefer to do so instead of using >> Rtpproxy ;-) Any particular reasons against using RTPProxy? - Julian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] T38 and changing media port for T38 SDP
Holy shit (pardon my French and excitement), I just ran through the install of MediaProxy on my development server and it all works! All my initial testing works. IT JUST WORKS. What a great piece of software! I was initially put off by the difficult install of MediaProxy and all the hype around RTPProxy being written in C but I can tell you I spent a lot of time trying to get RTPProxy working... And for a long time it was working until I brought on my new T38 Fax server and stuff just didn't work in any consistent and logical fashion. I've been beating my head against a wall for a week. MediaProxy works, RTPProxy doesn't work. Simple as that. Thank you! - Julian On Thu, Jul 15, 2010 at 9:03 PM, Saúl Ibarra Corretgé wrote: > Hi, > >> You mean 'Now all corner cases'? So I would use engage_media_proxy() >> and also need to implement the dialog module? >> > > No, there are two ways of using mediaproxy: > - Using the dialog module triggered function engage_mediaproxy: the > dialog module enables or disables MediaProxy by calling functions > internally. You only need to call this function once for the original > INVITE. > - Using the individual function use_media_proxy and > end_media_session: you'll need to call then manually in your routing script. > >> Does MediaProxy require 2 servers. One for OpenSIPS and one for the >> relay? Dispatcher can run on the same box as OpenSIPS? Or can it all >> go on 1 server. >> > > MedisPRoxy consists of two components: the dispatcher and the relay. The > dispatcher must be installer in the same box as OpenSIPS, but the relay > can be installed on a separate server. You may install both components > in the same server. > > > Regards, > > -- > Saúl Ibarra Corretgé > AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
