Re: [OpenSIPS-Users] explizit handling auf replyto
Bogdan-Andrei Iancu wrote:
2) if the requests was statelessly forwarded (via forward() ), the VIA
stack (in received reply) will contain all the info to route back the reply
I think the question is whether stateless forwarding can be used to
override default processing of Via and route the reply somewhere else.
You can, for example, do this (whether in stateless or stateful request
forwarding mode):
onreply_route[1] {
drop;
}
... I think it's along that general train of thought.
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Newbie To OpenSIPS
Kemp, Larry wrote: Can OpenSIPS be used as a Session Border controller sitting at my edge passing and receiving SIP traffic to others I SIP peer with? If not, what other open-source would anyone suggest to act as SBC's? I too would rather do it via open-source and x86 or 64bit chip, less costly. Thanks. This is mostly a semantic issue. OpenSIPS is not an SBC in the way that commercial SBCs are SBCs, and lacks a number of key aspects, including ASIC-assisted processing in the higher-end ones. But it can be used for subscriber or carrier-facing edge duty in the same way SBCs are often used (unnecessarily so). -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] B2BUA module question
Brett Nemeroff wrote: Question about the direction of the B2BUA module. I know one of the key feature is topology hiding. Does this also occur in the SDP? I would expect that it would need to still be paired with something like mediaproxy or rtpproxy to achieve topology hiding with SDP as well, is this correct? Yes. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Kamailio Opensser Call forwarding
Iñaki Baz Castillo wrote: El Jueves, 20 de Agosto de 2009, happyalways escribió: Hiii..I installed mysql5.o...and Kamailio 1.5 succesfully...Authentication is working properly. Next i'm going through blind call forwarding. I need your help in configuring. Please provide me the configuartion file for blind call forwarding. Sure, but first I need you to provide me some amount of money. Thanks. And after that amount is provided, please provide me the name and number of your principal so I can have a conversation with him about cheating on your homework assignments. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP Trunking
Matthew, Look for the mediaproxy module. That said, do be aware that a proxy is, by definition, not like an SBC. SBCs have many other capabilities a proxy does not; a proxy is a relatively thin interoperation layer. Perhaps the recently introduced b2bua module is brought to bear on that somewhat, but classically, OpenSIPS is a proxy. -- Alex Matthew S. Crocker wrote: Hello, I'm brand new to OpenSIPS, just going through the make process now. I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch. Where should I look for Documentation/Examples of a working config? Here is my scenario: OpenSIPS has two interfaces, private public. VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing) End user has an Asterisk server on a private lan behind their firewall (NAT) I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall. It then need to rewrite the SIP message and send it to the Gateway. The Gateway would see the messages coming from the internal IP of the OpenSIPS server. Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway. Any helpful hints on where to look? -Matt -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Module Path and function loose_route
Iñaki Baz Castillo wrote: El Jueves, 20 de Agosto de 2009, mayamatakeshi escribió: There is only one exception: If the request is out-of-dialog (no to-tag) and there is only one Route: header indicating the local proxy, then the Route: header is removed and the function returns FALSE. But why does it return FALSE? Because if an initial request (no To-tag) has a single Route header pointing to the proxy handling it, it's useless. That's correct - initial INVITEs (and all initial requests) are different than in-dialog requests (requests arising within a dialog created by the initial requests). They are routed manually, not using loose_route() in any way. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP Trunking
Matthew S. Crocker wrote: Can mediaproxy glue two RTP streams together (CallerA to CallerB)? Can mediaproxy glue two RTP streams together from different interfaces/IPs (eth0 eth1) ? Yes. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CODEC
smadhoo6 wrote: How to configure Opensips (version 1.5.0) to use a particular CODEC say.. Speex.? This is like asking how to put the milk back in the cow with JSON. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] handling a 422
Jeff Pyle wrote: For example, my SST module min_se value is set to 300. Let's say a far-end device responds with a 422 that contains Min-SE: 1800. Is there a way within Opensips to handle this and re-relay the call with an adjusted Min-SE/Session Expires header? Sure, use a failure route and append_branch(). -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] handling a 422
The SST module is designed for a scenario in which the proxy serves as the endpoint of the SST negotiation. Otherwise, SST is up to the UA endpoints to negotiate amongst themselves. So, SST does not deal with a situation in which the proxy *receives* a 422; it only equips the proxy to *send* a 422 if the Min-SE value from the request initiator does not meet *its* desiderata. Jeff Pyle wrote: It seems very strange to me to have to manually manipulate headers that an Opensips module added in the first place. Seems like bad things could happen if the modules expects them to be there with certain values and they have different values or gone altogether. If these headers are added in the request route does the same rule apply as with append_hf(), that is, they cannot be removed? The whole thing just seems odd. - Jeff On 8/18/09 9:01 AM, Alex Balashov abalas...@evaristesys.com wrote: If I'm understanding the documentation correctly, you'd probably have to do this with manual header manipulation. Jeff Pyle wrote: On 8/18/09 8:51 AM, Alex Balashov abalas...@evaristesys.com wrote: Sure, use a failure route and append_branch(). Ok, but how do I adjust the timer value so it doesn't get 422'd again? Or is this handled automatically? The SST module documentation doesn't appear to cover this. - Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] handling a 422
I see what you mean, yeah. Unfortunately, session timers are for the
UAs to negotiate from a design and SIP specification standpoint. The
only thing the SST module provides is a thin layer of SE value
enforcement by the proxy.
In keeping with the sort of thing that a proxy is, it is not an ultimate
negotiating party, as is true of many SIP attributes (SDP/media/codec
parameters, Supported: header values, etc. for example). All it can do
is block requests that do not meet certain SE requests on behalf of a
destination UA. If not for the way the module hooks into the dialog
module callbacks to allow dialog expiry to be connected with SST values,
the entire functionality of the SST module could be replicated thusly:
if(is_present_hf(Min_SE) $hdr(Min-SE) x) {
sl_send_reply(422, Session Timer Too Small);
# append_hf() any other necessary headers.
exit;
}
If the destination UA has a problem, the proxy can't answer on behalf of
the sender.
I agree, it's a sucky situation.
Jeff Pyle wrote:
Right. That was my fear. In my case the UAC knows nothing of session
timers. Its UAS (Opensips) adds the SST headers and relays the request. If
the far end replies with a 422, by default Opensips will relay the 422 to
the UAC who, well, won't know what to do with it.
It just doesn't seem fair to slap the UAC with a 422 it doesn't know how to
handle.
See what I mean?
- Jeff
On 8/18/09 9:10 AM, Alex Balashov abalas...@evaristesys.com wrote:
The SST module is designed for a scenario in which the proxy serves as
the endpoint of the SST negotiation. Otherwise, SST is up to the UA
endpoints to negotiate amongst themselves. So, SST does not deal with a
situation in which the proxy *receives* a 422; it only equips the proxy
to *send* a 422 if the Min-SE value from the request initiator does not
meet *its* desiderata.
Jeff Pyle wrote:
It seems very strange to me to have to manually manipulate headers that an
Opensips module added in the first place. Seems like bad things could
happen if the modules expects them to be there with certain values and they
have different values or gone altogether. If these headers are added in the
request route does the same rule apply as with append_hf(), that is, they
cannot be removed?
The whole thing just seems odd.
- Jeff
On 8/18/09 9:01 AM, Alex Balashov abalas...@evaristesys.com wrote:
If I'm understanding the documentation correctly, you'd probably have to
do this with manual header manipulation.
Jeff Pyle wrote:
On 8/18/09 8:51 AM, Alex Balashov abalas...@evaristesys.com wrote:
Sure, use a failure route and append_branch().
Ok, but how do I adjust the timer value so it doesn't get 422'd again? Or
is this handled automatically? The SST module documentation doesn't
appear
to cover this.
- Jeff
___
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
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] prevent multi-reg
Yes. Set max_contacts parameter in registrar module to 1. Alex G wrote: is there a way to prevent multi-reg of subscribers? ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] prevent multi-reg
Yes - provide a different argument to save(), as Bogdan says. Alex G wrote: is there a way to do this dynamically per subscriber? like an on/off switch? On Thu, Aug 13, 2009 at 10:53 AM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi, actually with OpenSIPS 1.6 (devel), this is no longer a global param, but a per AOR. The save() function takes as parameters a set of flags and one of them is the number of maximum contacts. See http://www.opensips.org/html/docs/modules/devel/registrar.html#id228526 Regards, Bogdan Alex Balashov wrote: Yes. Set max_contacts parameter in registrar module to 1. Alex G wrote: is there a way to prevent multi-reg of subscribers? ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] some idea
Well, I have no idea if your government has certain standards for interception technology and interfaces as the US does with CALEA. But if it's simply the broad goal of providing call interception, I suggest using port mirroring and some open-source recording tool like OrecX (there's also a commercial version), which collates SIP and RTP by watching the headers. It scales fairly well, especially if the use case will not involve attempting to record a very large amount of simultaneous calls. There are other, more OpenSER-native approaches involving media proxies as well. -- Alex josip.djuri...@voljatel.hr wrote: Hi there tnx for quick response. The main idea is to provide intercepting functions to the government, since they press really hard on us, and passive probes are way to expensive we thought about trying to build our own...now that would equire a lot of work, the most hard part would probably be state machine, and connecting sip and rtp together. So if you have any idea on how to acomplish that I and I think many others faced with same challenge would be very gratefull. Best regards, Josip On Fri, 7 Aug 2009 12:59:52 -0400, Alex Balashov abalas...@evaristesys.com wrote: It's certainly possible. But you'd do well to tell us what you're trying to accomplish to get the best advice. -- 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
Re: [OpenSIPS-Users] some idea
Josip Djuricic wrote: Is OrecX source available, or perhaps is it already able to do this (forward required targeted traffic to mediagw or b2bua instead of recording it? ) There is an open-source and a (more featureful) commercial version. I cannot speak in detail to what it can and can't do. One thing you have to keep in mind is that if you use a SIP proxy (like OpenSIPS) for this, it is event-driven, so you can't make it shunt a call to a different place mid-call. -- 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
Re: [OpenSIPS-Users] some idea
I do not know in detail. I know rtpproxy can do some sort of rudimentary call recording. I presume MediaProxy can too, although I am not sure. Josip Djuricic wrote: You mentioned other approach, could you elaborate a little bit more on this approach? There are other, more OpenSER-native approaches involving media proxies as well. Best regards, Josip Alex Balashov wrote: Josip Djuricic wrote: Is OrecX source available, or perhaps is it already able to do this (forward required targeted traffic to mediagw or b2bua instead of recording it? ) There is an open-source and a (more featureful) commercial version. I cannot speak in detail to what it can and can't do. One thing you have to keep in mind is that if you use a SIP proxy (like OpenSIPS) for this, it is event-driven, so you can't make it shunt a call to a different place mid-call. -- 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
Re: [OpenSIPS-Users] some idea
It's certainly possible. But you'd do well to tell us what you're trying to accomplish to get the best advice. -- Sent from mobile device On Aug 7, 2009, at 12:52 PM, josip.djuri...@voljatel.hr wrote: Hi there, I was wondering if there was a way to somehow pipe port mirrored sip calls to opensips, and then rewrite sip fields and forward them to b2bua, or anything similar that could know what to do next with them. I know this is a specific question but it would solve our problems. So the main thing would be to somehow pipe all sip traffic, rewrite sip body and then send it where needed. This is basically just an idea, and I know it doesn't make much sense, but I have to ask for ideas. Best regards, Josip ___ 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] ALIAS_DB: Is there a way to only change the user part of RURI?
I would like to know if is there a way to only change the user part of RURI when doing alias_db_lookup()? Not intrinsically, but you can always store the old domain prior to alias_db_lookup() and then revert to it after the lookup completes. -- 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
Re: [OpenSIPS-Users] Contact header
If we are going to have a cultured and dignified relationship between the Kamailio and OpenSIPS camps, which I assume is the goal of everyone for reasons of commercial self-preservation if nothing else, then the provocations need to stop from both sides. No, it is not very upstanding to come on the OpenSIPS list only to remind its members that you don't use OpenSIPS and that Kamailio is much better. Whether you think it's true or not, the OpenSIPS list is not the appropriate forum in which to air that thought; it's just not polite. The values and focus of every community must be respected, and this mailing list belongs to the OpenSIPS community and development team. There's a certain degree of when in Rome... that should be obeyed. I'm a very committed Debian user, and intensely dislike Redhat-derived distributions. But if I am on a mailing list centered chiefly around Fedora, CentOS, RHEL, etc. or products based on them, it's just not my place to bring up Debian or invite an RH vs. Debian flame war. That's just not what the list is for, and my ability to join it and ask a question is a privilege, not a right. That having been said, the provocations need to stop from both sides as I said above. That includes tongue-in-cheek comments that imply Kamailio defects or fundamental technical or political inferiorities, or ones that attempt to explain user perceptions of OpenSIPS in an ad hominem manner by way of some kind of Kamailio affiliation or anything like that. Just don't do it. It's bad for business, it's bad for both products, it's bad for everyone. NOBODY wins if commercial adopters see this kind of petty bickering and egotism, especially from lead developers and other significant stakeholders in the commercial ecosystem. -- 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
Re: [OpenSIPS-Users] Opensips + Dispatcher + asterisk + fail over problem
urmi lakkad wrote: modparam(dispatcher, ds_ping_method, INFO) Asterisk does not respond to these. Try using the OPTIONS method instead. -- 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
Re: [OpenSIPS-Users] Contact header
Bogdan-Andrei Iancu wrote: for hiding the topology, you do not really need to create a new call, but simple to hide some information from the messagessomething that a proxy can do in a more efficient way. Albeit, in a way that entirely breaks proxy spec, since the proxy isn't supposed to statefully hide anything. :-) -- 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
Re: [OpenSIPS-Users] mailing list reply-to : list or sender
I would favour This List. This is the way that most mailing lists seem to work in my experience, and it does the maximum to encourage discussion to stay on the list -- which is good, and serves the aim of keeping the discussion public, on record, archivable and searchable. I get a lot of private replies from people forgetting to post back to the list as well. Bogdan-Andrei Iancu wrote: normal is a fuzzy word. I can do the change, not a problem, but just to past what mailman says: Where are replies to list messages directed? Poster is *strongly* recommended for most mailing lists. - Poster This list Explicit address Currently is Poster(which is recommended) and you suggest This List. To be honest I consider logical the Poster setting as primarily I'm talking with a person on the list and not to the list. But any other opinions are welcome here. Regards, Bogdan Raúl Alexis Betancor Santana wrote: Please ... could be possible to setup the mailling list as normal mailling lists that with Reply just send the reply to the list and not to personal inbox .. ? ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [NEW] SDP codec manipulation feature
Olle E. Johansson wrote: As far as I know, there's no way in SIP you can determine what codec actually was used if the offer/answer resultet in multiple codecs. I was just going to say that. Even if you mimic the exact algorithm used by the offer and answer side, since there is no knowledge of their intrinsic codec capability set, there's no way to know what the decision rendered ultimately is. Also note that during a call, the codec may change. By means other than re-INVITEs? (Which can also be inspected for SDP.) -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [NEW] SDP codec manipulation feature
It's worth pointing out that no member of the OpenSER project stack has been a pure SIP proxy for very long; they have certain UAS features (registrar, PUA, NAT ping, etc.) As Bogdan said, a pure proxy would not be terribly useful in most scenarios in which the project is deployed. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Forcing reinvite for t38 on different pstn gw
Uwe Kastens wrote: Ok, thats not possible with T38, since the codec is 1st established as normale codec. If one of the devices gets a fax ton it will iniitate a reinvite with t38. Yep; so, you need to send the call to a device that supports both regular codec as well as T.38 and can make the switch. -- 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
Re: [OpenSIPS-Users] new CDRTool release 6.9.0
ram wrote: Hi Adrian I found some problem last version when iam patching freeradius.patch with Freeradius 2.1.6 is that Fixed in this version I have installed Freeradius 2.0.4 I would wager, on Adrian's behalf, that a description of some problem or a reference to a bug report number would be necessary to answer this question. -- 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
Re: [OpenSIPS-Users] Forcing reinvite for t38 on different pstn gw
Uwe Kastens wrote: Is it possible to handle reinvites in that way, that I can send them to a special pstn gw? This looks a little tricky, since I need to drop the 1st invite. No, that would not be in the slightest bit compatible with SIP protocol mechanics as described per the RFC. The initial INVITE establishes the dialog, and without that initial request there cannot be sequential in-dialog requests - and therefore, no re-INVITEs. In-dialog requests must be routed to the dialog peer that was established by the initial INVITE; you can't route them somewhere else instead once the dialog has been established. The following scenario plays out - and even then, only if Record-Route is turned on and sequential requests flow through the proxy: A Proxy B INVITE 1...@proxy --- 100 Trying --- INVITE 1...@b 100 Trying -- -- 180 Ringing --- -- 180 Ringing --- -- 200 OK w/Contact URI -- 200 OK + contact -- --- ACK cont...@b --- -- ACK cont...@b - - INVITE cont...@b --- --- INVITE cont...@b - The only way you can pull this off is to decide in advance whether the call needs to go through a special PSTN GW when routing the initial INVITE. -- 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
Re: [OpenSIPS-Users] accounting BYE
Jan,
You could set an AVP or a branch parameter. Those persist for the
lifetime of a transaction. Since subsequent BYEs are retransmissions
within the same transaction, you should be able to check if an AVP or
branch flag indicating that the BYE has already been accounted is set
before running the database command.
You could also dampen out the BYE retransmissions using t_check_trans().
Any reason you don't want to wait for 200 OK before cutting the CDR?
-- Alex
I have a problem with accounting the BYE in my mysql database (acc table).
Sometimes a BYE is sent more than one time, or sometimes the other site
does
not respond with an O.K. on the BYE.
Here is a peace of my config (example):
if(loose_route())
{
...
if(is_method(BYE))
{
#setflag(25); # account successful transactions
acc_db_request(200 OK, acc);
}
...
}
If I use the flag (in my case flag 25) and the other site does not respond
the BYE is not accounted. So I tried to use the acc_db_request. This works
also without the O.K., but is a BYE is sent twice or more I end up with a
lot of BYE's in my database.
I tried setting a flag, bflag after the acc_db_request but this did not
help.
Is there a way to account a BYE without an O.K. only one time?
Jan
--
View this message in context:
http://n2.nabble.com/accounting-BYE-tp3274605p3274605.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
___
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
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] User Authentication by IP in INVITE
There is. Try the 'permissions' module and the allow_address() allow_trusted() functions. Alberto Listas wrote: Hi, Today I test the IP in src_ip against a list in the opensips.cfg but there must be a way of doing the test against the database. Could someone please point me in the right direction? Thanks, Alberto ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] What is the role of t_check_trans at line 253 of opensips.cfg in SVN trunk
Thanks Bogdan. Appreciate your followup.
So let me put the question this way:
What is the benefit of creating a new transaction on top of the
retrans checks? Why would I not just want to wait until I call
t_relay(), which will also create a transaction if it does not already
exist. Why it would be beneficial to have it exist beforehand?It
seems that retransmission detection works the same way regardless.
--
Sent from mobile device
On Jul 16, 2009, at 11:54 AM, Bogdan-Andrei Iancu bog...@voice-system.ro
wrote:
Hi Alex,
No, t_check_trans() will NOT create a new transaction. Both function
will check (for non-ACK and non-CANCEL) if it is retransmission and
if so, it will sent (via TM) the last sent reply and stop the script
exectution. If it is not a retransmission, t_check_trans() will not
do anything else, but t_newtran() will create a new transaction.
I added this function in 1.0 (?!?) as it was mainly intended for
proper CANCEL and ACK routing.
Regards,
Bogdan
Alex Balashov wrote:
Bogdan,
Are you saying that t_check_trans() will create a new transaction
for a non-ACK/CANCEL retransmission too? Or that it retransmits
the last reply sent statelessly somehow?
--
Sent from mobile device
On Jul 14, 2009, at 9:10 AM, Bogdan-Andrei Iancu bog...@voice-system.ro
wrote:
Hi Stan,
when comes to handling retransmissions (and not CANCELs and ACKs
belonging to an INVITE transaction), both function do more or less
the
same - handle the retransmission (by retransmitting the last sent
reply)
and breaking the script execution - of course, the difference is
if no
retransmission, t_newtran() will create a new transaction for the
request.
So :
t_check_trans();
t_new_trans();
is a bit redundant. Only:
t_new_trans();
will do exactly the same job.
Again, this is true only in the context of non-CANCEL and non-ACK
requests!
Regards,
Bogdan
Stanisław Pitucha wrote:
2009/7/14 Alex Balashov abalas...@evaristesys.com:
http://www.opensips.org/html/docs/modules/1.5.x/tm.html#id272150
A bit related question. Since the docs mention:
If the processing of requests may take long time (e.g. DB lookups)
and the retransmission arrives before t_relay() is called, you
can use
the t_newtran() function to manually create a transaction.
Is there any situation where:
t_check_trans();
t_new_trans();
after all cancel / ack checks is a bad thing to do? Or maybe even:
t_check_trans();
if (is_method('INVITE|UPDATE|REFER')) t_new_trans();
since everything else can be safely duplicated / is rather light
in processing.
___
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] What is the role of t_check_trans at line 253 of opensips.cfg in SVN trunk
Stanisław Pitucha wrote: bogdan_vs: second statement is the correct oneboth first check if retransmission (and if so resend the reply). If no retransmission, only t_newtran() will force the creation of the transaction; tr_check_tran() will do nothing And what is the practical effect of this, from a request processing speed / computational overhead perspective? In other words, what is my incentive to do t_newtran()? Why don't I just wait and use t_relay() -- which creates the transaction -- at the bottom? What is useful about having a transaction created before the request forwarding is actually initiated, especially if I cannot change the request body in any way after I create the transaction manually (which I understand the documentation to be saying)? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] What is the role of t_check_trans at line 253 of opensips.cfg in SVN trunk
Marc Leurent wrote:
Hello,
What is the purpose of t_check_trans(); at line 253 in opensips.cfg
trunk version.
This function is only a check so should not be necessary here?
No?
# CANCEL processing
if (is_method(CANCEL))
{
if (t_check_trans())
t_relay();
exit;
}
* t_check_trans();*
The function is not only a check--it also has an effect on execution
behaviour for non-CANCEL / non-ACK requests.
Consider:
http://www.opensips.org/html/docs/modules/1.5.x/tm.html#id272150
[I]f the request belongs to a transaction (it's a retransmision), the
function will do a standard processing of the retransmission and ***will
break/stop the script***. The function return false if the request is
not a retransmission.
-- Alex
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] What is the role of t_check_trans at line 253 of opensips.cfg in SVN trunk
Bogdan,
Are you saying that t_check_trans() will create a new transaction for
a non-ACK/CANCEL retransmission too? Or that it retransmits the last
reply sent statelessly somehow?
--
Sent from mobile device
On Jul 14, 2009, at 9:10 AM, Bogdan-Andrei Iancu bog...@voice-
system.ro wrote:
Hi Stan,
when comes to handling retransmissions (and not CANCELs and ACKs
belonging to an INVITE transaction), both function do more or less the
same - handle the retransmission (by retransmitting the last sent
reply)
and breaking the script execution - of course, the difference is if no
retransmission, t_newtran() will create a new transaction for the
request.
So :
t_check_trans();
t_new_trans();
is a bit redundant. Only:
t_new_trans();
will do exactly the same job.
Again, this is true only in the context of non-CANCEL and non-ACK
requests!
Regards,
Bogdan
Stanisław Pitucha wrote:
2009/7/14 Alex Balashov abalas...@evaristesys.com:
http://www.opensips.org/html/docs/modules/1.5.x/tm.html#id272150
A bit related question. Since the docs mention:
If the processing of requests may take long time (e.g. DB lookups)
and the retransmission arrives before t_relay() is called, you can
use
the t_newtran() function to manually create a transaction.
Is there any situation where:
t_check_trans();
t_new_trans();
after all cancel / ack checks is a bad thing to do? Or maybe even:
t_check_trans();
if (is_method('INVITE|UPDATE|REFER')) t_new_trans();
since everything else can be safely duplicated / is rather light in
processing.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips Monitoring Current connections and Concurrent calls
Well, snmpstats *is* the answer to all of your questions. What's not clear? Hugo Serna wrote: Hi All, I would appreciate if someone let me know what tools are available out there to monitor Opensips current connections, concurrent calls and to create snmptraps (alerts) when calls are dropped below/above a min/max threshold I have found out some information using smnpstats module but not quite clear. Any help its much appreciated. Thanks in advance Alberto ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How To Ask Questions The Smart Way
Thank you for posting this. It is something that very, very often needs to be said and bears repeating. This a good read for those who show up on mailing lists without any guidance about how to ask the right questions and then complain that nobody answers their questions as they want. http://www.catb.org/~esr/faqs/smart-questions.html It was also a good read for me. Regards, Adrian ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How To Ask Questions The Smart Way
I'm on a rather nontrivial number of other mailing lists associated with various open-source projects and ecosystems, including quite a few in the VoIP space. I can tell you that what you say here is definitely not the case. li...@grounded.net wrote: Bunch of self important blowhards, this is the only mailing list that acts this way! On Fri, 10 Jul 2009 02:48:05 -0400 (EDT), Alex Balashov wrote: Thank you for posting this. It is something that very, very often needs to be said and bears repeating. This a good read for those who show up on mailing lists without any guidance about how to ask the right questions and then complain that nobody answers their questions as they want. http://www.catb.org/~esr/faqs/smart-questions.html It was also a good read for me. Regards, Adrian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Number portability
Yes, you can. Just beware that you will _have_ to use something like 302s. If you send the INVITE request back to the switch, it will be considered a call loop. -- Sent from mobile device On Jul 10, 2009, at 2:09 PM, Paul Mancheno H. pmanch...@gmail.com wrote: Hello. I have a project to do a system to implement numerical portability, the calls go out from my Softswitch and they would go directly to OpenSIPs and I look in a database (Postgresql or MySql) for the route that I must take, return a message with code 302 using a prefix depending on the route and this way my Softswtich, on having reanalyzed the number now, sends it on the other hand. Can I do that with OpenSIP? Can I have a pool of connections to the database so that one is not gaining access all the time? Perhaps is it better to use a project as Sailfin? A lot of thanks. ___ 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] Number portability
Iñaki Baz Castillo wrote: El Viernes, 10 de Julio de 2009, Alex Balashov escribió: Victor Pascual Avila wrote: On Fri, Jul 10, 2009 at 8:12 PM, Alex Balashovabalas...@evaristesys.com wrote: Yes, you can. Just beware that you will _have_ to use something like 302s. If you send the INVITE request back to the switch, it will be considered a call loop. Unless you added ;npdi or ;rn parameters to the RURI I am not sure how adding those parameters would circumvent the fundamental problem. Softswitch -- call leg 1 -- proxy -- still call leg 1 -- softswitch npdi and rp are *userinfo* parameters (in fact they are TEL URI paremeters so when converting to SIP URI they become part of the userinfo part). http://www.tech-invite.com/Ti-sip-abnf.html#teluri So, if the original RURI is: sip:+12345...@mydomain.org and OpenSIPS modifies it to: sip:+12345678;npdi=123;rn=...@mydomain.org then both RURI's are differents and the softsiwtch won't consider it a loop. However, if the parameters are added as SIP URI parameters (after the hostpart) the it would be a loop (except if they are maddr, user, ttl). How does that change the other logical attributes of a call leg, i.e. Call ID GUID, From tag, CSeq, etc? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Number portability
Brett is very right. I think one of the reasons I reacted instinctively to this scenario was because I tried to implement something similar with a well-known switch once (I think it was a Metaswitch) and the signaling agent reacted to my spiral (which I didn't know to be such) as though it were a loop. Brett Nemeroff wrote: Just throwing this out.. Not all equipment can handle SIP Spiral properly. cough asterisk cough (although I know there was work done on Asterisk+SIP Sprial, I don't know where that ended up) so be careful before you spend a lot of time on that. I'd love to hear how all of that works for you. I've got plans to do something similar in the LNP space.. -Brett On Fri, Jul 10, 2009 at 2:02 PM, Iñaki Baz Castillo i...@aliax.net mailto:i...@aliax.net wrote: El Viernes, 10 de Julio de 2009, Alex Balashov escribió: npdi and rp are *userinfo* parameters (in fact they are TEL URI paremeters so when converting to SIP URI they become part of the userinfo part). http://www.tech-invite.com/Ti-sip-abnf.html#teluri So, if the original RURI is: sip:+12345...@mydomain.org mailto:sip%3a%2b12345...@mydomain.org and OpenSIPS modifies it to: sip:+12345678;npdi=123;rn=...@mydomain.org mailto:4...@mydomain.org then both RURI's are differents and the softsiwtch won't consider it a loop. However, if the parameters are added as SIP URI parameters (after the hostpart) the it would be a loop (except if they are maddr, user, ttl). How does that change the other logical attributes of a call leg, i.e. Call ID GUID, From tag, CSeq, etc? If the RURI changes, then it's *not* a loop, but a spiral. Re-read the appropiate section in RFC 3261 :) -- Iñaki Baz Castillo i...@aliax.net mailto:i...@aliax.net ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] t_on_failure()
You need both; they do different things.
The failure_route[x] won't get triggered by default unless you associate
it with a transaction - in effect, telling OpenSIPS to trigger
failure_route[x] if a failure code is received for this transaction
after stateful relay. That's what t_on_failure() does.
route {
...
t_on_failure(1);
if(!t_relay()) {
sl_reply_error();
exit;
}
}
...
# This will never be run unless t_on_failure(1) is set
# above.
failure_route[1] {
...
}
Patrick wrote:
Is it wise to have a t_on_failure inside of a failure_route[x] ? Or
is there another method I could / should use?
Thanks,
Patrick
___
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
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] t_on_failure()
I think from a methodological perspective, you're doing just fine.
Failure_route[1] isn't going to inherently be called cyclically because
failure replies that trigger it are final replies. The only way you can
cycle through the same failure_route is if you created another branch
and armed that failure route for it, too, after the t_relay(). Both of
these have a recursion bottom; failures only happen once, unless you
manually cause a certain (branch) sequence of events to transpire beyond it.
If you saw failure_route[1] getting called twice, make sure it wasn't in
response to a CANCEL from the near-end. You need to have something like
this in there, at the beginning.
failure_route[1] {
if(t_was_cancelled()) {
log that we got a cancel, blah blah
exit;
}
}
When you get a CANCEL, first failure_route[1] is called as part of
CANCEL processing (automatically, if armed, by TM), and then, you're
going to get it called again in response to the 487 Session Terminated
message that is returned by the far end in response to the CANCEL. The
487 is part of the INVITE transaction, and since the proxy is only
transaction-stateful, that's the best it can do.
Patrick wrote:
Sorry, I should have included the code like you have to illustrate my
question (if you don't mind, I will borrow it):
route {
...
t_on_failure(1);
if(!t_relay()) {
sl_reply_error();
exit;
}
}
...
failure_route[1] {
t_on_failure(1); - here is what I am asking
about t_on_failure inside of a failure_route[x]
t_relay();
...
}
Prior to setting this, I only saw entries in failure route twice:
1) the first time the call was attempted
2) if the call failed
It would stop there even when I had a third option. Now it is trying
all three options, but just wanted to make sure this was a logical
methodology I have safe guards in place to stop it from endlessly
looping
Patrick
On Jul 9, 2009, at 6:00 PM, Alex Balashov wrote:
You need both; they do different things.
The failure_route[x] won't get triggered by default unless you associate
it with a transaction - in effect, telling OpenSIPS to trigger
failure_route[x] if a failure code is received for this transaction
after stateful relay. That's what t_on_failure() does.
route {
...
t_on_failure(1);
if(!t_relay()) {
sl_reply_error();
exit;
}
}
...
# This will never be run unless t_on_failure(1) is set
# above.
failure_route[1] {
...
}
Patrick wrote:
Is it wise to have a t_on_failure inside of a failure_route[x] ? Or
is there another method I could / should use?
Thanks,
Patrick
___
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
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Voicemail system
There's many ways to approach this. It is also possible to parallelise Asterisk apps and use shared voicemail storage so that it is not hostwise. Paul Mancheno H. wrote: Hi friends. I want to implement a voicemail system for the telecommunications company I work, I tried but it seems that Asterisk supports only 150 concurrent calls. Could it be better to use Asterisk and OpenSIPS to improve this system?, Can I use SEMS? Thanks ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Is opensips a front end to asterisk?
Specific and well-parameterised questions really are the key. -- Sent from mobile device On Jul 7, 2009, at 2:00 PM, Uwe Kastens ki...@kiste.org wrote: You are right. We all started from the same point and asked questions to learn a lot. The more specific the question is, the better the answer would match. I think your setup is not new, but it depends on your requirement and your setup. BTW: What was the initial question? :) BR Uwe li...@grounded.net schrieb: I love how joining pretty much any new mailing list and asking initial questions leads to the typical 'you should realize how difficult this is' replies. That's nothing new since there are countless folks who have aspirations without the follow through but not everyone. And really, all of you learned the same way, asking sometimes stupid, but a lot of questions, reading, playing with and getting to know, the software. Well, maybe not the developers :). Anyhow, I'd still love to see some feedback on my original question. Mike ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ 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] limit number of outbound calls using DIALOG
Use a dialog profile. Use $fu as the key (value). Check if profile size is 1, refuse the call. That's what they're there for. Jayesh Nambiar wrote: Hi, I am looking at an option to limit more than one call per user even if they are registered from multiple locations. Basically if User A calls from location A and if the call is active, User A registered from location B should not be allowed to make a call. What I did was: 1) Create a dialog after every initial INVITE initiated by users 2) Before creating the dialog, query the dialog table to check if $fu has an entry in the dialog table using avp_db_query. 3) If yes, means user A is already on a call so send a 403, Forbidden. 4) Else, create the dialog and process call. Although this works, i just wonder if doing avp_db_query everytime to check if the caller has a call active is an efficient way of doing it?? Is it possible to store these dialog parameters in localcache using memcache module or access the dialog parameters from memory and compare it with the INVITE messages !!! Just trying to find a more efficient way of achieving this. Thanks for any inputs you might have !! --- Jay ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ACK bug?
Can you paste your OpenSIPS config? It may be that the ACK is not being properly routed in all circumstances. Charles Solar wrote: I am experiencing an ack bug in opensips I believe. I have a caller register to a server, call it 231, and I have 231 send invites to 228 which processes the route and does lcr. 228 sends calls to the best gateway, which in my tests is just one asterisk server (also on 228, port 5059). I have 231 and asterisk record their route, 228 does not show up in the route header. The problem comes in when asterisk sets up a call it tries to bridge the caller and callee with reinvites. I see the 200 OK message and my caller sends a ACK back, but opensips does not forward the ACK properly. This is a wireshack graph of the conversation from 231's perspective http://img197.imageshack.us/img197/7889/sshot2mfv.png I have tried shifting through the debug messages in syslog but all I can tell is that 231 is trying to forward the ACK to itself. Has anyone else experienced this problem or know whats going on? Thank you for your time ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] string transformation with avps formating
Brett Nemeroff wrote:
Hey All,
I can't seem to get the format right here:
$rU = $(rU{s.substr,$avp(s:nprefixlen),0});
This is possible with $var(...)s. Not sure about AVPs. I use it, it
works for me.
Now, what doesn't work is using nested transformations or arithmetical
operations on numerical transformation values. They won't be evaluated
properly. Have to assign them to an outside variable first. For
instance, can't do something like:
$(fU{s.substr,fU{s.len} - 10,10})
--
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
Re: [OpenSIPS-Users] NAT and media/signaling IPs different
I did not mean to imply it was only useful in large-scale architecture. Good point. Uwe Kastens wrote: Hi, To use different IPs for signaling and media gives some option not only for big installations: - give a customer the media gw which has the best ip connection (based on src.ip and geographic location), - scale with dump server instead of sbcs, BR Uwe Alex Balashov schrieb: The topology you describe is an alternative, if you've got the capital to blow on SBCs. Jeff Pyle wrote: Alex, That makes sense, but for NAT? Vonage, for example. Signaling and media are the same last time I looked. Since the provider has immediate control of where the client registers, scaling is available by adding more SBCs and controlling which users hit which SBCs. - Jeff On 6/8/09 8:29 PM, Alex Balashov abalas...@evaristesys.com wrote: It is absolutely indispensable to separate signaling and media for large-scale service delivery platforms. Think about traditional switch architecture (signaling agent - media gateway farm). Jeff Pyle wrote: Alex Iñaki, Thanks for the info. I knew in a non-NAT scenario this was the case; I had never seen it done separately in a NAT scenario. That's good news. - Jeff On 6/8/09 8:22 PM, Alex Balashov abalas...@evaristesys.com wrote: No, it is not necessary. The signaling and the bearer plane can be separate entirely. And on 6/8/09 8:16 PM, Iñaki Baz Castillo i...@aliax.net wrote: Not at all. -- 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
Re: [OpenSIPS-Users] [Kamailio-Users] Maintenance of Modules
Iñaki Baz Castillo wrote: The modules are compatible between plataforms ? No. They can be easily ported since both projects come from OpenSIPS not so much time ago. For the time being. That may change in the future (possibly even the near future) as the two projects inevitably diverge somewhat, and assuming their proprietors do not see a mutual interest in module compatibility. -- 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
Re: [OpenSIPS-Users] NAT and media/signaling IPs different
Jeff Pyle wrote: Hello, In my experience with SIP thus far I've been rather insulated from the ill effects of NAT on SIP and RTP. My honeymoon is over. In every NAT-supporting commercial SBC I've seen the signaling IP is the same as the media IP. Is this necessary? In Opensips/Mediaproxy terms, does Opensips need to be operating on the same IP address as the media relay? No, it is not necessary. The signaling and the bearer plane can be separate entirely. -- 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
Re: [OpenSIPS-Users] NAT and media/signaling IPs different
The topology you describe is an alternative, if you've got the capital to blow on SBCs. Jeff Pyle wrote: Alex, That makes sense, but for NAT? Vonage, for example. Signaling and media are the same last time I looked. Since the provider has immediate control of where the client registers, scaling is available by adding more SBCs and controlling which users hit which SBCs. - Jeff On 6/8/09 8:29 PM, Alex Balashov abalas...@evaristesys.com wrote: It is absolutely indispensable to separate signaling and media for large-scale service delivery platforms. Think about traditional switch architecture (signaling agent - media gateway farm). Jeff Pyle wrote: Alex Iñaki, Thanks for the info. I knew in a non-NAT scenario this was the case; I had never seen it done separately in a NAT scenario. That's good news. - Jeff On 6/8/09 8:22 PM, Alex Balashov abalas...@evaristesys.com wrote: No, it is not necessary. The signaling and the bearer plane can be separate entirely. And on 6/8/09 8:16 PM, Iñaki Baz Castillo i...@aliax.net wrote: Not at all. -- 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
[OpenSIPS-Users] Database and other high-level functionality (was: Re: Sqlops in opensips ?)
Bogdan, Bogdan-Andrei Iancu wrote: When comes to none-SIP related stuff, there is a logical limit of what should be in or not - after all we do a SIP server and not a DB wrapper, neither a XMLRPC server, not an advanced language interpreter (and so one). I can certainly appreciate that. In principle, I agree that at the end of the day, underneath it all, OpenSIPS is a SIP proxy, along with some lightweight UAS features (registrar, presence user agent, etc.). It's still much more a piece of service delivery *infrastructure* - it is low-level relative to some other network elements that frequently make their appearance in the open-source VoIP ecosystem. No argument there. But I think this does have to be balanced with the reality that a great deal of the OpenSER technology stack's usefulness does come from the fact that it can be deployed in application-aware configurations, the extensibility of the route script, and so on. This ability to add intelligence and integration paths is precisely where it has an edge over the expensive commercial proxies and, in certain situations, SBCs. To give you an example of how I see this delimitation of what should be in OpenSIPS and what not. Let's take the DB case (anyhow the discussion started from there): the current DB support in script is a very decent one (via the avpsops functions) - you can do mostly all types of queries and DB interaction. If you need something more complex (from DB), I think you must work on the DB side and do the enhancements there. Make no sense to invest effort in doing super DB stuff in a SIP server, when the DB engine itself may already offer this support. Sure, but taking advantage of some of the more sophisticated DB capabilities on the back side also requires adequate interfaces on the front side. For example, how does one deal efficiently and easily with multiple rows returned by a DB query? At present the only way is to iterate through AVP arrays in a rather obfuscated manner that is hard to understand and not particularly terse. All this benefits from improvements to language syntax and semantics as well as the DB layer. Again, an example - couple of weeks ago I had to interface OpenSIPS with some really complex data structures in a postgres DB - the solution was simple - some postgres procedures were created to hide the DB complexity and also to incorporate some logic. Form OpenSIPS point of view, the problem was reduced to running a select over the procedures. I completely agree. In fact, this is exactly how I have been doing my work (I almost exclusively use Postgres) for years precisely because of the fact that the route script does not have the capabilities of a general-purpose programming languages, including native support for nontrivial primitives and other semantics that are needed to do that kind of logic programmatically. I rely very heavily on stored procedures and triggers. Nevertheless, the interface could use some enhancements to make this coupling easier. It's just little things. Like, for example, right now, if you issue a DB query that returns no rows, you need to use is_avp_set() to check whether the corresponding AVP(s) are set. So, the effort is better focused on SIP part rather is peripheral interaction, where we can use the already existing and specific tools and mechanism. Nevertheless, a lot of the peripheral tools hold the key to a great deal of the value. I invite you to remember why OpenSER grew to such popularity after the fork with SER in 2005: it is the mass of community-contributed modules and novel functionality. SER may have focused on a good core, but OpenSER could be said to have won precisely because of the additional baked-in capabilities. Whatever your opinion of bells whistles modules, I think it is very important to preserve the inherent benefit offered by most open-source software compared to proprietary alternatives: the integration paths. Asterisk for example has AGI and the Manager interface, which both allow outside processes and outboard logic controllers to touch and manipulate the engine. MI[_DATAGRAM], XML-RPC, and a relatively flexible route script are all very important for that reason. The capability to integrate is paramount above all else, and is a governing factor in the technology choice. About the place of the DB related stuff - well, originally they were operating only with AVP and this is why they were put in the avpops. Now, indeed, there is no dependency for this, but the questions is what will be the advantage for a users of moving some functionality in a separate module ? I fail to see any... Expanded database-centric functionality, especially free from the constraints of unrelated AVP constructs, such as the need to define an avp_table as a modparam even if one is not going to use it. -- Alex -- Alex Balashov Evariste Systems Web: http
Re: [OpenSIPS-Users] Sqlops in opensips ?
Admittedly, OpenSER 1.3.x was the last time I tried the Perl module, but my performance results with the Perl module have been very bad, and there are memory leaks in it as well. I think much of it has to do with the fact that the maintainer has been effectively unavailable for several years. Not sure if anything has happened to the Perl module in OpenSIPS since then. Alex Massover wrote: I think perl module is most practical for you. -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. *From:* users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Sharath *Sent:* Wednesday, May 27, 2009 12:42 AM *To:* users@lists.opensips.org *Subject:* [OpenSIPS-Users] Sqlops in opensips ? hello, Is there any module equivalent to sqlops of openser in opensips ? Basically I want to run sql queries from proprietary tables and use them in the route script file. thank you -Sharath This mail was received via Mail-SeCure System. This mail was sent via Mail-SeCure System. ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Sqlops in opensips ?
60,000 times a day is about 41 times a minute (or, a little less than one operation per second), assuming uniform distribution. That probably won't stress it too badly. My issues were with bigger loads. Jeff Pyle wrote: For what it's worth, I use the perl module to execute some custom database operations for custom route decision making. It runs about 60k times per day in a Xen VM with no memory or performance issues. I've been quite pleased. - Jeff On 5/28/09 8:46 AM, Alex Balashov abalas...@evaristesys.com wrote: Admittedly, OpenSER 1.3.x was the last time I tried the Perl module, but my performance results with the Perl module have been very bad, and there are memory leaks in it as well. I think much of it has to do with the fact that the maintainer has been effectively unavailable for several years. Not sure if anything has happened to the Perl module in OpenSIPS since then. Alex Massover wrote: I think perl module is most practical for you. -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. *From:* users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Sharath *Sent:* Wednesday, May 27, 2009 12:42 AM *To:* users@lists.opensips.org *Subject:* [OpenSIPS-Users] Sqlops in opensips ? hello, Is there any module equivalent to sqlops of openser in opensips ? Basically I want to run sql queries from proprietary tables and use them in the route script file. thank you -Sharath This mail was received via Mail-SeCure System. This mail was sent via Mail-SeCure System. ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Sqlops in opensips ?
Oh, heaven knows I don't use the perl approach anymore. I was just remembering. Jeff Pyle wrote: It’s during business hours. We sipp-tested it at 40 cps without an issue. I probably should have mentioned that. In our case we have to evaluate the costs returned by some stored procedures, mix in some spices, fluff the egg whites, flash fry and return a route list. Brett’s right. In your application, gflags sound like the hot ticket. - Jeff On 5/28/09 10:23 AM, Brett Nemeroff br...@nemeroff.com wrote: gflags + avp_db_load should be able to do much much more than this without a problem.. even gflags + the perl module would be better On Thu, May 28, 2009 at 9:18 AM, Alex Balashov abalas...@evaristesys.com wrote: 60,000 times a day is about 41 times a minute (or, a little less than one operation per second), assuming uniform distribution. That probably won't stress it too badly. My issues were with bigger loads. Jeff Pyle wrote: For what it's worth, I use the perl module to execute some custom database operations for custom route decision making. It runs about 60k times per day in a Xen VM with no memory or performance issues. I've been quite pleased. - Jeff On 5/28/09 8:46 AM, Alex Balashov abalas...@evaristesys.com wrote: Admittedly, OpenSER 1.3.x was the last time I tried the Perl module, but my performance results with the Perl module have been very bad, and there are memory leaks in it as well. I think much of it has to do with the fact that the maintainer has been effectively unavailable for several years. Not sure if anything has happened to the Perl module in OpenSIPS since then. Alex Massover wrote: I think perl module is most practical for you. -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. *From:* users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Sharath *Sent:* Wednesday, May 27, 2009 12:42 AM *To:* users@lists.opensips.org *Subject:* [OpenSIPS-Users] Sqlops in opensips ? hello, Is there any module equivalent to sqlops of openser in opensips ? Basically I want to run sql queries from proprietary tables and use them in the route script file. thank you -Sharath This mail was received via Mail-SeCure System. This mail was sent via Mail-SeCure System. ___ 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 ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Sqlops in opensips ?
No generic database operations module. But you can use avp_db_query() from avpops, which is the traditional way to go for this problem. Sharath wrote: hello, Is there any module equivalent to sqlops of openser in opensips ? Basically I want to run sql queries from proprietary tables and use them in the route script file. thank you -Sharath ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Sqlops in opensips ?
Brett Nemeroff wrote: The original intent was to be a fast, scalable SIP router. Having runtime queries against your database didn't fit into that model. Not only that there were no variables. So there was no way to manipulate or otherwise really use the resultant data. Sure. But there is a limit to what can be done to meet application-specific needs within the box of the existing modules provided. Along with the UAS features that co-evolved onto the proxy layer, the increasing generalisation of the pseudoprogrammatic route script environment is a logical direction. I agree that this stuffed into the AVP module seems odd, but given the AVP module gives the scripting language it's variable capabilities, it makes sense. I wouldn't dispute that; on the one hand, it is an odd place to put the database interaction functionality, but on the other hand, it is probably the most conceptually self-evident place of the existing module library. I think the ideal answer is C, though - none of the above, make a special module for it. Before AVPs, you did routing based on module logic and there wasn't anyway to customize it without writing your own modules by hand. As much odd as the avpops module integrates arbitrary database interactions, I'm not sure how I'd change it rather than a typical kind of prepare / execute/ fetch kind of loop. But that isn't an efficient design for a real-time switch. I rather like how it is today. It does pose a formidable design challenge; there's not a lot of usefulness in asynchronous database calls because it's no good - the response from the database is still needed to carry on processing a request, and that can only happen if the process blocks on database response. What I think is in dire need of more asynchronous-minded renovation is the fact that database calls can block an entire worker process. Since there are no threads used (that is to say, POSIX threads), a spuriously latent database operation will block a whole child process. Child processes handle many requests concurrently in a high-volume scenario. So, that needs to change. -- 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
Re: [OpenSIPS-Users] Integration with Asterisk/Trixbox
Iñaki Baz Castillo wrote: For sure :) Unfortunatelly it seems that people integrating OpenSIPS with Asterisk always comes to OpenSIPS maillist to ask question, in fact, about Asterisk :( There's always the SER-Asterisk-Interwork list: http://lists.evaristesys.com/mailman/listinfo/ser-asterisk-interwork -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CANCEL with a To tag.
It sounds like the CANCEL with the To-tag should have a Route header as
well in order for it to be processed like any other sequential/in-dialog
request -- that is to say, under loose_route().
Or, the CANCEL is intended for OpenSIPS itself, in which case it should
not have a To-tag.
I would not try to accommodate this broken UA if I were you. When
breakage is so fundamental, this way lies madness.
Chris Maciejewski wrote:
You can see a SIP flow before I added CANCEL to a lose routing
section of my Opensips config here:
http://wima.co.uk/sip/2009-05-11_10-18-39-test-call_index.html
Note: F23 is rejected by OpenSIPs as it got tag in a To: header.
And after I added:
if (is_method(CANCEL))
{
t_relay();
exit;
}
to my lose routing logic, OpenSIPs generates CANCEL and sends it to
the next hop:
http://wima.co.uk/sip/2009-05-11_10-46-46-test-call_index.html
2009/5/11 Iñaki Baz Castillo i...@aliax.net:
2009/5/11 Chris Maciejewski ch...@wima.co.uk:
Hi,
I would like to ask what would be the best way to handle CANCEL
request with a To tag. I know such a CANCEL request is not RFC
compatible
CANCEL is hop-by-hop. This means that when OpenSIPS receives a CANCEL,
it *doesn't* route it, but it generates a new one (this occurs when
you do t_relay() for a CANCEL).
It's impossible to add To tag to a CANCEL generated by OpenSIPS
(expect if the CANCEL occurs for a re-INVITE being into an already
established dialog, so arriving CANCEL has To tag and OpenSIPS routes
it as any other in-dialog request).
but unfortunately I came across some buggy UAs doing this.
What do you mean with it? what does this UAS?
--
Iñaki Baz Castillo
i...@aliax.net
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CANCEL with a To tag.
Iñaki Baz Castillo wrote: 2009/5/11 Alex Balashov abalas...@evaristesys.com: It sounds like the CANCEL with the To-tag should have a Route header as well in order for it to be processed like any other sequential/in-dialog request -- that is to say, under loose_route(). But it would be incorrect anyway. A CANCEL for an initial-INVITE shouldn't have To tag since the CANCEL must end the whole UAC transaction, not just an early-dialog. Agreed, but I think the more harmless approach would be for the To tag issue to be ignored by the proxy and passed to the receiving UA to deal with. Or, the CANCEL is intended for OpenSIPS itself, in which case it should not have a To-tag. The CANCEL is always for OpenSIPS since CANCEL is hop by hop. Well, true. I meant a stateless vs. stateful CANCEL -- which also changes the domain destination of the RURI. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CANCEL with a To tag.
Alex Balashov wrote: Iñaki Baz Castillo wrote: 2009/5/11 Alex Balashov abalas...@evaristesys.com: It sounds like the CANCEL with the To-tag should have a Route header as well in order for it to be processed like any other sequential/in-dialog request -- that is to say, under loose_route(). But it would be incorrect anyway. A CANCEL for an initial-INVITE shouldn't have To tag since the CANCEL must end the whole UAC transaction, not just an early-dialog. Agreed, but I think the more harmless approach would be for the To tag issue to be ignored by the proxy and passed to the receiving UA to deal with. Although, since the has_totag() check is done first and loose_route() second in stock configs from which people derive theirs, I guess that really wouldn't work... -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips and asterisk retransmits
You may wish to consider posting this to the SER-Asterisk-Interwork list. troxlinux wrote: Hi list , I have some days fighting with asterisk and opensips to solve this problem, when I use asterisk to listen my voicemail and to call to the pstn, asterisk shows me this error message: WARNING[3196]: chan_sip.c:1976 retrans_pkt: Maximum retries exceeded on transmission d5a57aa528f5c...@192.168.10.30 for seqno 45371 (Critical Response) -- See doc/sip-retransmit.txt. [Apr 28 19:34:44] WARNING[3196]: chan_sip.c:1998 retrans_pkt: Hanging up call d5a57aa528f5c...@192.168.10.30 - no reply to our critical packet (see doc/sip-retransmit.txt). I read the documentation in asterisk, and there are possibly several factors for those that I could give this problem: Firewall - (I Don`t have) A badly configured SIP proxy - ( with the version 1.3.4 of openser I work me well and I never had this problem ) A SIP middlebox (SBC) - (I Don`t have) I use opensips with asteriks in the same server but in different port, and I have asterisk set in mode comedia any idea? some person that has presented him previously this problem? help!... -- 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
Re: [OpenSIPS-Users] t_release() not found - missing loadmodule?
It was removed in 1.5.0. Transactions are now automatically released where appropriate. Franz Edler wrote: Hi all, I am a little bit confused now. May someone can help me. I use opensips 1.5.1 and get an error in my opensips.cfg when referring to t_release() function. Below are the relevant parts of the console output during loading. --- snip Apr 27 06:40:40 [2847] DBG:core:yyparse: loading module /usr/local/lib/opensips/modules/tm.so Apr 27 06:40:40 [2847] DBG:core:register_module: register_pv: tm Apr 27 06:40:40 [2847] DBG:core:pv_add_extra: extra items list is not initialized --- snip - and some lines later: --- snip Apr 27 06:40:40 [2847] DBG:core:find_cmd_export_t: t_release not found Apr 27 06:40:40 [2847] DBG:core:find_cmd_export_t: t_release not found Apr 27 06:40:40 [2847] CRITICAL:core:yyerror: parse error in config file, line 123, column 31-32: unknown command, missing loadmodule? --- snip Shouldn't t_release() be exported by tm.so? Regards Franz ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [asterisk-users] Asterisk is not designed for University with largeuser base?
This isn't really the right mailing list for that question. The answer, though, is, as always: it depends. Yehavi Bourvine wrote: Hello, After a long time we had a meeting with our university's management and got a green light to have a proof of concept with open source telephony. Now I have to select the right software to experiment with... Up to now I thought of going with OpenSER for the masses and Asterisk for voicemail and other media related things. However, from reading around it seems like FreeSwitch can give me the benefits of both packages. Anyone has an experience with it? Thanks, __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensip and asterisk
Brett Nemeroff wrote: Both OpenSIPs and Asterisk are telephony toolkits and both provide similar features (some better than others). So you're task is to figure out what you want to do on which box. I would have to disagree; there is virtually zero imaginable correlation (that I can see) between what Asterisk provides - or is designed for - and what OpenSIPS does. They seem to be most emphatically dissimilar. -- 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
Re: [OpenSIPS-Users] Opensip and asterisk
Those are relatively superficial applications belonging to a narrow class. What is more instructive here, I think, is the formal difference; OpenSIPS is a proxy, which is necessarily a lightweight and relatively transparent network element designed to facilitate *SIP* request and reply *routing*. Asterisk is designed to be an *endpoint* of a SIP call and has an event loop replete with all sorts of application-level features, and is also a B2BUA. For all practical purposes, OpenSIPS is a great, great deal more low-level than Asterisk in terms of the functionality it exposes and the roles for which it is intended. Brett Nemeroff wrote: Both can act as a registrar, both can route calls. You may not like the way asterisk does it (I certainly don't). But they both can do it. Yes, you can setup phones to register to asterisk and opensips to provide LCR. Alternatively, you can have opensips as a registrar and asterisk do the lcr. Yeah, asterisk doing LCR would be nuts, but it can do it. I certainly wouldn't recommend it. But the point is, deciding which platform you want to do what. And as far as what asterisk is designed for. That's entirely a matter of opinion. I personally think it's designed for a low grade pbx. While others will argue that they distribute thousands of calls with it (in fact compare it to opensips even!). I see several places of overlap, and like I said, each product has it's own strenghs. It's simply a matter of opinion. On Wed, Mar 25, 2009 at 8:33 AM, Alex Balashov abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote: Brett Nemeroff wrote: Both OpenSIPs and Asterisk are telephony toolkits and both provide similar features (some better than others). So you're task is to figure out what you want to do on which box. I would have to disagree; there is virtually zero imaginable correlation (that I can see) between what Asterisk provides - or is designed for - and what OpenSIPS does. They seem to be most emphatically dissimilar. -- 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
Re: [OpenSIPS-Users] Rtp proxy issue
It means you are applying the NAT UAC test function for SDP to a request that does not have an SDP payload. It should only be applied to messages that contain SDP payloads. Easy way to check: if(search(Content-Type: application/sdp)) Also, only the following kinds of messages can contain SDP descriptors: 1) Initial INVITEs; 2) Sequential INVITEs; 3) 200 OKs to INVITE transactions; 4) Non-100 1xx provisional messages -- these are usually 183 Session in Progress and 180 Ringing messages. However, technically, any non-100 1xx message can contain an SDP body per the RFC. In practise, this is rare, so t_check_status(200|183|180) will work for most scenarios. But if you want to be strictly correct, do: if((t_check_status(200|183|180) search(Content_Type: application/sdp)) || search(Content-Type: application/sdp)) michel freiha wrote: Hi all, I'm getting the below error when trying to make a call through OpenSIPS DBG:core:parse_headers: flags= Mar 6 20:43:29 [7117] ERROR:nathelper:extract_body: message body has length zero Mar 6 20:43:29 [7117] ERROR:nathelper:force_rtp_proxy2_f: can't extract body from the message Can you explain please how this is affecting the call specially that the call is working fine Regards ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSip SIP, SIP-I e SIP-T
Not possible, Asterisk doesn't understand SIP-T. What do you mean by that, anyway? Only a device that supports ISUP interworking will support SIP-T. Secondly, as is often repeated here, SIP-T is a specification for a _PAYLOAD_ - an extension - of SIP. It is not a different protocol. Daviramos Roussenq Fortunato wrote: Hi List. I have a trunk SIP-T must deliver it to the Asterisk in SIP. The Opens is that? How should be my opensip.cfg? ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSip SIP, SIP-I e SIP-T
SIP-T *is* SIP. Anything that processes SIP can interpret the SIP part. What it can't do is interpret the SIP-T part, so it's passed through, ignored, or stripped, depending on what type of SIP agent it is. If it's a proxy (like OpenSIPS), it's conservative and passes everything it receives in terms of message bodies and additional parameters. If it's a B2BUA, who knows. Daviramos Roussenq Fortunato wrote: Hi Alex. If a different protocol is not to say that you can connect directly to Asterisk and will work, just taking the resources of ISUP? SIP-T is not talking with SIP protocol are different, after all SIP-T carries information that the SIP can not interpret. The Problem is the following I get a SIP-T trunk and Asterisk to deliver precise, how best to do. 2009/2/19 Alex Balashov abalas...@evaristesys.com mailto:abalas...@evaristesys.com Not possible, Asterisk doesn't understand SIP-T. What do you mean by that, anyway? Only a device that supports ISUP interworking will support SIP-T. Secondly, as is often repeated here, SIP-T is a specification for a _PAYLOAD_ - an extension - of SIP. It is not a different protocol. Daviramos Roussenq Fortunato wrote: Hi List. I have a trunk SIP-T must deliver it to the Asterisk in SIP. The Opens is that? How should be my opensip.cfg? ___ Users mailing list Users@lists.opensips.org mailto: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 ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSip SIP, SIP-I e SIP-T
That is accurate. Brett Nemeroff wrote: From what I understand about SIP-T it's SIP + ISUP params in the message. The required bits such as RURI and SDP all work as expected. Group, feel free to correct me there. Depending on your specific setup and network architecture, it should work. However, you may not be able to do anything with the ISUP components of the messaging. -Brett On Thu, Feb 19, 2009 at 11:14 AM, Daviramos Roussenq Fortunato daviramo...@gmail.com wrote: Hi Alex. If a different protocol is not to say that you can connect directly to Asterisk and will work, just taking the resources of ISUP? SIP-T is not talking with SIP protocol are different, after all SIP-T carries information that the SIP can not interpret. The Problem is the following I get a SIP-T trunk and Asterisk to deliver precise, how best to do. 2009/2/19 Alex Balashov abalas...@evaristesys.com Not possible, Asterisk doesn't understand SIP-T. What do you mean by that, anyway? Only a device that supports ISUP interworking will support SIP-T. Secondly, as is often repeated here, SIP-T is a specification for a _PAYLOAD_ - an extension - of SIP. It is not a different protocol. Daviramos Roussenq Fortunato wrote: Hi List. I have a trunk SIP-T must deliver it to the Asterisk in SIP. The Opens is that? How should be my opensip.cfg? ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov 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
Re: [OpenSIPS-Users] OpenSip SIP, SIP-I e SIP-T
Any standard ISUP attribute has a corresponding map into SIP-T. So, yes, any bearer-related information is going to be in there as well. Brett Nemeroff wrote: One question that I'm not sure of.. Are there any extensions in epru SIP-T that specify remote interface to use, such as used in H.248/Megaco? ie: dial 123 on TCIC 10012 -Brett On Thu, Feb 19, 2009 at 11:27 AM, Alex Balashov abalas...@evaristesys.com wrote: That is accurate. Brett Nemeroff wrote: From what I understand about SIP-T it's SIP + ISUP params in the message. The required bits such as RURI and SDP all work as expected. Group, feel free to correct me there. Depending on your specific setup and network architecture, it should work. However, you may not be able to do anything with the ISUP components of the messaging. -Brett On Thu, Feb 19, 2009 at 11:14 AM, Daviramos Roussenq Fortunato daviramo...@gmail.com wrote: Hi Alex. If a different protocol is not to say that you can connect directly to Asterisk and will work, just taking the resources of ISUP? SIP-T is not talking with SIP protocol are different, after all SIP-T carries information that the SIP can not interpret. The Problem is the following I get a SIP-T trunk and Asterisk to deliver precise, how best to do. 2009/2/19 Alex Balashov abalas...@evaristesys.com Not possible, Asterisk doesn't understand SIP-T. What do you mean by that, anyway? Only a device that supports ISUP interworking will support SIP-T. Secondly, as is often repeated here, SIP-T is a specification for a _PAYLOAD_ - an extension - of SIP. It is not a different protocol. Daviramos Roussenq Fortunato wrote: Hi List. I have a trunk SIP-T must deliver it to the Asterisk in SIP. The Opens is that? How should be my opensip.cfg? ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov 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
Re: [OpenSIPS-Users] OpenSip SIP, SIP-I e SIP-T
See: http://tools.ietf.org/html/draft-jfp-sip-isup-header-00 Grep for CIC / cic. Alex Balashov wrote: Any standard ISUP attribute has a corresponding map into SIP-T. So, yes, any bearer-related information is going to be in there as well. Brett Nemeroff wrote: One question that I'm not sure of.. Are there any extensions in epru SIP-T that specify remote interface to use, such as used in H.248/Megaco? ie: dial 123 on TCIC 10012 -Brett On Thu, Feb 19, 2009 at 11:27 AM, Alex Balashov abalas...@evaristesys.com wrote: That is accurate. Brett Nemeroff wrote: From what I understand about SIP-T it's SIP + ISUP params in the message. The required bits such as RURI and SDP all work as expected. Group, feel free to correct me there. Depending on your specific setup and network architecture, it should work. However, you may not be able to do anything with the ISUP components of the messaging. -Brett On Thu, Feb 19, 2009 at 11:14 AM, Daviramos Roussenq Fortunato daviramo...@gmail.com wrote: Hi Alex. If a different protocol is not to say that you can connect directly to Asterisk and will work, just taking the resources of ISUP? SIP-T is not talking with SIP protocol are different, after all SIP-T carries information that the SIP can not interpret. The Problem is the following I get a SIP-T trunk and Asterisk to deliver precise, how best to do. 2009/2/19 Alex Balashov abalas...@evaristesys.com Not possible, Asterisk doesn't understand SIP-T. What do you mean by that, anyway? Only a device that supports ISUP interworking will support SIP-T. Secondly, as is often repeated here, SIP-T is a specification for a _PAYLOAD_ - an extension - of SIP. It is not a different protocol. Daviramos Roussenq Fortunato wrote: Hi List. I have a trunk SIP-T must deliver it to the Asterisk in SIP. The Opens is that? How should be my opensip.cfg? ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov 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
Re: [OpenSIPS-Users] OpenSip SIP, SIP-I e SIP-T
To expand on this just a little bit: While here in the VoIP cottage industry we mostly deal with SIP to begin with, in that we use ISDN gateways for connecting to carriers, get SIP trunking from our carriers/ITSPs, and so on, the reality is that most stuff in the PSTN carrier space is still done with big-iron TDM equipment, at least here in the US. If you want to be a competitive carrier, you *must* interconnect with the incumbent telco using SS7; no ands, buts, ors. That doesn't mean there aren't a lot of opportunities to deploy SIP internally inside the service delivery core. The main benefit SIP provides there is that it is so high-level and easy to manipulate. As a result, a lot of mediation, logging, billing, analysis, translation, LCR can be done easily and inexpensively. Before SIP and H.323 came along, doing this kind of stuff required a box that did all that and spoke SS7 or, at the very least ISDN Q.931, and that is much more expensive, inflexible, and difficult to manipulate. Promoting this traffic to a higher-level protocol stack that has more applications and tools to deal with it allows the development of solutions for doing sophisticated telco-world stuff using commodity hardware and open methodologies, open-source style. That has triggered a wave of new products and paradigms in the telco space in a way that is analogous to how Asterisk et al have revolutionised the PBX space. One example of this is TransNexus' NexOSS/NexSRS product (www.transnexus.com). They use the OSP (Open Settlement Protocol) module for OpenSER and/or for Asterisk (depending on whether a B2BUA is required) internally inside their product to perform a lot of neat AAA and routing functions (e.g. the NexSRS route server). Their ability to do this benefits precisely from the fact that the traffic can be moved onto a higher-level protocol plane and away from proprietary, expensive, closed and inflexible stuff that has been a defining feature of the telco world. If you can turn the traffic into SIP or H.323, they can deal with it, but if it's SS7 or PRI, they can't. The world is going more soft[ware]. At the same time, the telco space is not a SIP world right now; the network edges are still SS7, and the market really hasn't settled on a good private SIP interconnection/peering strategy and implementation for intercarrier settlement. So, for the most part SIP trunking is used for customer access only. The SS7 information must be conserved in this type of setup, and that's one of the reasons the sort of thing that SIP-T is exists. Alex Balashov wrote: Adrian Georgescu wrote: Why should SIP-T still exist? Is it cheaper than having a gateway? What is the practical use case for investing in such technology? I am eager to learn We've used it extensively in work with CLECs that operate TDM switches such as the Metaswitch, Lucent LCS/Telica, etc. When a carrier operates more than one switch, SS7 interconnection between them is generally required so, for the same basic reasons an internal iBGP mesh or partial mesh (confederation) between two border routers is required for IP. One switch must be aware of numbers routed or ported into the other switch, and so on. The reason for its existence is that if both network elements support SIP-T, it allows you to replace an SS7 IMT (inter-machine trunk) with an IP-based mechanism for this interconnection. This allows you to move the traffic over a data network and get all the benefits that this brings; economies of scale through decreased facilities, oversubscription, etc. The main benefit is the elimination of TDM trunk exhaust; SS7 IMTs are physically bundles (trunk groups/TCICs) of DS0s, usually consisting of one or more T1s, and sometimes DS3s or more. That means that when a large volume of calls is running between the two switches, you could burn up all your SS7 trunks. Running the calls as SIP-T allows you to use something like a gigabit network core to make that problem go away somewhat -- a key benefit of VoIP in most other scenarios with which you are familiar with. At the same time, the switches still need ISUP attributes carried in SS7 IAMs and ACMs for billing, because that's just the information they operate on internally. SIP-T provides an IP-based way to encapsulate that information. SIGTRAN (essentially, SS7-over-IP) is another way to do this. However, SIP-T is lightweight and easier to deploy. It also allows you to use existing SIP network elements (proxies, session border controllers, etc.) to route and manage the traffic. For example, if you were using OpenSIPS + ACC + FreeRADIUS as a CDR catcher, you could run the SS7 calls between two switches and log the appropriate information as custom attributes. There are no good open-source implementations for SIGTRAN - nothing as turn-key as Kamailio or OpenSIPS. SIP is high-level
Re: [OpenSIPS-Users] OpenSip SIP, SIP-I e SIP-T
Oh, I don't. :) I didn't take it that way. I have no personal investment in it whatsoever. Just trying to help identify the relevance. Adrian Georgescu wrote: So far SIP-T occurred sporadically on this mailing list. I simply try identify the relevance of it in this context do not take personally mu comments. On Feb 19, 2009, at 10:39 PM, Alex Balashov wrote: The problem is that outside of the VoIP cottage industry, this stuff isn't legacy by any stretch of the imagination, in any way, shape, or form. We're just used to fancifully imagining that it is. Adrian Georgescu wrote: Hm, It is very hard to judge the benefits of performing all the nice to have feature at a higher level protocol while still having to support legacy expensive infrastructure underneath. Now, last time I heard about SIP-T was by an ECMA standard a few years ago. ECMA is a sort of inverse pyramid European standards body that nobody listens to. Basically, they are sponsored by vendors to endorse 'standards' because they posses an EU stamp. The word here in Europe goes that if something went to the extent of geting an ECMA official endorsement, one knows that it is a standard with no future and no company invests in it anymore. Maybe I am wrong and this has much more sense in the US. Adrian On Feb 19, 2009, at 8:43 PM, Alex Balashov wrote: To expand on this just a little bit: While here in the VoIP cottage industry we mostly deal with SIP to begin with, in that we use ISDN gateways for connecting to carriers, get SIP trunking from our carriers/ITSPs, and so on, the reality is that most stuff in the PSTN carrier space is still done with big-iron TDM equipment, at least here in the US. If you want to be a competitive carrier, you *must* interconnect with the incumbent telco using SS7; no ands, buts, ors. That doesn't mean there aren't a lot of opportunities to deploy SIP internally inside the service delivery core. The main benefit SIP provides there is that it is so high-level and easy to manipulate. As a result, a lot of mediation, logging, billing, analysis, translation, LCR can be done easily and inexpensively. Before SIP and H.323 came along, doing this kind of stuff required a box that did all that and spoke SS7 or, at the very least ISDN Q.931, and that is much more expensive, inflexible, and difficult to manipulate. Promoting this traffic to a higher-level protocol stack that has more applications and tools to deal with it allows the development of solutions for doing sophisticated telco-world stuff using commodity hardware and open methodologies, open-source style. That has triggered a wave of new products and paradigms in the telco space in a way that is analogous to how Asterisk et al have revolutionised the PBX space. One example of this is TransNexus' NexOSS/NexSRS product (www.transnexus.com http://www.transnexus.com http://www.transnexus.com). They use the OSP (Open Settlement Protocol) module for OpenSER and/or for Asterisk (depending on whether a B2BUA is required) internally inside their product to perform a lot of neat AAA and routing functions (e.g. the NexSRS route server). Their ability to do this benefits precisely from the fact that the traffic can be moved onto a higher-level protocol plane and away from proprietary, expensive, closed and inflexible stuff that has been a defining feature of the telco world. If you can turn the traffic into SIP or H.323, they can deal with it, but if it's SS7 or PRI, they can't. The world is going more soft[ware]. At the same time, the telco space is not a SIP world right now; the network edges are still SS7, and the market really hasn't settled on a good private SIP interconnection/peering strategy and implementation for intercarrier settlement. So, for the most part SIP trunking is used for customer access only. The SS7 information must be conserved in this type of setup, and that's one of the reasons the sort of thing that SIP-T is exists. Alex Balashov wrote: Adrian Georgescu wrote: Why should SIP-T still exist? Is it cheaper than having a gateway? What is the practical use case for investing in such technology? I am eager to learn We've used it extensively in work with CLECs that operate TDM switches such as the Metaswitch, Lucent LCS/Telica, etc. When a carrier operates more than one switch, SS7 interconnection between them is generally required so, for the same basic reasons an internal iBGP mesh or partial mesh (confederation) between two border routers is required for IP. One switch must be aware of numbers routed or ported into the other switch, and so on. The reason for its existence is that if both network elements support SIP-T, it allows you to replace an SS7 IMT (inter-machine trunk) with an IP-based mechanism for this interconnection. This allows you to move the traffic over a data network
Re: [OpenSIPS-Users] Paid Consultation Request
Geoffrey Mina wrote: I generally don't like to presume that individuals want to help me Pro Bono But we do it all day. When you ask the question in $300 terms, you make it a $300 issue. When you ask a question, you make it into a compelling challenge for those who love to help others in the community. -- 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
Re: [OpenSIPS-Users] About new new project of a middle application enabling opensips modules
Oh, I see. Yes, that would be an inappropriate suggestion then, my apologies. Bogdan-Andrei Iancu wrote: Hi Alex, I think Matteo is looking for something to generate OpenSIPS config file and not a simple web interface to add users.. Regards, Bogdan Alex Balashov wrote: http://siremis.asipto.com/ mmarzu...@interfree.it wrote: Hi all. I'm considering the possibility of achieving a middle application between a client who needs to configure a certain scenario in the opensips.cfg and OpenSIPS. The idea is to use a web application that allows you to choose which modules to load, enter the appropriate parameters and enable the appropriate routes in the script. Someone has a suggestion or can suggest a project with similar goals? Thanks a lot for your support. Marzuola matteo Vuoi essere presente online? Vuoi dare voce alla tua attivita`? Acquista un dominio su domini.interfree.it. A partire da 18,59 euro ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [NEW Module] SIP Identity
What's your view of OSP? Adrian Georgescu wrote: Beyond being plain interesting, it is the most cost-efective way to implement secure identity between SIP Proxies serving different domains. Adrian On Feb 10, 2009, at 8:57 PM, Iñaki Baz Castillo wrote: El Martes, 10 de Febrero de 2009, Bogdan-Andrei Iancu escribió: Hello, OpenSIPS 1.5.0 has a new module. The identity module is an implementation of SIP identity as per RFC 4474 (http://www.ietf.org/rfc/rfc4474.txt). Abstract (from RFC) : The existing security mechanisms in the Session Initiation Protocol (SIP) are inadequate for cryptographically assuring the identity of the end users that originate SIP requests, especially in an interdomain context. This document defines a mechanism for securely identifying originators of SIP messages. It does so by defining two new SIP header fields, Identity, for conveying a signature used for validating the identity, and Identity-Info, for conveying a reference to the certificate of the signer Really interesting :) -- Iñaki Baz Castillo ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov 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
Re: [OpenSIPS-Users] OpenSIPS 1.4 / Asterisk / Dispatcher / Probe / 4XX Error
Hi Geoff, It's very strange that Asterisk answers OPTIONS pings with a 4xx error, because OPTIONS is the method Asterisk uses to do its own availability pings -- that's what the qualify= setting for peers in sip.conf enables. What exactly is the 4xx error? Is it 403 Forbidden? Might it have something to do with the domain of the From URI of the request, or the IP it is coming from? Perhaps you just need to set up a SIP peer for OpenSIPS in Asterisk to get it to accept the messages? It would also be helpful -- but not essential -- if you could take a packet capture and post the OPTIONS message OpenSIPS is actually sending to Asterisk, as well as the reply. Cheers, -- Alex Geoffrey Mina wrote: Hello, I am hoping someone can point me in the right direction. I have configured my OpenSIPs server to load balance 10+ asterisk servers using the dispatcher module. To date I have not been able to implement the probe functionality because the OPTIONS and INFO methods both cause asterisk to return a 4XX series error. What options to I have here? Thanks! Geoff ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS 1.4 / Asterisk / Dispatcher / Probe / 4XX Error
Iñaki Baz Castillo wrote: El Domingo, 1 de Febrero de 2009, Alex Balashov escribió: It's very strange that Asterisk answers OPTIONS pings with a 4xx error, because OPTIONS is the method Asterisk uses to do its own availability pings -- that's what the qualify= setting for peers in sip.conf enables. Asterisk only replies 200 for an OPTIONS in case a INVITE with same RURI would be allowed in that context and for that user. This is: if you send an OPTIONS with a RURI that would get a 404 in case of being an INVITE then you will get also a 404 when using the same RURI in an OPTIONS. Ah, OK. So he needs to add a valid peer. -- 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
Re: [OpenSIPS-Users] Warning message at startup
Nothing to worry about; if you are not using the permissions module but loading it anyway, or are using it but keeping your list of authorised peers in the database, you do not need to worry about this. It only matters if you actually wish to list your authorised peers (the things you use the permissions module to authorise) in a text file. Gonzalo Gonzalez wrote: Thanks. It is something I am missing or should I just don't worry about? --- On *Sun, 1/25/09, Iñaki Baz Castillo /i...@aliax.net/* wrote: From: Iñaki Baz Castillo i...@aliax.net Subject: Re: [OpenSIPS-Users] Warning message at startup To: users@lists.opensips.org Date: Sunday, January 25, 2009, 10:46 PM El Domingo, 25 de Enero de 2009, Gonzalo Gonzalez escribió: Jan 25 17:40:44 sipproxy /usr/local/sbin/opensips[6640]: WARNING:permissions:parse_config_file: file not found: /usr/local/etc/opensips/permissions.allow Jan 25 17:40:44 sipproxy /usr/local/sbin/opensips[6640]: WARNING:permissions:mod_init: default allow file (/usr/local/etc/opensips/permissions.allow) not found = empty rule set Jan 25 17:40:44 sipproxy /usr/local/sbin/opensips[6640]: WARNING:permissions:parse_config_file: file not found: /usr/local/etc/opensips/permissions.deny Jan 25 17:40:44 sipproxy /usr/local/sbin/opensips[6640]: WARNING:permissions:mod_init: default deny file (/usr/local/etc/opensips/permissions.deny) not found = empty rule set Try yourself: what do you think file not found can mean? ;) -- Iñaki Baz Castillo ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov 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
Re: [OpenSIPS-Users] Asteriak load balance
Dispatcher. Gonzalo Gonzalez wrote: What is the best module to use for load balance with 5 asterisk servers? user Opensips --- Asterisk --- PSTN ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CDRTool destinations /rates question
Unlike Brian, I am not familiar with CDRtool beyond a cursory level, so perhaps I'm headed down the wrong track here. The general problem seems to be that the multiple destination problem (variable-length prefixes) is multidimensional, so it is not just a matter of sending to the longest dial prefix match for a given destination. The carrier must also be taken into account. So, what is needed seems to be a destination metric that is a composite rate of a gateway and a longest-prefix destination. The terminating carriers are fixed by a static LCR process. Is that right, Brian? Adrian Georgescu wrote: Alex, I am trying to understand what precisely you are trying to achieve. What precisely are you working around that cannot be done in a natural way? Adrian On Jan 20, 2009, at 7:29 PM, Alex Balashov wrote: Good workaround is to use translations in the proxy to prepend a prefix for each carrier to the DNIS so you can set the rating engine loose on that. This is how billing systems attached to traditional softswitch EMSs work. Brian Chamberlain wrote: Thanks Adrian, As I said, just trying to find an efficient way of doing this, all the providers use different destination names, some have codes that don't exist in the other's databases so trying to pull it all together in CDRtool is proving a bit testing. It is mentioned as a known limitation 'The rating engine does not calculate prices based on the outbound carriers or outbound gateways, the rating plan is is assigned by the calling party and not by called party.' I guess I am trying to figure out an efficient way to deal with the slight nuances with different providers destination codes and descriptions and the overlaps in between.. If it was possible to rate with the destination gateway it would make things a lot easier. Thanks, Brian On 20 Jan 2009, at 15:38, Adrian Georgescu wrote: If dest is 1 only rate for dest 1 is applied. There is no longest match performed for a dest column in a rate table entry. If you want a rate for 1617, add it to the dest table too. Adrian On Jan 20, 2009, at 4:19 PM, Brian Chamberlain wrote: Hi Adrian, Thanks for the quick response. As I thought! Can you just confirm that if I have 1 as a destination,1 as a rate and also 1617 as a rate and 1617 is the number dialled then according to the documentation the rating engine will find the 1 destination but will do a longest match and find 1617 as the rating record or am I hoping for too much? Regards, Brian On 20 Jan 2009, at 15:03, Adrian Georgescu wrote: Hi Brian, The logic of the rating first determines the destination then it searches for a price for it. So for every entry in destinations table you MUST have an entry in the rates table otherwise the price is zero. The best practice is to maintain a central minium destination table common for all customers (add entries to it as it grows) and define custom rates for each of them. Also if you have lot of resellers you can create a main rating table and add only exceptions for the destinations particular to some of them. Adrian On Jan 20, 2009, at 3:56 PM, Brian Chamberlain wrote: Hi All, I am sending calls to a number of different sip providers. I have rates destinations from all of them. Some of the providers have broken up the amount of destinations into 30,000 different codes. I am trying to build the rates and destinations tables so it is easy to maintain in the future. Would I be best having a minimal set of destinations to cover each country and my local countries/areas and having the rates being more specific. I suppose my questions are the folowing. If I have a destination: 1 USA and a rate for 1 USA .02 and a rate for 1617 USA (Boston) and the customer dials Boston then looking at the logic, even though I don't have a boston Destination CDRTool will still rate the call using the rate for 1617 If the reverse was through and I had a destination 1617 for boston but only a rate for 1 USA would CDRTool use the 1 rate even though it found the destination for 1617 in the destinations table? Thanks, Brian ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users Brian Chamberlain Dot Net Solutions Ltd. 68 Parkwest Enterprise Centre, Parkwest, Dublin 12, Ireland. DDI: [+353] 1 6296521 FAX: [+353] 1 6237029 mobile: [+353] 86 3883003 web: www.asterisk.ie http://www.asterisk.ie * Looking for the most advanced PBX available that can also save you a fortune in communication costs? asterisk.ie * e-mail disclaimer This e-mail and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended
Re: [OpenSIPS-Users] Registered user
But doesn't that check if the AOR in the RURI can be located. Michel, The proper way to do this -- assuming your motive is security and authorisation -- is to challenge the incoming INVITE initial request of the caller (who is supposed to be registered) with a 407 proxy challenge, i.e. proxy_authorize()/proxy_challenge(). Bogdan-Andrei Iancu wrote: Hi Michel, See the registered() function from the registrar module: http://www.opensips.org/html/docs/modules/1.4.x/registrar.html#id271407 Regards, Bogdan michel freiha wrote: Dear All, I need to ask please about which function should I use in order to check while making a call if the user who is dialing the number is making the call from a registered account or not? 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 -- 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
Re: [OpenSIPS-Users] [Kamailio-Users] Registered user
Daniel, I am curious, what is the intended use case of this: check if a user is calling from a registered device and if not, deny the call Why not just issue a 407 Proxy Challenge for the incoming INVITE? -- ALex Daniel-Constantin Mierla wrote: Hello, On 01/16/2009 03:31 PM, michel freiha wrote: Dear All, I need to ask please about which function should I use in order to check while making a call if the user who is dialing the number is making the call from a registered account or not? if you want to check if the user is calling from a registered phone, you have to use kamailio trunk. See second example here: http://openser.blogspot.com/2008/10/registrar-enhancements.html Module documentation at: http://www.kamailio.org/docs/modules/devel/registrar.html Cheers, Daniel -- 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
Re: [OpenSIPS-Users] [Kamailio-Users] Registered user
Do you care if it's online, as long as it answers the challenge successfully with the same credentials it provides when it registers? Daniel-Constantin Mierla wrote: On 01/16/2009 03:53 PM, Alex Balashov wrote: Daniel, I am curious, what is the intended use case of this: check if a user is calling from a registered device and if not, deny the call Why not just issue a 407 Proxy Challenge for the incoming INVITE? you must authenticate the call, this check comes after, to be sure the user is calling from a phone that was previously registered (so it is online). If you check the discussions from the last days, one good thing of doing this is to prevent SIP Digest Access Authentication RELAY. One can call from a sip phone even that phone is not registered (REGISTER-200ok). Cheers, Daniel -- ALex Daniel-Constantin Mierla wrote: Hello, On 01/16/2009 03:31 PM, michel freiha wrote: Dear All, I need to ask please about which function should I use in order to check while making a call if the user who is dialing the number is making the call from a registered account or not? if you want to check if the user is calling from a registered phone, you have to use kamailio trunk. See second example here: http://openser.blogspot.com/2008/10/registrar-enhancements.html Module documentation at: http://www.kamailio.org/docs/modules/devel/registrar.html Cheers, Daniel -- 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
Re: [OpenSIPS-Users] [Kamailio-Users] Registered user
Bogdan, In OpenSIPS, are AVP searches still linear for AVPs that have string identifiers? My understanding is that traditionally they were linear, as opposed to hashed. Thanks, -- Alex Bogdan-Andrei Iancu wrote: Of course this notation is present since openser 1.3 and it was inherited by both OpenSIPS 1.4.4 and Kamilio 1.4.3, but now we try to get a better approach of this functionality: why put the value into an AVP and let the function search all the time for that AVP (set or not set), when you can simply take advantage and directly pass the value as parameter to the function. You get read of (1) useless transit via an AVP and (2) useless AVP search all the time. Also you get a more compact and clear scripting -- 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
Re: [OpenSIPS-Users] Multi Tenant System
ram wrote: Hi is this possible with Opensips Multi Tenant system ( integrating with Asterisk or Freeswitch) Yes. if yes, any advise how this can be achived ? any documents Well, you integrate OpenSIPS with Asterisk or FreeSWITCH. -- 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
Re: [OpenSIPS-Users] alias_db failing to look up alias for IP using multi_domain
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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] alias_db failing to look up alias for IP using multi_domain
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 abalas...@evaristesys.com wrote: Have you tried adding both combinations to the DB manually without using opensipsctl? On Jan 4, 2009, at 10:55 PM, Julian Yap julianok...@gmail.com 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 abalas...@evaristesys.com 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] alias_db failing to look up alias for IP using multi_domain
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 abalas...@evaristesys.com 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 abalas...@evaristesys.com wrote: Have you tried adding both combinations to the DB manually without using opensipsctl? On Jan 4, 2009, at 10:55 PM, Julian Yap julianok...@gmail.com 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 abalas...@evaristesys.com 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
Re: [OpenSIPS-Users] Too many hops
Most likely you are relaying INVITEs or other end-to-end requests
without altering the Request URI domain, thus forwarding them back to
the proxy.
On Dec 29, 2008, at 3:02 PM, Jeff Pyle jp...@fidelityvoice.com wrote:
Most likely there is a problem in your routing logic that is causing
a loop. Each iteration of the loop causes one more “hop”. At
some point, there are too many hops.
I would suggest adding some xlog lines to your script and watching
the log output to see where your routing is taking you. You should
be able to see where the problem is with that.
- Jeff
On 12/29/08 2:22 PM, J Santos jsantos5...@gmail.com wrote:
Hi all,
I have two Xlite registered with opensips when configured with the
box IP address. I can make calls between these phones.
I created a SRV record and configured one of the phones with the
domain and it is not registering. It is returning the message
Registration error:483 Too many hops.
It looks like it is coming from
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
but why ? I have only these two phones in the network.
Firewall is disabled and I am forwarding ports UDP 5060-5070 to the
opensips box.
Any ideas?
thanks
Jair Santos
___
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] SIP - Transaction
What do you mean by implement SIP-T? SIP-T provides ISUP encapsulation in the payload to replace SS7 interworking; it's still SIP, and you still treat it the same way you do any other SIP. Bruno Rodrigues wrote: Hi All, Have any idea to implement SIP-T in opensips ? Thank You, Bruno Rodrigues ___ 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) (706) 338-8599 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP - Transaction
OpenSIPS does not control this signaling, and is not involved at the MTP3 level. Bruno Rodrigues wrote: I mean use Opensips like a SCP. The GW using SS7 sending the mtp3 to Opensips using sip-t and Opensips controlling this signaling. -Original Message- From: Alex Balashov [mailto:abalas...@evaristesys.com] Sent: segunda-feira, 22 de dezembro de 2008 23:14 To: Bruno Rodrigues Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] SIP - Transaction What do you mean by implement SIP-T? SIP-T provides ISUP encapsulation in the payload to replace SS7 interworking; it's still SIP, and you still treat it the same way you do any other SIP. Bruno Rodrigues wrote: Hi All, Have any idea to implement SIP-T in opensips ? Thank You, Bruno Rodrigues ___ 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) (706) 338-8599 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP - Transaction
For one, that would require some sort of user agent (M3UA), which OpenSIPS is not. If anything, Asterisk would be better suited for that role if it could somehow combine its incipient SS7 ISUP support with real, solid call control/media gateway control support (i.e. MGCP or H.248/MEGACO). Bruno Rodrigues wrote: I know OpenSips dont control the ISUP signaling and not is involved with MTP3 Level. My doubts if the opensips will can control SigTran in future or if have any idea to implement this. I ask about this because we don't have a open source softswitch what can control Sigtran signaling using any protocol (H.248/SIP-T) -Original Message- From: Alex Balashov [mailto:abalas...@evaristesys.com] Sent: segunda-feira, 22 de dezembro de 2008 23:37 To: Bruno Rodrigues Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] SIP - Transaction OpenSIPS does not control this signaling, and is not involved at the MTP3 level. Bruno Rodrigues wrote: I mean use Opensips like a SCP. The GW using SS7 sending the mtp3 to Opensips using sip-t and Opensips controlling this signaling. -Original Message- From: Alex Balashov [mailto:abalas...@evaristesys.com] Sent: segunda-feira, 22 de dezembro de 2008 23:14 To: Bruno Rodrigues Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] SIP - Transaction What do you mean by implement SIP-T? SIP-T provides ISUP encapsulation in the payload to replace SS7 interworking; it's still SIP, and you still treat it the same way you do any other SIP. Bruno Rodrigues wrote: Hi All, Have any idea to implement SIP-T in opensips ? Thank You, Bruno Rodrigues ___ 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) (706) 338-8599 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] route
textops module. Raghavendra D P wrote: Hi Route :sip:190.10.19.20, sip:45:128 I am using oopensips 1.4 How to remove fist route information *Thanks and Regards* *Raghavendra DP**|** Tech Mahindra* 9/7, Hosur Road, Bangalore – 560 029, India ( Office: +91 80 4024 3458 *|* Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] *www.techmahindra.com http://www.techmahindra.com* Disclaimer: This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review the policy at http://www.techmahindra.com/Disclaimer.html externally and http://tim.techmahindra.com/Disclaimer.html internally within Tech Mahindra. ___ 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) (706) 338-8599 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [Kamailio-Users] rtpproxy
Check out www.rtpproxy.org. On Fri, November 14, 2008 9:24 am, michel freiha wrote: Hi all, I installed OpenSer on my Centos machine and everything worked fine...I need now to install rtpproxy on the same machine but did not find any good documentation that can help me about that... I need to know please from where I can download the rtpproxy package and how I can configure it Regards ___ Users mailing list [EMAIL PROTECTED] http://lists.kamailio.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) (706) 338-8599 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Check request come from registered user
Or that. Brett Nemeroff wrote: maybe check_from() in uri_db? On Fri, Nov 14, 2008 at 3:38 PM, Giuseppe Roberti [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi. How can i check that a request come from a registered user ? Regards. -- Giuseppe Roberti [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] reply with CANCEL message
Alex R.S.M wrote: The INVITE request to End-point B generated with append_branch() within openSIP. So how openSIP knows to generate a CANCEL message when one End-point answers the call? Are you generating it manually or using the registrar's forking mechanism? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] a simple perl question
Chris wrote:
On Wednesday 12 November 2008 1:50:59 pm Robert R wrote:
How can I return a string value from perl function in openSER?
return $x; is not working.
Here's the way I'm doing it...from my Perl script code:
if ($routeid) {
# set AVP variable with the destination route ID to route call to
OpenSIPS::AVP::add(369,$routeid);
}
And then in the OpenSIPS script opensips.cfg, I can read it:
if ($avp(i:369) == whatever) {
..
}
Ditto. That is the only way I have gotten it to work.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
