sr-users November 2005

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245 participants
571 discussions

[Users] How to gaurd PSTN gateway with openser + NAT
by Lokesh Kumar 17 Nov '05

17 Nov '05

Hello Friends, I installed openser, it is working fine. sipsak-0.9.0 is also working fine, previously i was using 0.9.2 version of sipsak that is having problem. Now I want to ask you 2 important things. 1>> Forget Asterisk for sometime, I am having a openser with one public IP and one private IP, and I am having one voip gateway which is connected to E1 PRI, and voip gateway is also having private IP. So, my question is can I forward all the calls coming on public IP of openser to voip gateway via private IP's of openser and voip gateway.After that voip gateway will forward the call to E1 PRI. SER Admin guide mentioned about how to gaurd your PSTN gateway with SER, but the documentation is not suffciente there. So, can you please suggest in this scenario, how do i can configure request routing in openser, like pstn.cfg file. 2>> How do I have to register SIP phone on openser. Where the sip phones are behind the NAT, but NAT device supports STUN and UPnP. Do I need to setup seperate STUN server or I can run this service on openser server. Please help. Regards Lokesh ----- Original Message ----- From: <deepak.dhiman(a)uf4.net> To: "Lokesh Kumar" <lokesh(a)interacesso.pt> Sent: Tuesday, November 15, 2005 8:42 PM Subject: Re: Need Help in configuring Asterisk with SER and PSTN gateway > Hi Lokesh ! > > here i am writing my own experience baed views and facts: - > > 1) openser i certainly better than ser with better nat handling tricks, it > also has got TLS support first time ina sip server and certain other > enhanced feature are alo there, that can be find out on www.voip-info.org > > 2)sipsak is working fine on my pc and i really had no problem in installing > it, i have installed vesion sipsak-0.9.0 > > 3) i have not istalled any ser administrative program, so i can't tell you > which one is better. > > 4) in case of Radius server try with radius 0.9.0. > > 5) Finally, ser doesn't have any pstn support so you can't connect it to the > pstn gateway, > you can forward all sip packets to asterisk server and it would forward it > to pstn gateway and i think that's is the possible solution of your problem. > > regards, > > Deepak > > Lokesh Kumar writes: > > > Hello Deepak, > > > > Here are some of the questions which if you can respond then i will be quite > > happy. > > > > 1>> What is the difference between SER and openSER and which one is best to > > adopt in terms of security and ease of use and customization. > > 2>> I am unable to install sipsak on SER, it gives me error, when i execute > > make. > > 3>> Which is the best web tool to manage SER, I tried SERweb and SERadmin, > > but unfortunately unable to configure them with SER > > 4>> How do I have to setup a Voip trunk between asterisk and SER, where SER > > forwards any requests from asterisk to PSTN gateway. > > 5>> How do I have to inter conect Radius with SER. > > > > I had searched everything on the internet but sufficiente documents are not > > available. > > > > Looking for your reply > > Thanks a lot > > Lokesh > > > > > > ----- Original Message ----- > > From: <deepak.dhiman(a)uf4.net> > > To: "Lokesh Kumar" <lokesh(a)interacesso.pt> > > Sent: Monday, November 14, 2005 8:49 PM > > Subject: Re: Need Help in configuring Asterisk with SER and PSTN gateway > > > > > >> Hi Lokesh ! > >> > >> I think u can easily configure it, let me know what have u tried so far so > >> that i can tell u exactly where is the problem.. > >> > >> regard, > >> > >> Deepak > >> > >> Lokesh Kumar writes: > >> > >> > Hello, > >> > > >> > Deepak, > >> > > >> > I got your mail from the asterisk lists. I wanted to ask you about how > > you had configured asterisk with SER. I am facing lots of difficulties with > > configuration of asterisk with SER. Can you please guide me . I will be very > > thankfull to you. > >> > > >> > Regards > >> > > >> > >> > >> > >> Software Engg. Trainee > >> Trail Ridge Software India Pvt. Ltd. > >> Noida > >> > > > > > > Software Engg. Trainee > Trail Ridge Software India Pvt. Ltd. > Noida >

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[Serusers] Ser Business Mailing List ?
by Rehan Ahmed 17 Nov '05

17 Nov '05

Hello All, Is there a SER Business Mailing list for commerical only emails on ? Rehan www.supertec.com <http://www.supertec.com> www.didxchange.com <http://www.didxchange.com>

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[Users] exec_dset syntax error
by Iqbal 17 Nov '05

17 Nov '05

Hi can anyone see what wrong with exec_dset("/path/to/program.php") unless of course exec_dset only likes shell scripts Iqbal

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[Users] exec-msg
by Iqbal 17 Nov '05

17 Nov '05

Hi Anyoe got shell script to parse all the parameters sent, from my understanding the entire sip message is sent, into the script, I just want to pull the username out from From: iqbal

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[Users] A tutorial on the OSP peering module is now available
by Dmitry Isakbayev 17 Nov '05

17 Nov '05

Dear Users and Developers, A tutorial on the OSP peering module is now available at http://openser.org/docs/osp/Multi-Lateral_Peering_with_OpenSER_1_0_0.pdf. The module enables secure, multi-lateral peering with SIP devices that support OSP (i.e. OpenSER 1.1.x, OpenSER 1.0.x, (Open)SER 0.9.x, Asterisk 1.2). Open source OSP servers are available at www.sipfoundry.org/OSP. Regards, Dmitry

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[Serusers] Dropping 200-ok?
by Ricardo Poppi 17 Nov '05

17 Nov '05

Hi list. I want to make an analisis at a 200-ok response from an INVITE. I implemented at ser 0.9.4 an "on_reply_route" to check if the 200-ok contains the SDP address of the gateway I want to connect. It its true, I let the 200-ok flows. If it´s not, I want to drop this and terminate the call (SIP ERROR or whatever). Here is the example ============= ------------- "A" =======> | "SER" | =========> "B" ------------- 1) "A" INVITEs "B" through SER 2) "B" says that its trying and ringing to "A" 3) User at "B" picks-up 4) I check the 200-ok at SER. If it´s ok, let it flow to "A". If it´s not, I want to drop it and terminate the call establishment. DO ANYBODY DID THAT OR CAN HELP ME WITH ANY CLUE/STARTING POINT? best regards to all. Ricardo Poppi.

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[Users] obscufucation...spelt that wrong ;-)
by Iqbal 17 Nov '05

17 Nov '05

Hi Any chance in next versions of openser, of having a module if activated that it would be able to read/create a unreadable config file, but would be readable by ser itself. Or is there already a way of doing this iqbal

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[Users] in dialog request when openser listens on multiple ports
by Klaus Darilion 17 Nov '05

17 Nov '05

Hi! I've configure openser to listen also on port udp:6060. The INVITE comes in to 6060, forwarded to other client on port 5060. Responses and ACK will be handled correctly - on call leg uses port 6060, the other call leg 5050. If now the callee sends a BYE (Cisco, strict router) the BYE will be forwarded from port 5060 instead of 6060. I didn't find a problem in the BYE request sent to the openser. Is openser behaving wrong here? Or do I have to configure special routing for this scenario? thanks klaus BYE from callee (5060) to proxy U 2005/11/09 16:47:48.254756 83.136.33.19:5060 -> 83.136.32.83:5060 BYE sip:83.136.32.83:5060;r2=on;ftag=a613b273;nat=yes;lr=on SIP/2.0. Via: SIP/2.0/UDP 83.136.33.19:5060;branch=z9hG4bK3dc73e84. From: <sip:01505641636@enum.at>;tag=000dedfb04cc011e597bda33-277ea073. To: klaus enum.at<sip:klaus@enum.at>;tag=a613b273. Call-ID: 7f49f21ed451bb7a. Date: Wed, 09 Nov 2005 15:47:47 GMT. CSeq: 101 BYE. User-Agent: CSCO/6. Content-Length: 0. RTP-RxStat: Dur=??,Pkt=??,Oct=??,LatePkt=??,LostPkt=??,AvgJit=??. RTP-TxStat: Dur=??,Pkt=??,Oct=??. Route: <sip:83.136.32.83:6060;r2=on;ftag=a613b273;nat=yes;lr=on>, <sip:klaus@84.20.167.143:7404>. . BYE from proxy to caller: is sent from 5060 althouth it should be sent from 6060. # U 2005/11/09 16:47:48.264940 83.136.32.83:5060 -> 84.20.167.143:7404 BYE sip:klaus@84.20.167.143:7404 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:83.136.32.83;ftag=000dedfb04cc011e597bda33-277ea073;nat=yes;lr=on>. Via: SIP/2.0/UDP 83.136.32.83;branch=z9hG4bKaf5b.fbc8b274.0. Via: SIP/2.0/UDP 83.136.33.19:5060;branch=z9hG4bK3dc73e84. From: <sip:01505641636@enum.at>;tag=000dedfb04cc011e597bda33-277ea073. To: klaus enum.at<sip:klaus@enum.at>;tag=a613b273. Call-ID: 7f49f21ed451bb7a. Date: Wed, 09 Nov 2005 15:47:47 GMT. CSeq: 101 BYE. User-Agent: CSCO/6. Content-Length: 0. RTP-RxStat: Dur=??,Pkt=??,Oct=??,LatePkt=??,LostPkt=??,AvgJit=??. RTP-TxStat: Dur=??,Pkt=??,Oct=??. .

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Re: [Serusers] Re: [Serdev] ACK before INVITE on re-INVITE causes CiscoAS's to return a 500 error
by Evan Borgström 17 Nov '05

17 Nov '05

Hey Ettore, Yes, both INVITEs contain valid and correct SDP info. We have a maintenance scheduled to upgrade IOS on the Cisco devices so hopefully that will help it. -Evan Ettore Benedetti wrote: > Hallo Evan, > > I reckon Nils is right on the matter. The ACK of the dialog-forming INVITE is actually > the first in-dialog request and, as a matter of fact, it is a separate transaction. As far > as the UAS is concerned, the dialog is established as soon as the 200 response is sent. > The Cisco GW just misbehaves provided the first INVITE from Asterisk did contain > an SDP offer (didn't it?). > > > Ettore Benedetti > > THALES COMMUNICATIONS B.V. > Bestevaer 46, 1271 ZA Huizen > The Netherlands > > Unclassified > > >>>> Evan Borgstrom <evan.borgstrom(a)ca.mci.com> 11/16/05 5:27:52 PM >>> > Hey Nils, > > You are correct that each message is handled by a different process, as > it to be expected, and I know that SER is not call statefull so my > concern lies around the fact that SER is delivering an INVITE which > requires FAR more processing and at least 2 database hits before a > loosely routed ACK which requires little to none for processing. > > I'm going to differ with you, and my self from last night, on the views > about the RFC as it does indeed cover the problem (I was just looking at > it the wrong way before). A re-INVITE cannot modify a dialog that does > not exist, and without the ACK being delivered in response to the 200 OK > the dialog is not yet created which is why the Cisco is returning a > CheckRequest failure since there's no dialog to map the re-INVITE to. > So, the device is correct in returning an error. > > When the Cisco returns the 500 error it's only for the re-INVITE so the > initial call stays established and a BYE is still required to tear it > down. The problem is that Asterisk is expecting that to work and has > already re-INVITED to the client so there's no audio and to the caller > the call failed. > > -Evan > > Nils Ohlmeier wrote: >> Evan, >> >> I would not invest the time to look into the SER side of your problem. Because >> this is a general problem. Instead of SER any other network element (router, >> switch, etc.) could change the order of the packets. >> BTW I guess this problem occurs on your SER because the ACK and the INVITE are >> handled by different SER processes. And as SER is not call-statefull there is >> no parameter to solve this situation. >> >> I fear you are right that this problem is not covered by any RFC, but the >> solutions seems logical to me: >> if this a generic problem, then either the UAC is not allowed to send a >> re-INVITE until the first INVITE is completed, or the UAS has to accept a >> re-INVITE after sending the 200 OK. But how does the the UAC know that the >> ACK reached the UAS so that it can send the re-INVITE now? It cant know that, >> so it would have to wait for a timer before it could send the re-INVITE. >> On the other side the INVITE transaction is completed for the UAS by sending >> the 200 OK (the ACK is a new transaction, according to the branch). >> Thus I think the UAS (the Cisco gateway) should accept the re-INVITE without >> complaining. >> >> Just for curiosity: is the call established or does the Cisco gateway drop the >> call after sending the 500? >> >> Regards >> Nils >> >> On Wednesday 16 November 2005 07:51, Greger V. Teigre wrote: >>> Evan, >>> I'm not able to match your text with the messages you show. The message >>> timestamped 16:51:51.559657, does it go to Cisco? It is shown to go to >>> Asterisk and it doesn't make sense. >>> However, until the Cisco gw receives the ACK from Asterisk, no call has >>> been established and the reINVITE cannot be matched. IMO, this is not a >>> Cisco bug. It seems very strange indeed that ser should send a reINVITE >>> before the ACK in the same call. Even without FIFO, an INVITE is likely to >>> take more time to process than a loose routed ACK. I would try to log more >>> thoroughly in SER to see what happens in processing the ACK and the >>> reINVITE. It could be a problem in your ser.cfg, but it seems strange >>> indeed. >>> g-) >>> >>> ----- Original Message ----- >>> From: "Evan Borgstrom" <evan.borgstrom(a)ca.mci.com> >>> To: <serusers(a)lists.iptel.org>; <serdev(a)lists.iptel.org> >>> Sent: Wednesday, November 16, 2005 12:05 AM >>> Subject: [Serdev] ACK before INVITE on re-INVITE causes Cisco AS's to >>> return a 500 error >>> >>>> Hey all, >>>> >>>> This has more to do with Asterisk & Cisco AS's IMO but I thought it >>>> would be interesting to get some responses from this group before I >>>> begin the battle with Cisco to get a bug report underway. >>>> >>>> We have an Asterisk box acting as a PBX sending calls to our SER >>>> instances. The asterisk box has "reinvite=yes" in the sip.conf file >>>> which causes Asterisk to send a re-INVITE for the call once it receives >>>> the 200 OK. This re-INVITE has it's SDP changed to that of the IP Phone >>>> behind the Asterisk box so that all media doesn't need to traverse it. >>>> This works fine except in certain instances where messages get out of >>>> order and it causes our Cisco GW's to error the call with a generic 500 >>>> message. When the messages are in the correct order the 500 error does >>>> not appear. >>>> >>>> So here's the situation, user behind the asterisk box picks up the >>>> phone and dials a PSTN destination. Asterisk fires off the initial >>>> invite and everything proceeds as normal and we eventually receive the >>>> 200 OK message to which the Asterisk box replies with ACK and a new >>>> INVITE that come in this order (X = Asterisk, Y = SER, Z = Cisco): >>>> >>>> U 2005/11/15 16:51:51.557322 X.X.X.X:5060 -> Y.Y.Y.Y:5060 >>>> ACK >>>> >>>> # >>>> U 2005/11/15 16:51:51.557948 X.X.X.X:5060 -> Y.Y.Y.Y:5060 >>>> INVITE >>>> >>>> SER accordingly processes the two messages and replies 100 Trying to >>>> the new INVITE message. Here's where the problem occurs, SER then sends >>>> these messages out to the PSTN gateway but the order has been reversed: >>>> >>>> # >>>> U 2005/11/15 16:51:51.558937 Y.Y.Y.Y:5060 -> X.X.X.X:5060 >>>> SIP/2.0 100 trying >>>> >>>> # >>>> U 2005/11/15 16:51:51.559000 Y.Y.Y.Y:5060 -> Z.Z.Z.Z:5060 >>>> INVITE >>>> >>>> # >>>> U 2005/11/15 16:51:51.559657 Y.Y.Y.Y:5060 -> X.X.X.X:5060 >>>> ACK >>>> >>>> When this happens the Cisco GW replies with a 500 error message and the >>>> following is sent to the Cisco's logs: >>>> >>>> # >>>> U 2005/11/15 16:51:51.711355 Z.Z.Z.Z:5060 -> Y.Y.Y.Y:5060 >>>> SIP/2.0 500 Server Internal Error. >>>> >>>> >>>> Nov 15 21:51:51.613: HandleUdpSocketReads :Msg enqueued for SPI with >>>> IPaddr: Y.Y.Y.Y:5060 >>>> Nov 15 21:51:51.613: *****CCB found in UAS Request table. ccb=0x653BCD0C >>>> Nov 15 21:51:51.613: CCSIP-SPI-CONTROL: act_sentsucc_new_message >>>> Nov 15 21:51:51.617: CCSIP-SPI-CONTROL: act_sentsucc_new_message_request >>>> Nov 15 21:51:51.617: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE >>>> Nov 15 21:51:51.617: sip_stats_status_code >>>> Nov 15 21:51:51.617: sipSPICheckRequest: CheckRequest fail on method 102 >>>> error code: 2 and status: 500 >>>> >>>> >>>> Is there any parameter that I can set to ensure that message leave SER >>>> in the order they enter? Should the Cisco be able to handle this >>>> condition (I check a number of different sections of the RFC but >>>> couldn't find anything specific for either side)? >>>> >>>> Thanks, >>>> Evan >>>> >>>> _______________________________________________ >>>> Serdev mailing list >>>> serdev(a)lists.iptel.org >>>> http://lists.iptel.org/mailman/listinfo/serdev >>> _______________________________________________ >>> Serdev mailing list >>> serdev(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serdev > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] mangler module
by harry gaillac 17 Nov '05

17 Nov '05

Hello, I try to change the contact field in SIP HF with mangler module if (src_ip == 10.0.0.0/8) encode_contact("enc_prefix","193.175.135.38"); However asterisk receive a bad AOR because of "enc_prefix" . How can I rewrite contact field with public ip without adding this prefix in th user part ? Regards Harry Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 ---------------- | asterisk pbx | ---------------- || ---------- ---------- | SER |=======|NAT box |======== private network ---------- ---------- ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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sr-users November 2005