sr-users November 2005

sr-users@lists.kamailio.org

245 participants
570 discussions

by Marcello Lupo

Hi i'm getting this errors of files not found compiling OPenser stable 1.0.0 on Suse Linux Enterprise Server 9 x86_64 . Can you address me to solve the issue? After the compilation the Openser compile but i don't know if it can be an operational problem. After make proper i get: venus:/data/src/openser-1.0.0 # make Makefile.rules:91: action.d: No such file or directory Makefile.rules:91: crc.d: No such file or directory Makefile.rules:91: daemonize.d: No such file or directory Makefile.rules:91: data_lump.d: No such file or directory Makefile.rules:91: data_lump_rpl.d: No such file or directory Makefile.rules:91: dprint.d: No such file or directory Makefile.rules:91: dset.d: No such file or directory Makefile.rules:91: error.d: No such file or directory Makefile.rules:91: fifo_server.d: No such file or directory Makefile.rules:91: flags.d: No such file or directory Makefile.rules:91: forward.d: No such file or directory Makefile.rules:91: hash_func.d: No such file or directory Makefile.rules:91: ip_addr.d: No such file or directory Makefile.rules:91: items.d: No such file or directory Makefile.rules:91: main.d: No such file or directory Makefile.rules:91: md5.d: No such file or directory Makefile.rules:91: md5utils.d: No such file or directory Makefile.rules:91: modparam.d: No such file or directory Makefile.rules:91: msg_translator.d: No such file or directory Makefile.rules:91: pass_fd.d: No such file or directory Makefile.rules:91: proxy.d: No such file or directory Makefile.rules:91: qvalue.d: No such file or directory Makefile.rules:91: re.d: No such file or directory Makefile.rules:91: receive.d: No such file or directory Makefile.rules:91: resolve.d: No such file or directory Makefile.rules:91: route.d: No such file or directory Makefile.rules:91: route_struct.d: No such file or directory Makefile.rules:91: script_cb.d: No such file or directory Makefile.rules:91: socket_info.d: No such file or directory Makefile.rules:91: sr_module.d: No such file or directory Makefile.rules:91: stats.d: No such file or directory Makefile.rules:91: tcp_main.d: No such file or directory Makefile.rules:91: tcp_read.d: No such file or directory Makefile.rules:91: timer.d: No such file or directory Makefile.rules:91: tsend.d: No such file or directory Makefile.rules:91: udp_server.d: No such file or directory Makefile.rules:91: unixsock_server.d: No such file or directory Makefile.rules:91: usr_avp.d: No such file or directory Makefile.rules:91: mem/f_malloc.d: No such file or directory Makefile.rules:91: mem/mem.d: No such file or directory Makefile.rules:91: mem/memtest.d: No such file or directory Makefile.rules:91: mem/q_malloc.d: No such file or directory Makefile.rules:91: mem/shm_mem.d: No such file or directory Makefile.rules:91: mem/vq_malloc.d: No such file or directory Makefile.rules:91: parser/hf.d: No such file or directory Makefile.rules:91: parser/msg_parser.d: No such file or directory Makefile.rules:91: parser/parse_allow.d: No such file or directory Makefile.rules:91: parser/parse_content.d: No such file or directory Makefile.rules:91: parser/parse_cseq.d: No such file or directory Makefile.rules:91: parser/parse_disposition.d: No such file or directory Makefile.rules:91: parser/parse_diversion.d: No such file or directory Makefile.rules:91: parser/parse_event.d: No such file or directory Makefile.rules:91: parser/parse_expires.d: No such file or directory Makefile.rules:91: parser/parse_fline.d: No such file or directory Makefile.rules:91: parser/parse_from.d: No such file or directory Makefile.rules:91: parser/parse_hname2.d: No such file or directory Makefile.rules:91: parser/parse_hostport.d: No such file or directory Makefile.rules:91: parser/parse_methods.d: No such file or directory Makefile.rules:91: parser/parse_nameaddr.d: No such file or directory Makefile.rules:91: parser/parse_param.d: No such file or directory Makefile.rules:91: parser/parse_refer_to.d: No such file or directory Makefile.rules:91: parser/parser_f.d: No such file or directory Makefile.rules:91: parser/parse_rpid.d: No such file or directory Makefile.rules:91: parser/parse_rr.d: No such file or directory Makefile.rules:91: parser/parse_sipifmatch.d: No such file or directory Makefile.rules:91: parser/parse_to.d: No such file or directory Makefile.rules:91: parser/parse_uri.d: No such file or directory Makefile.rules:91: parser/parse_via.d: No such file or directory Makefile.rules:91: parser/digest/digest.d: No such file or directory Makefile.rules:91: parser/digest/digest_parser.d: No such file or directory Makefile.rules:91: parser/digest/param_parser.d: No such file or directory Makefile.rules:91: parser/contact/contact.d: No such file or directory Makefile.rules:91: parser/contact/parse_contact.d: No such file or directory Makefile.rules:91: db/db.d: No such file or directory Makefile.rules:91: db/db_fifo.d: No such file or directory Makefile.rules:91: db/db_id.d: No such file or directory Makefile.rules:91: db/db_pool.d: No such file or directory Makefile.rules:91: lex.yy.d: No such file or directory Makefile.rules:91: cfg.tab.d: No such file or directory bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce flex cfg.lex I checked in the main dir of openser src and the files are present. May be the Makefile is searching somewhere else ? Thanks, BYe, Marcello

19 years

[Serusers] Cisco 3640 connect to PBX

by Akarach Rattanawaree

Hello, I have Cisco 3640 with Voice and E1 card connected to Nortel PBX for call out from PBX to PSTN. I have problem when 1. user hang-up the phone and got dial tone from PBX 2. dial prefix 9 and telephone number (9,024455566) 3. user will got dial tone again from Cisco 4. If user dial telephone number again, it can go to PSTN. (024455566) User ------ > PBX ----- > Cisco 3640 ----- > SIP Server ----- > PSTN Have you ever found this case? I have tried to use command "isdn overlap-receiving" on serial interface. Thank you Ake

19 years

[Serusers] Mediaproxy on Multiprocessor System

by tcchan

Hi All, I run mediaproxy on a 2 CPUs machine (Fedora Code 3). When I stress the mediaproxy with rtpgenerator locally, i.e., just rtpgenerator.py --count=50, I noticed that the load is distributed evenly between the 2 CPUs. But when I stress remotely, using a seperate PC, i.e., rtpgenerator.py --ip=x.x.x.x --proxy=y.y.y.y --count=50, where x.x.x.x is rptgenerator's ip and y.y.y.y is mediaproxy's ip, I noticed that the load is mainly taken by CPU0. Can someone tell me what happenned? Has anyone encountered this? If it is python problem, then why it can distribute the load evenly when stream locally? Kernel Problem?? Code??? Thanks for any help available. Regards, TC Chan

19 years

[Serusers] UPDATE instead of re-INVITE and RTPProxy

by Ronald Voermans

Hello all, I have a problem with some user-agents (namely Grandstream and Cisco ATA 18X). My SIP provider provides a UPDATE instead of a re-INVITE, because the mentioned user-agents support the UPDATE method. However, SER/RTPProxy doesn't know how to handle this UPDATE request, and forwards the request to the user-agent. Result: one-way-voice. Does someone know how to solve this. I solved it for now by replacing the UPDATE in the request, so the SIP provider doesn't see that my user-agents supports the UPDATE method. Thanks in advance, Regards / Met vriendelijke groet, --------------------------------------------- R.L.L.M. Voermans Manager Network Connectivity Intern Global-e Raadhuisstraat 32 5126 CJ Gilze (NL) T: +31-(0)161-888888 F: +31-(0)161-888899 E: r.voermans(a)global-e.nl <mailto:r.voermans@global-e.nl> W: http://www.global-e.nl <http://www.global-e.nl/> ---------------------------------------------

19 years

Re: [Serusers] mediaproxy for incoming

by Greger V. Teigre

You need in ser.cf both: alias="mydomain.com" alias="server.mydomain.com" g-) ----- Original Message ----- From: "George Lambson" <LambsonGE(a)mtc.byu.edu> To: <greger(a)teigre.com> Cc: <serusers(a)lists.iptel.org> Sent: Sunday, November 13, 2005 1:50 AM Subject: Re: [Serusers] mediaproxy for incoming > Any idea what I might be doing wrong? > > I have had some issues with setting up the served domain. I use x-lite for > my SIP UAs and have to put the host name of my SER server to get them to > log in, even though I set the environment variable for the SIP domain of > the server to my domain. Do you think that it might be related? > > Thanks, > George > >>>> "Greger V. Teigre" <greger(a)teigre.com> 11/11/2005 12:19:59 AM >>> > I believe the standard NAT traversal configs from onsip.org should handle > that. > g-) > ----- Original Message ----- > From: "George Lambson" <LambsonGE(a)mtc.byu.edu> > To: <serusers(a)lists.iptel.org> > Sent: Friday, November 11, 2005 12:51 AM > Subject: [Serusers] mediaproxy for incoming > > >> Is anyone using mediaproxy for INCOMING calls? >> >> What I mean is: is SER capable of accepting INVITE messages from an >> unregistered UA that is behind a NAT Firewall and connecting them with a >> registered UA? >> >> I would like to be able to receive incoming calls for my domain from >> anyone on the internet and connect them to the proper user. Please tell >> me >> anyone if you are successfully doing this. >> >> Thanks, >> George >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

19 years

[Serusers] Usrloc question

by Darren Nay

Does anyone have any advice for me on issue? Thanks for your help! Darren Nay "Darren Nay" wrote: > > Hey all, > > > > I have a question regarding usrloc. I have run into a problem.. > > > > We have static routes sent to an asterisk server for all of our SIP > usernames. In addition our IAD’s will register with the same > username, so that calls coming into our switch for that username will > be routed to both the asterisk box and the SIP IAD. This way, > whichever endpoint (IAD or asterisk) answers the call first will take > the call. > > > > For example. > > > > root:/ # serctl ul show +18646404810 > > <sip:+18646404810@192.168.1.60>;q=1;expires=-1012151 > > <sip:+18646404810@192.168.1.157:5060>;q=;expires=403 > > > > 192.168.1.60 is the asterisk server. This is a static route added by > serctl. > > 192.168.1.157 is my IAD which registers with the switch every 10 minutes. > > So when calls are made to (864) 640-4810 then SER will send an INVITE > to both location. > > > > I explained all of this just to explain now what my problem is, and > ask if anyone may know a possible solution. > > > > Now, we also use asterisk to perform call fwd’ing functions. Asterisk > will answer the call and then originate another call out back to SER > to a new location. Now the problem! (finally!) This call fwd’ing > method works very well in most cases, except that if the call fwd’ing > is being sent to another location registered with SER then it will be > redirected back to asterisk again, albeit to a different URI, and > asterisk will kill the call because it thinks that it has looped > (which I guess it has… sort of). > > > > So, I’m wondering if there is possibly a way to retrieve only the > usrloc locations that don’t contain the IP address 192.168.1.60 in the > contact URI? This way I can just check if the src_ip is 192.168.1.60 > and if so then retrieve all the usrloc locations – without asterisk – > and the call will not be redirected back to asterisk. > > > > Is this possible? Or if anyone has any other ideas that may help then > I am definitely open to suggestions. > > > > Thanks for your help!! > > > > Darren Nay > > Ionosphere, Inc > > VoIP Network Development > > dnay(a)ionosphere.net <mailto:dnay@ionosphere.net> > > >

19 years

[Serusers] Need help: SER and MYSQL issue: PID file /var/run/ser.pid does not exist

by Dhiman D Chowdhury

Hi all, I need help to fix the following issue which only occurs when I am loading mysql.so. The problem is that serctl will fail to start SER and sometimes if SER start with service SER command. SER process is not found. I have observed that for all those issues, the cause is as system log as follows: Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: Maxfwd module- initializing Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: new_connection(): Client does not support authentication protocol requested by server; consider upgrading MySQL client Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: db_init(): Could not create a connection Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: ERROR: uridb_db_ver: unable to open database connection Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: ERROR: uri_db:mod_init(): Error while querying table version Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: init_mod(): Error while initializing module uri_db When those situation occur SER always unable to start, please see this message: Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed I think it is to do with mysql connection. I will appreciate if someone, please help me how to fix this. I am sure a number of other people will be facing this issue as well. Please help !! Please Help! Thanks Dhiman

19 years

[Serusers] Need help: SER and MYSQL issue: PID file /var/run/ser.pid does not exist

by Dhiman D Chowdhury

19 years

[Serusers] mediaproxy module: RTP packet mapping after reINVITE with new SDP

by Joel Rosenfield

As I am new to this list, please tell me if there is a more appropriate list to which I should post this question. Does the mediaproxy handle changes in the media description within the same call dialog? If not, are there plans to do so? Here is what I have done: I am using the SER mediaproxy module v1.4.2 and a co-located proxydispatcher and mediaproxy to accomplish NAT traversal between a 3PCC and an remote ATA. All basic call scenarios work fine. However, when the module receives a reINVITE to reconnect the ATA to a different party, the mediaserver fails to forward the RTP packets between the ATA and party described in the new SDP. I saw in the mediaproxy logs that RTP packets had been received from both sides, but the signIn method was being invoked as "called" for both parties. I explored the rtphandler.py and saw that the "catchall" clause at the end of RTPStream.handle_read was being invoked as you can see by the debug statement that I added to the code. It appears that this is invoked because both parties had already signed in based upon the previous media description, so none of the other mapping/forwarding clauses are invoked. It seems that there needs to be a mechanism to clear the mappings when the mediaproxy receives a "request" command for the same session (based upon the SIP Call-ID) but a new SDP. I got it work by adding code to the beginning of RTPSession.updateStreams() that clears the list "mediaStreams" and letting the rest of the code in that method recreate the mappings from scratch. See the attached file containing the code, a mediaproxy log before the change, and a log from after the change. Although this works, the solution is a bit incomplete since it can cause brief interruptions in the RTP stream when a reINVITE is received that does not include an SDP change, etc. Please let me know if there is something I am missing with respect to how I should using the mediaproxy. Thanks, - Joel Rosenfield __________________________________ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com

19 years

[Users] How to gaurd PSTN gateway with openser + NAT

by Lokesh Kumar

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 >

19 years

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sr-users November 2005