sr-users August 2006

sr-users@lists.kamailio.org

184 participants
361 discussions

Re: [Users] force_rtp_proxy() and m=video
by Atle Samuelsen 25 Aug '06

25 Aug '06

HI Adam, The reason for this is 2 reasons.. the nathelper module does not understand the port/1 parameter (wich says "use 2 ports, port 5010 and 5011) and it does also not understand the RTP/SAVPF. If you fix these 2 things, this should work :-) (Is this a Ojo?) -A In-Reply-To: <029301c6c82a$68a8c370$1601a8c0@oralnetws2> * Adam Linford <adam.linford(a)oralnet.co.uk> [060825 11:40]: > I've been happily using the force_rtp_proxy() function, exported from the > nathelper module for some time now. Recently, I've added some videophone > UA's to our system, and we are finding that rtpproxy is only relaying the > audio for sessions between video phones. > > Having ran some SIP traces I've noticed that the SDP is not being correctly > rewritten. Whilst the contact header for the audio is being allocated a > port from rtpproxy's range correctly, the corresponding m= field for the > video is not being rewritten at all. I've included an excerpt of the SDP I > am seeing below. > > v=0 > o=1000 16724 16724 IN IP4 192.168.1.66 > s=videophone > c=IN IP4 openser.ip.address.here > t=0 0 > a=sendrecv > m=audio 35030 RTP/AVP 98 0 8 > a=ptime:20 > a=rtpmap:98 iLBC/8000 > a=fmtp:98 mode=20 > m=video 5010/1 RTP/SAVPF 99 > b=AS:110 > a=rtpmap:99 H264/90000 > a=fmtp:99 > packetization-mode=1;parameter-add=0;sprop-parameter-sets=J0KgHpWgsTk=,KM4Ec > Q== > a=key-mgmt:key CnNnTdBxKycXpOh/Z6VPdeF1/u3+cVcQzAJAN4dWuGM > a=direction:active > > RTPproxy's UDP port range is 35000-65000, and as you can see, the m=audio > field is rewritten by calling force_rtp_proxy(), with the port value of > 35030, but the m=video field retains the port that the UA defines as its > starting UDP port for media. > > This is the first time I've dabbled with video and SIP, so was hoping > someone with more experience might already have hit this problem, before I > start delving into the code > > I'm running rtpproxy 0.3, and have also tried various other versions of > rtpproxy from cvs to no avail. > > Any help appreciated. > > Cheers, > Adam > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[Serusers] About Cisco 7960 an option Call Forwarding All
by Javier Oviedo 25 Aug '06

25 Aug '06

Hi all In my SIP environment I have several Cisco 7960. These telephones have the option "Call Forwarding All", when I use this option with ser, the telephone generete a 302 SIP Code response, but the sip proxy does not forward the call to the new destiny. Any Idea to solve this problems Thanks in advance Joe

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[Users] need to find out OUTGOING IF / SOCKET
by Gerry 25 Aug '06

25 Aug '06

Dear openser-users, I am trying to set-up a configuration where openser + RTPproxy are relaying messages between disconnected networks (i.e. openser + RTPproxy having more than one network-interface). With this method, I could additionally bridge IPv4-only and IPv6-only clients. Therefore I need to find out, on which interface (or socket, respectively) openser will send the message out because RTPproxy needs this info and is too dumb to find out by itself. I currently have a _very_ complicated solution which utilizes AVPs and mysql and it fails with NATed clients (detail: because the "contact" changes, I cannot look it up in the DB again). So, if you have any solutions or hints I would be VERY happy to "hear" from you. Greetings, Gerry. P.S.: Of course, I would be happy to share my config, but I don't want to SPAM this list. Therefore, if you are interested, just ask! But remember: my solution ONLY makes sense, if you're relaying between disconnected networks!

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[Users] force_rtp_proxy() and m=video
by Adam Linford 25 Aug '06

25 Aug '06

I've been happily using the force_rtp_proxy() function, exported from the nathelper module for some time now. Recently, I've added some videophone UA's to our system, and we are finding that rtpproxy is only relaying the audio for sessions between video phones. Having ran some SIP traces I've noticed that the SDP is not being correctly rewritten. Whilst the contact header for the audio is being allocated a port from rtpproxy's range correctly, the corresponding m= field for the video is not being rewritten at all. I've included an excerpt of the SDP I am seeing below. v=0 o=1000 16724 16724 IN IP4 192.168.1.66 s=videophone c=IN IP4 openser.ip.address.here t=0 0 a=sendrecv m=audio 35030 RTP/AVP 98 0 8 a=ptime:20 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 m=video 5010/1 RTP/SAVPF 99 b=AS:110 a=rtpmap:99 H264/90000 a=fmtp:99 packetization-mode=1;parameter-add=0;sprop-parameter-sets=J0KgHpWgsTk=,KM4Ec Q== a=key-mgmt:key CnNnTdBxKycXpOh/Z6VPdeF1/u3+cVcQzAJAN4dWuGM a=direction:active RTPproxy's UDP port range is 35000-65000, and as you can see, the m=audio field is rewritten by calling force_rtp_proxy(), with the port value of 35030, but the m=video field retains the port that the UA defines as its starting UDP port for media. This is the first time I've dabbled with video and SIP, so was hoping someone with more experience might already have hit this problem, before I start delving into the code I'm running rtpproxy 0.3, and have also tried various other versions of rtpproxy from cvs to no avail. Any help appreciated. Cheers, Adam

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[Serusers] Asterisk t38passthrough
by Ricardo Carvalho 25 Aug '06

25 Aug '06

Hi, I've installed Asterisk t38passthrough branch and I'm using one Grandstream ATA to connect Asterisk to a Fax machine. Every time I send a fax, it gets sent using codec G711, and never T.38. I added the following parameters in the [general] section as well as in device configurations: t38pt_udptl = yes t38pt_rtp = yes t38pt_tcp = yes I think that's the only thing that is needed to do to enable T.38 pass through... Why does Asterisk keeps sending in G711? Any help? Regards, Ricardo.

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[Serusers] How to fix IP address stored in location db
by İlker Aktuna (Koç.net) 25 Aug '06

25 Aug '06

Hi, I am trying to change stored uri in the location db. I assume it could be done with the help of nathelper module and I use fix_nated_register() function just before save_noreply("location") command. However I still see the internal IP address of the client in the location db. What is the easiest way to change the stored contact address ? Thanks, ilker _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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Re: [Serusers] Voice Mail problem
by Greger V. Teigre 25 Aug '06

25 Aug '06

Please copy serusers. It's your failure route you should look at! g-) ram wrote: > Hi > > iam following the same document only to make my voice mail work > below is my config, its works if the user is registered locally > and if not available it will moving to voicemail > > But when iam dialing PSTN still its going to voicemail > > Aug 25 11:05:19 cdrouter ser[9337]: **************** vm start - begin > ****************** > Aug 25 11:05:19 cdrouter Sems[802]: Error: 404 voicemail: no email > address for user <1XXXXXX360> > Aug 25 11:05:19 cdrouter ser[9348]: ACC: call missed: from=99999 < > sip:99999@domain.com <mailto:sip:99999@domain.com>>;tag=2858293102, > i-uri=sip:1XXXXXXX360@domain.com <mailto:sip:1XXXXXXX360@domain.com>, > method=INVITE, o-uri= sip:1XXXXXX360@domain.com > <mailto:sip:1XXXXXX360@domain.com>, code=404 Not Found > Aug 25 11:05:19 cdrouter ser[9337]: **************** vm start - end > ****************** > > > ------------ > if (!lookup("location")) { > > # Voicemail specific configuration - begin > > if(method=="ACK" || method=="INVITE" || > method=="BYE"){ > > setflag(2); > if(t_newtran()){ > > t_reply("100","Trying -- just wait a > minute !"); > > if(method=="INVITE"){ > log(1,"**************** vm > start - begin ******************\n"); > > if(!vm("/tmp/am_fifo","voicemail")){ > log("could not contact the > answer machine\n"); > t_reply("500","could not > contact the answer machine"); > }; > log(1,"**************** vm start - end > ******************\n"); > break; > }; > > if(method=="BYE"){ > log(1,"**************** vm end - begin > ******************\n"); > if(!vm("/tmp/am_fifo","bye")){ > log("could not contact the > answer machine\n"); > t_reply("500","could not > contact the answer machine"); > }; > log(1,"**************** vm end - end > ******************\n"); > break; > }; > } > else { > log("could not create new transaction\n"); > sl_send_reply("500","could not create new > transaction"); > }; > > > /* Not found*/ > sl_send_reply("404","Not Found"); > }; > # Voicemail specific configuration - end > }; > > > > > On 8/24/06, *Greger V. Teigre* <greger(a)teigre.com > <mailto:greger@teigre.com>> wrote: > > Look at the features-callfwd.cfg example from onsip.org > <http://onsip.org/>, now available here: > http://www.iptel.org/ser/doc/gettingstarted > g-) > > ram wrote: > Hi > > I have existing SER Running > > i have local users ( extensions) and DID ( all other countries) > > iam able to make calls in and out , there is no problem > > but when the user not available, it has to send to Voice mail, > but its only sending the users > who are locally start with 9*, > > how can i send the user start with 9 or any DID not availble > send to voicemail > > here is my config > > > if (!lookup("location") && > ( uri=~"^sip:9.*@domain.com <mailto:9.*@domain.com>" ) > ) { > > # Voicemail specific configuration - begin > > if(method=="ACK" || method=="INVITE" || > method=="BYE"){ > > setflag(2); > if(t_newtran()){ > > t_reply("100","Trying -- just wait > a minute !"); > > if(method=="INVITE"){ > log(1,"**************** vm > start - begin ******************\n"); > > if(!vm("/tmp/am_fifo","voicemail")){ > log("could not contact the > answer machine\n"); > t_reply("500","could not > contact the answer machine"); > }; > log(1,"**************** vm start - > end ******************\n"); > break; > }; > > if(method=="BYE"){ > log(1,"**************** vm end - > begin ******************\n"); > if(!vm("/tmp/am_fifo","bye")){ > log("could not contact the > answer machine\n"); > t_reply("500","could not > contact the answer machine"); > }; > log(1,"**************** vm end - > end ******************\n"); > break; > }; > } > else { > log("could not create new transaction\n"); > sl_send_reply("500","could not create new > transaction"); > }; > > break; > }; > break; > }; > > > > any help will be great > > > > Ram > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers > > >

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[Users] Internet Providers changing the SIP headers
by Bruno Machado 25 Aug '06

25 Aug '06

Hi fellows. We are in trouble with this "deceitful" error. I mean, errors. Look at RURI. Our proxy receive this "thing". We have a number there! Sometimes, this number is shown within the word REGISTER, like that: "REGI:34846:STER". We can rewrite the URI, but Openser doesnt understand the method "REGI:34846:STER". Second error: Via header. The REGISTER is showing the NAT_DEVICE_IP instead UA_IP. And this VoIP isnt behind any VoIP device! So, "who" are changing it? "Who" can does it? Is possible that some "device" between UA and our proxy change the headers intentionaly? This kind of problem had been happening some months ago and began at the same day to VoIPs were different regions of the country (Brazil). We would like to know with someone already has face it. Thank you all. Bruno Machado U NAT_DEVICE_IP:50764 -> PROXY_IP:5060 REGISTER :50764sip:proxy.com.br SIP/2.0 Via: SIP/2.0/UDP NAT_DEVICE_IP:50765;branch=z9hG4bK427600003ca00000 From: <sip:7011423@proxy.com.br>;tag=ab510000bf39ffff To: <sip:7011423@proxy.com.br> Contact: * Call-ID: 499effff9ff20000(a)192.168.1.5 CSeq: 100 REGISTER Expires: 0 User-Agent: Grandstream HT386 1.0.3.17 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 U PROXY_IP:5060 -> NAT_DEVICE_IP:50765 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP NAT_DEVICE_IP:50765;branch=z9hG4bK427600003ca00000 From: <sip:7011423@proxy.com.br>;tag=ab510000bf39ffff To: <sip:7011423@proxy.com.br>;tag=22aae43fbf528d3860ded8429db379db.546f..Call-ID: 499effff9ff20000(a)192.168.1.5 CSeq: 100 REGISTER Proxy-Authenticate: Digest realm="proxy.com.br", nonce="44edfeda1fe71fa609e7599bf2513e999e3741dd" Content-Length: 0 --------------------------------- Novidade no Yahoo! Mail: receba alertas de novas mensagens no seu celular. Registre seu aparelho agora!

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[Users] How to configure openser as a redirect server
by Padmaja RV 25 Aug '06

25 Aug '06

Hi all, I need to implement Openser as a redirect server for all the 3XX class of responses. I have checked the redirect.cfg example in the cvs and it includes only for the 300 response. The other two like 302 moved temporarily or 301 moved permanently are not included anywhere. Can anybody send me the script for the same? Thank You, Padmaja

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[Serusers] maximum number of routes?
by Jean-Michel Hiver 25 Aug '06

25 Aug '06

Hi, I have found that ser yields an error when using up to 20 routes (i.e. route[20] block) and openser 40 routes. Is there an option to increase the number of routes ser can handle? Cheers, Jean-Michel.

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sr-users August 2006