sr-users April 2006

sr-users@lists.kamailio.org

188 participants
431 discussions

[Users] changing SDP session and media attributes in 200 OK

by Antanas Masevicius

Hello, i am using rtpproxy with openser to act as a RTP bridge to IP-SS7 gateway. IP-SS7 GW should not be exposed to external clients directly. Architecture diagram is like this: |UA1|--SIP/RTP---| |------------------| |-----------| | OpenSer+RTPProxy | --SIP/RTP-- | IP-SS7-GW | |------------------| |-----------| |UA2|--SIP/RTP---| Problem is when i try to send fax RTP packets destined for UA2 from IP-SS7 GW are sent directly. They should go through RTPProxy, otherwise media ports does not match. onreply route looks like: onreply_route[1] { if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); } force_rtp_proxy(); } each request is record_route()'ed before it is sent to be routed back through proxy. I noticed that OpenSer removes top most via header in 200 OK messages: 200 OK message for re-Invite received by OpenSer: SIP/2.0 200 OK\r\n From: <sip:870000049@sp2.ntt.lt>;tag=4fbb0552\r\n To: <sip:70000050@sp2.ntt.lt>;tag=0-13c4-444c9fc6-f4eac-fdd\r\n Call-ID: cc36b0-0-13c4-444c9fc6-f4eac-2c87(a)sp2.ntt.lt\r\n CSeq: 666497904 INVITE\r\n Via: SIP/2.0/UDP 84.32.0.18;branch=z9hG4bKf67f.2aeace41.0\r\n Via: SIP/2.0/UDP 84.32.0.7:5060;branch=z9hG4bK444cbbfca6214604dc9\r\n Supported: 100rel\r\n User-Agent: SIP-RG\r\n Contact: <sip:70000050@84.32.134.143>\r\n Content-Type: application/sdp\r\n Content-Length: 317\r\n \r\n v=0\r\n o=SIP-N-TA 0 0 IN IP4 84.32.134.143\r\n s=5420868F 0000 5420868F 0000 01 0\r\n c=IN IP4 84.32.134.143\r\n t=0 0\r\n m=image 4008 udptl t38\r\n c=IN IP4 84.32.134.143\r\n a=T38FaxVersion:0\r\n a=T38MaxBitRate:14400\r\n a=T38FaxFillBitRemoval:0\r\n a=T38FaxTranscodingMMR:0\r\n a=T38FaxTranscodingJBIG:0\r\n a=T38FaxRateManagement:transferredTCF\r\n 200 OK message sent to IP-SS7 GW by OpenSer: SIP/2.0 200 OK From: <sip:870000049@sp2.ntt.lt>;tag=4fbb0552 To: <sip:70000050@sp2.ntt.lt>;tag=0-13c4-444c9fc6-f4eac-fdd Call-ID: cc36b0-0-13c4-444c9fc6-f4eac-2c87(a)sp2.ntt.lt CSeq: 666497904 INVITE Via: SIP/2.0/UDP 84.32.0.7:5060;branch=z9hG4bK444cbbfca6214604dc9 Supported: 100rel User-Agent: SIP-RG Contact: <sip:70000050@84.32.134.143> Content-Type: application/sdp Content-Length: 333 v=0 o=SIP-N-TA 0 0 IN IP4 84.32.134.143 s=5420868F 0000 5420868F 0000 01 0 c=IN IP4 84.32.134.143 t=0 0 m=image 39246 udptl t38 c=IN IP4 84.32.0.18 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=nortpproxy:yes In this message there is no indication that is came from SIP proxy (no record-route, no via header), then IP-SS7 GW sends ACK message directly. This, i suppose, is normal thing, is it? Problem is that IP-SS7 also sends RTP traffic to UA2 directly. This probably happens because of two 'c=' fields in SDP. OpenSer changes only one in media part. Is there a way to change 'c=' field in SDP not only on media part? Which should take precedence if both can't be changed? regards, Antanas Masevicius

18 years, 7 months

[Users] question about adding users

by B Boc

I am now trying to "add users" but i don't understand why username requires domain name and the implication down the road. if i issue the command: openserctl add 2002 spa3k user2(a)mysipserver.com I get the response: "domain unknown: use usernames with domain or set default domain in SIP_DOMAIN" why does it expect my username to be 2002(a)mysipserver.com? does this therefore mean the UAs will have to have their username field set to username@domain instead of just username. What's the relevance of capturing the email address?

18 years, 7 months

[Users] awkward loose_route behaviour

by Jean-Michel Foucher

Hello all, i'm facing some problems with loose_route, may it be with ACK or BYE messages. Here is the situation : - 2 softphones : let's call them A and B (on different IP addresses : /A_ip /and/ B_ip/). A's username is "voip2". A registers on this platform : - a stateless SER on /stateless_ip/ - a statefull SER on /statefull_ip/ B registers on another platform that i do not know (see /Bplatform_ip/) When B calls A, INVITE messages are smoothly forwarded and relayed. Same case for trying, OK and 180 ringing. However, when B sends ACK after (INVITE, trying, OK), here is what ser stateless receives : ACK sip:voip2@/stateless_ip/ SIP/2.0 Record-Route: <sip:voip2@/Bplatform_ip/;ftag=368e6f71;lr=on> Via: SIP/2.0/UDP /Bplatform_ip/;branch=0 Via: SIP/2.0/UDP /B_ip/:64064;received=/B_public_ip/;branch=z9hG4bK-d87543-226b496c2343f701-1--d87543-;rport=2301 Max-Forwards: 69 Route: <sip:/stateless_ip/;lr=on> (****) Route: <sip:/statefull_ip/;ftag=368e6f71;lr=on> (****) Route: <sip:/stateless_ip/;lr=on> this message is loose_routed and then forwarded (loose_route(); forward(uri:host, uri:port);) and then here is what ser stateless sends to ser statefull : ACK sip:/stateless_ip/;lr=on SIP/2.0 Record-Route: <sip:voip2@/Bplatform_ip/;ftag=368e6f71;lr=on> Via: SIP/2.0/UDP /stateless_ip/;branch=0 Via: SIP/2.0/UDP /Bplatform_ip/;branch=0 Via: SIP/2.0/UDP /B_ip/:64064;received=/B_public_ip/;branch=z9hG4bK-d87543-226b496c2343f701-1--d87543-;rport=2301 Max-Forwards: 16 Route: <sip:/stateless_ip/;lr=on> Here are my parameters for ser stateless concerning rr module : modparam("rr", "enable_full_lr", 1) modparam("rr", "enable_double_rr", 0) modparam("rr","append_fromtag",0) Then, there are two things that i don't understand and make confusion to ser statefull: - why did (****) Routes disappear ? - why did ruri changed in ACK message sent to ser statefull? any help would be appreciated, thanks in advance, -- Jean-Michel Foucher OpenWengo, the free and multiplatform VoIP client http://dev.openwengo.com/

18 years, 7 months

Re: [Users] Problem with openser.cfg

by Klaus Darilion

li gh wrote: > Thank you! > The local user is 1001@proxy1 <mailto:1001@proxy1>, calling > sip:2001@proxy2, > so the SIP signaling should be relayed to proxy2 by proxy1, > but I cannot capture any SIP packet from proxy1 to proxy2. Put log statements into openser.cfg and watch them in syslog. This will give you details what is going wrong. make sure proxy1 is able to resolve "proxy2" to the corresponding IP address (DNS). > When I call another user also registered to proxy1 > (e.g. sip:1004@proxy1), there is no problem. > > How to "start it manually" ? Just type "openser stop" then "openser start"? > Or by other ways? I have no idea. just "openser" as root will start openser. Then you can use "ps -Alf|grep openser" to find out the PID of the openser processes. To stop openser manually, find the lowest PID and kill openser with "kill PID" (replace PID with the value) regards klaus > > Thank you very much! > Best regards. > > On 4/24/06, *Klaus Darilion* <klaus.mailinglists(a)pernau.at > <mailto:klaus.mailinglists@pernau.at>> wrote: > > How do you call? sip:2001@proxy2 ? or sip:2001@proxy1? > > Use "ngrep port 5060" to debug SIP signaling > > You have to start it manually (no startup scripts). > > Better set log_stderror to "no" and watch the log messages in syslog. > > e.g. on debian syslog is senet to /var/log/messages. Some Linus > distributions use other file names but it should be easily to figure > that out: > > tail -f /var/log/messages > > regards > klaus > >

18 years, 7 months

[Serusers] debug level

by Allan Chao \( 策略性新技術開發處申請 \)

Dear all: debug parameter in the ser.cfg has nine levels, anybody knows what means for each one of them, thanks allan DISCLAIMER: Sample Disclaimer added in a VBScript.

18 years, 7 months

[Serusers] Creating Database

by openser＠comcast.net

Error message during creating database. Anyone seen this before [root@localhost ~]# /usr/local/sbin/openser_mysql.sh create MySql password for root: Domain (realm) for the default user 'admin': localhost.localdomain creating database openser ... ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2)

18 years, 7 months

[Users] Mediaproxy problems

by Sam Lee

Hi all, I have been getting this error for some of the calls made from mediaproxy, error: use_media_proxy(): empty response from mediaproxy Any idea what is it ? It appears when i tried to do a use_media_proxy() What does it means ? Regards, Sam

18 years, 7 months

[Users] ACK is answered after the INVITE. But ACK arrived first!

by Bruno Machado

Hi all Im in trouble with this problem: I have a Openser the connect to a carrier. I need that this proxy answer the messages in order that it receive. In my situation, other device one hope behind it, sends an ACK and, miliseconds after, it sends an INVITE to change the session. The server where the Openser is installed receive the ACK first than INVITE, but it ALWAYS forward this INVITE first. This way, I cant connect my proxy to the carrier. I need to know how I can change this behavior: the SIP proxy needs to send the ACK first. First in, first out..... Anybody knows how to do it? Thanks for any suggest Bruno Machado --------------------------------- Abra sua conta no Yahoo! Mail - 1GB de espaço, alertas de e-mail no celular e anti-spam realmente eficaz.

18 years, 7 months

[Users] cdrtool and radius

by Mike Williams

Has anyone been using CDRtool? It seems quite comprehensive, and it would be nice to get away from another custom written component. However, it seems that it only works through RADIUS on top of MySQL. Is this right? Is there any way to use it without RADIUS, and just with the accounting information stored in MySQL by the acc module in OpenSER? ---Mike

18 years, 7 months

[Serusers] SERWEB installation and configuration....

by fay butt

Hi all, I am facing problem in serweb configuration. I installed the serweb but when i type http://localhost/index.php in the browser, it gives the messge "The requested URL /user was not found on this server." the path of the serweb directory is like : /~serweb also in : /var/www/html/serweb please help me how can i find that page...my apache is runnig fi9 and if makes a file in the /var/www/html/page.php it works fi9....plz help me --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

18 years, 7 months

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sr-users April 2006