sr-users June 2007

sr-users@lists.kamailio.org

191 participants
314 discussions

by Szasz Szabolcs

Hi! I know the prefix() and the strip() functions, but how can I change the from_uri (caller)? Thanks for the help. > > ----- Original Message ----- > From: "SIP" <sip(a)arcdiv.com> > To: "Szasz Szabolcs" <sszasz(a)digicomm.ro> > Sent: Monday, June 25, 2007 12:34 AM > Subject: Re: [Serusers] add prefix to source > > >> Use the prefix() function. >> >> i.e. to add the prefix 555 onto the beginning of a URI, >> >> prefix("555"); >> >> N. >> >> >> Szasz Szabolcs wrote: >>> Hi all >>> I need to add a prefix to the source uri/number. >>> How ca I do this? Can you help me guys? >>> Thanks a lot >>> Szasz Szabolcs >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> -- >> This message was scanned for spam and viruses by BitDefender. >> For more information please visit http://linux.bitdefender.com/ >> >> > -- This message was scanned for spam and viruses by BitDefender. For more information please visit http://linux.bitdefender.com/

16 years, 10 months

[Serusers] Error 483 response to ACK and BYE

by SIP

In response to a BYE and an ACK from some (not all) servers that send to our proxy, our server's spitting back a 483 'Too many hops' reply. It doesn't happen with all peers, and it certainly doesn't happen locally... but it does happen with some, and I'm not entirely sure as to why. The 483 block at the beginning is normal: if (msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; Exchange looks like this: U 198.65.166.131:5060 -> 63.64.65.66:5060 BYE sip:1101XXXXXXX@63.64.65.66:5060 SIP/2.0. Record-Route: <sip:198.65.166.131;ftag=gpp0q468oh;lr>. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK9b05.b75fbe14.0. Via: SIP/2.0/UDP 192.168.1.51:2054;received=63.XX.XX.XX;branch=z9hG4bK-wdtud2nffq88;rport=2054. Route: <sip:63.64.65.66;ftag=gpp0q468oh;lr=on>. From: "User One" <sip:1747XXXXXXX@their.proxy.server:5060>;tag=gpp0q468oh. To: <sip:1101XXXXXXX@their.proxy.server;user=phone>;tag=as7fbbcb66. Call-ID: 3c267038e7ef-586eg1emcfsi@snom190. CSeq: 2 BYE. Max-Forwards: 16. Contact: <sip:1747XXXXXXX@192.168.1.51:2054;line=oxd9zlst;nat=yes>. User-Agent: snom190/3.60x. Content-Length: 0. RemoteIP: 63.XX.XX.XX. P-hint: rr-enforced. P-NATed-URI: YES (1). P-RTP-Proxy: YES (1). U 63.64.65.66:5060 -> 198.65.166.131:5060 SIP/2.0 483 Too Many Hops. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK9b05.b75fbe14.0. Via: SIP/2.0/UDP 192.168.1.51:2054;received=63.XX.XX.XX;branch=z9hG4bK-wdtud2nffq88;rport=2054. From: "User One" <sip:1747XXXXXXX@their.proxy.server:5060>;tag=gpp0q468oh. To: <sip:1101XXXXXXX@their.proxy.server;user=phone>;tag=as7fbbcb66. Call-ID: 3c267038e7ef-586eg1emcfsi@snom190. CSeq: 2 BYE. Server: Sip EXpress router (0.9.6 (i386/linux)). Content-Length: 0. Warning: 392 63.64.65.66:5060 "Noisy feedback tells: pid=3488 req_src_ip=63.64.65.66 req_src_port=5060 in_uri=sip:1101XXXXXXX@63.64.65.66:5060 out_uri=sip:1101XXXXXXX@63.64.65.66:5060 via_cnt==18". What would be causing this? Is it because the IP address is being used in the URI to us as opposed to the domain (I tried adding the IP to the domain table and to an alias line alternately, but it didn't fix things) ? It's irksome in that it only happens with certain peers and not others, so there's something in the way we're handling messages from them that's not right or different, but since we handle all incoming the same way, I'm at a loss as to why it works with some but not with others. N.

16 years, 10 months

[OpenSER-Users] Any demo for stateful proxy setting with 301/302 forwarded?

by Lucian Romi

Hi, I'm using openser basic config now. And I'm totally a freebie here. I want to set up a system with handle 301/302 forward for INVITE sending from openser, so it must be a stateful proxy and seems like there must be something in ser.cfg to support this. If don't, then openser will forward 301 and 302 all the way back, that's not I want. I'm really appreciate any help. Thanks!!!!

16 years, 10 months

[OpenSER-Users] Openser 1.2.1 with FreeRADIUS 1.1.6

by Neeraj Gupta

16 years, 10 months

[OpenSER-Users] NEW MODULE: Datagram implementation for MI

by Bogdan-Andrei Iancu

Hi, thanks to Ancuta Onofrei, we added today a new transport implementation for the Management Interface (MI). This is Datagram (via unix and network sockets) and it completes the migration to MI (after fifo), but not the last from the transport implementations :). It is in alpha stage, so more testing is needed. NOTE that all MI commands are available via datagram interface; also the syntax of the commands and replies is exactly as for FIFO. For more info, please see the online documentation: http://www.openser.org/docs/modules/1.2.x/mi_datagram.html or ask on the mailing list. Any feedback - bugs or improvements - are welcomed. Also I want to bring in discussion a related subject - dropping the old unix sockets implementation from the core. Before doing it I would like to know who is still using it and if there are any issues with migrating to the new implementation. Regards, Bogdan

16 years, 10 months

[OpenSER-Users] openser.cfg debug howto

by Rizwan Hisham

Hi all, I am a new user of openser.i just installed and configured it to send and recieve calls. im having some problems with scripting. Whenever there is a problem in openser.cfg, it just displays the number of errors. is there anyway to know at what line the error is and also what kind of error. As im a new user and so i dont know the syntax and symantics of the scripting very well, im not very good at finding the error by looking at the file. Hopefully this is possible -- Best Regards Rizwan Hisham Software Engineer AXVOICE Inc. www.axvoice.com

16 years, 10 months

[OpenSER-Users] Passing information/Headers

by Kelvin Williams

I need to pass information between OpenSER and Asterisk for a couple applications I have created. I figured the best way to do this would be to append a header field and have either OpenSER or Asterisk peel it off and take a look at what it says. However, I've found that Asterisk some times gets perturbed by the additional header field. Is it possible/wise to create your own headers? What other way could I pass information (used for scripts in Asterisk/OpenSER) between the two servers? Thanks, kw

16 years, 10 months

Re: [Serusers] How SER communicate to other SIP server

by roayu

Hi, Olaf! Ok, but which section that I need to concentrate on the most? I've search through the whole documentation and I can't find the configuration/setting on the ser.cfg file. Really appreciate on your help... Thanks. Best regards, Roa Yu -----Original Message----- From: Olaf Bergmann [mailto:Olaf.Bergmann@freenet-ag.de] Sent: Thursday, June 28, 2007 4:20 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server roayu wrote: > Hi, Olaf! > > Is there any settings that I need to do on SER or ser.cfg? Sure. At this point, you really should read <http://www.iptel.org/ser/doc/gettingstarted> before proceeding. Best regards, Olaf

16 years, 10 months

[OpenSER-Users] wengophone and openser

by Radu Spineanu

Hi, I have a working basic openser + asterisk setup. Twinkle works OK and wengophone doesn't. It tries to connect and then after a while times out. While debugging I noticed a lot of OPTIONS ping requests are going unanswered by openser because it contains an user in the uri. Registration seems to be working OK. I've attached the ngrep output that shows this. Has anyone got wengophone to work with openser? -- Radu Spineanu interface: eth0 (217.73.174.0/255.255.255.224) filter: (ip or ip6) and ( port 5060 ) # U 82.79.243.82:5060 -> 217.73.174.13:5060 REGISTER sip:217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=123456789..Route: <sip:217.73.174.13;lr>..From: nobo dy <sip:nobody@217.73.174.13>;tag=123456789..To: <sip:nobody@217.73.174.13>..Call-ID: 000001@ping..Contact: <sip:nobody@82.79.243.82>..C Seq: 2 REGISTER..Content-Length: 0.... # U 217.73.174.13:5060 -> 82.79.243.82:5060 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 82.79.243.82:5060;rport=5060;branch=123456789..From: nobody <sip:nobody@217.73.174.13>;tag=12 3456789..To: <sip:nobody@217.73.174.13>;tag=5e02e7658def22f683939a852e493344.3f4e..Call-ID: 000001@ping..CSeq: 2 REGISTER..WWW-Authentic ate: Digest realm="217.73.174.13", nonce="46823bd8e5f68588801e0655bbc907e2425ee03f"..Server: OpenSER (1.2.1-notls (x86_64/linux))..Conte nt-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 REGISTER sip:217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1094221745..Route: <sip:217.73.174.13:5060;lr >..From: <sip:1001@217.73.174.13>;tag=1065314925..To: <sip:1001@217.73.174.13>..Call-ID: 807187534@82.79.243.82..CSeq: 1 REGISTER..Conta ct: <sip:1001@82.79.243.82:5060>..Max-Forwards: 70..User-Agent: wengo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 2940..Content-Length: 0.... # U 217.73.174.13:5060 -> 82.79.243.82:5060 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 82.79.243.82:5060;rport=5060;branch=z9hG4bK1094221745..From: <sip:1001@217.73.174.13>;tag=106 5314925..To: <sip:1001@217.73.174.13>;tag=5e02e7658def22f683939a852e493344.4bcc..Call-ID: 807187534@82.79.243.82..CSeq: 1 REGISTER..WWW- Authenticate: Digest realm="217.73.174.13", nonce="46823bd9c467d1377ec65d343c7dfc7bd40513c3"..Server: OpenSER (1.2.1-notls (x86_64/linux ))..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 REGISTER sip:217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK2031845783;xxx-nat-type=sym..Route: <sip:217. 73.174.13:5060;lr>..From: <sip:1001@217.73.174.13>;tag=1065314925..To: <sip:1001@217.73.174.13>..Call-ID: 807187534@82.79.243.82..CSeq: 2 REGISTER..Contact: <sip:1001@82.79.243.82:5060>..Authorization: Digest username="1001", realm="217.73.174.13", nonce="46823bd9c467d137 7ec65d343c7dfc7bd40513c3", uri="sip:217.73.174.13", response="2e3d2c8bb188495863d4306f5cd5ba5c", algorithm=MD5..Max-Forwards: 70..User-A gent: wengo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 2940..Content-Length: 0.... # U 217.73.174.13:5060 -> 82.79.243.82:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 82.79.243.82:5060;rport=5060;branch=z9hG4bK2031845783;xxx-nat-type=sym..From: <sip:1001@217.73.174.13>; tag=1065314925..To: <sip:1001@217.73.174.13>;tag=5e02e7658def22f683939a852e493344.142c..Call-ID: 807187534@82.79.243.82..CSeq: 2 REGISTE R..Contact: <sip:1001@82.79.243.82:5060>;expires=2940..Server: OpenSER (1.2.1-notls (x86_64/linux))..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 OPTIONS sip:ping@217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1028071012..From: Radu <sip:1001@217.73.1 74.13>;tag=675779892..To: <sip:ping@217.73.174.13>..Call-ID: 1333741103@82.79.243.82..CSeq: 20 OPTIONS..Max-Forwards: 70..User-Agent: we ngo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 120..Accept: application/sdp..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 OPTIONS sip:ping@217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1028071012..From: Radu <sip:1001@217.73.1 74.13>;tag=675779892..To: <sip:ping@217.73.174.13>..Call-ID: 1333741103@82.79.243.82..CSeq: 20 OPTIONS..Max-Forwards: 70..User-Agent: we ngo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 120..Accept: application/sdp..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 OPTIONS sip:ping@217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1028071012..From: Radu <sip:1001@217.73.1 74.13>;tag=675779892..To: <sip:ping@217.73.174.13>..Call-ID: 1333741103@82.79.243.82..CSeq: 20 OPTIONS..Max-Forwards: 70..User-Agent: we ngo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 120..Accept: application/sdp..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 OPTIONS sip:ping@217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1028071012..From: Radu <sip:1001@217.73.1 74.13>;tag=675779892..To: <sip:ping@217.73.174.13>..Call-ID: 1333741103@82.79.243.82..CSeq: 20 OPTIONS..Max-Forwards: 70..User-Agent: we ngo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 120..Accept: application/sdp..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 OPTIONS sip:ping@217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1028071012..From: Radu <sip:1001@217.73.1 74.13>;tag=675779892..To: <sip:ping@217.73.174.13>..Call-ID: 1333741103@82.79.243.82..CSeq: 20 OPTIONS..Max-Forwards: 70..User-Agent: we ngo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 120..Accept: application/sdp..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 OPTIONS sip:ping@217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1028071012..From: Radu <sip:1001@217.73.1 74.13>;tag=675779892..To: <sip:ping@217.73.174.13>..Call-ID: 1333741103@82.79.243.82..CSeq: 20 OPTIONS..Max-Forwards: 70..User-Agent: we ngo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 120..Accept: application/sdp..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 OPTIONS sip:ping@217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1028071012..From: Radu <sip:1001@217.73.1 74.13>;tag=675779892..To: <sip:ping@217.73.174.13>..Call-ID: 1333741103@82.79.243.82..CSeq: 20 OPTIONS..Max-Forwards: 70..User-Agent: we ngo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 120..Accept: application/sdp..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 OPTIONS sip:ping@217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1028071012..From: Radu <sip:1001@217.73.1 74.13>;tag=675779892..To: <sip:ping@217.73.174.13>..Call-ID: 1333741103@82.79.243.82..CSeq: 20 OPTIONS..Max-Forwards: 70..User-Agent: we ngo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 120..Accept: application/sdp..Content-Length: 0.... exit 14 received, 0 dropped

16 years, 10 months

Re: [Serusers] How SER communicate to other SIP server

by roayu

Hi, Olaf! Is there any settings that I need to do on SER or ser.cfg? Thanks. Best Regards, Roa Yu -----Original Message----- From: Olaf Bergmann [mailto:Olaf.Bergmann@freenet-ag.de] Sent: Thursday, June 28, 2007 3:04 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server roayu wrote: > Hi, Olaf! > > Thanks for your reply. Could you please further explain on the connectivity > between those to DIFFERENT sip servers? If I wanna call using a softphone to > call from SER to SIPserver B, how should I make the call? In a typical scenario, you configure your softphone to use SER as outbound proxy (registrar, etc.) and let it route outgoing SIP requests according to its configuration. When calling foreign domains (i.e., the callee has a domain that is different from your softphone's), a simple t_relay would do. More elaborate routing policies could be based on numeric "dial plans" etc. to forward SIP requests according to your needs. Best regards, Olaf

16 years, 10 months

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sr-users June 2007