sr-users November 2003

sr-users@lists.kamailio.org

116 participants
267 discussions

by Darren Sessions

I am in the process of evaluating SER to replace my company's current SIP Proxy/GateKeeper. I've been racking my brain with the routing piece in the ser.cfg file along with a friend. Here are my variables: Router IP: 10.10.1.5 SER Sun Box IP: 10.10.0.55 Sipura IAD: 10.10.1.134 Sipura Box user ID is: 8646783188 Testing with G711alaw and G711ulaw but would like to move to g729a/b. Cisco Dialpeer is: dial-peer voice 1005 voip huntstop destination-pattern 3188 progress_ind progress enable 8 translate-outgoing called 1 session protocol sipv2 session target ipv4:10.10.0.55 codec g711alaw The translation-outgoing called 1 simply adds the 864678. No Authentication currently needed in testing phase. No DNS currently in use. Cisco 3640 Router connected to phone company with DID based T1 and the PBX is connected to the router. We currently have about 100+ DID's from our Bell company - because the T1 to bell is a DID line, we only get four digits on an inbound call. Essentially the router does nothing more than forward all calls to the PBX via a dial-peer - but strips off the specific phone numbers we want to forward to SIP based devices and pipes them out VoIP. In this whole process, the router adds the prefix 864678 to the dialed numbers so when the router sends the call to SER, it's 864678xxxx. Once SER gets that call (i.e. 8646783188(a)10.10.1.5) it should forward it to the Sipura Box. I'd post what I've got, but I'm not sure it'd even classify as a configuration. :) I am a newbie - so everyone is forewarned. Thanks a ton in advance, - Darren Sessions

20 years, 7 months

[Serusers] ser 0.8.12 released

by Andrei Pelinescu-Onciul

We have just released a new ser stable version: ser 0.8.12. This version fixes a lots of bugs, has some performance improvements (it's a little faster :-)) a.s.o (for more information read the attached NEWS file). The code is based on the cvs head from 14 Oct 2003. The following things which are on the current CVS head are not part of this release: - tcp aliases - automatic content-length updating (this means nathelper won't work as well as with the HEAD version, so for nat traversal stick to the unstable version) - latest permissions updates All 0.8.11 users which are using tcp a lot should upgrade, since 0.8.12 contains many tcp fixes which were not backported to 0.8.11-r1. Three new modules were added to the default build process and to the binary distribution: - auth_diameter (diameter based authentication with disc) - mangler (changes Contact and SDP ip addresses/ports, can be used in some nat traversal scenarios) - pdt (translates numerical codes into domains, based on prefixes, and updates the URI) This release was tested on: linux/x86, linux/arm, freebsd 4.5 & 5.1/x86, netbsd 1.6/sparc64, openbsd 3.2/x86, solaris 8/sparc64. The new release can be downloaded from: ftp://ftp.berlios.de/pub/ser/0.8.12 (source code, pre-compiled binaries for various arhitectures, packages for various operating systems and linux distributions; for more informations see ftp://ftp.berlios.de/pub/ser/0.8.11/README). You can also use cvs, just remember to add -r rel_0_8_12 to your cvs checkout command. Andrei

20 years, 7 months

[Serusers] FreeRadius and SER

by dhiraj.2.bhuyan＠bt.com

Greetings List, I am trying to configure FreeRadius to work with SER. When I try to test FreeRadius with - radiusd -X I get the error message - acct_unique: key = "User-Name, Acct-Session-Id, NAS-IP-Address, Client-IP-Address, NAS-Port-Id" rlm_acct_unique: Cannot find attribute 'Client-IP-Address' in dictionary radiusd.conf[1012]: acct_unique: Module instantiation failed. I checked the dictionary that comes with radiusclient and there is no attribute called "Client-IP-Address". I have appended the dictionary.ser from SER CVS to the radiusclient dictionary. Does anyone know what's the correct attribute? Also I noted that the "User-Password" (as documented by Jan Janak in the Radius howto) has changed to Password in the radiusclient dictionary. Thanks in advance, Dhiraj

20 years, 7 months

[Serusers] ser 0.8.11-r1 bugfix release

by Andrei Pelinescu-Onciul

We have also released an update to ser 0.8.11. It is available in source format only. You should use this only if you don't want to switch to 0.8.12. We stronlgy recommend upgrading to 0.8.12, it involves only one config change ( len_gt(number) must be replaced with msg:len > number) and includes lots of tcp fixes which were not backported to 0.8.11. We are not going to release any more updates to the 0.8.11 version. Download: ftp://ftp.berlios.de/pub/ser/0.8.11/src/ser-0.8.11-r1_src.tar.gz CVS branch: rel_0_8_11 Andrei

20 years, 7 months

[Serusers] Forwarding to more than one destination --

by Kannaiyan Natesan

I want to forward it to more than one SIP address, I used the following rule but not working. if( uri=~"^sip:12345@mydomain.com") { rewritehost("iptel.org"); forward(iptel.org,5060); rewritehost("fwd.pulver.com"); forward(fwd.pulver.com,5060); break; } I want to ring in multiple destinations when a single call arrives, whichever is first picked up, the other calls were cancelled. Am I missing anything here ? Kannaiyan

20 years, 7 months

[Serusers] Sample Routing Script

by Darren Nay

Hello All, Sorry for all the questions. I am new to this (both SIP and SER) and just learning as I go. Thanks so much to all who have helped me so far. I am just trying to setup some routing rules for our application and am not having any luck with getting them to work correctly. I have a Cisco Router which acts as the PSTN gateway. SER is installed on a Solaris box which is, of course, the SIP router, and then I have an SIP phone as the end point. The Cisco Router (PSTN gateway) is located at IP 10.10.1.5, the SER box is 10.10.0.55, and the SIP phone registers as 10.10.1.134. The telephone number assigned to the SIP phone is 864-678-3188. The SIP phone registers with the SER server without any problems, and all calls are routed through the Cisco router to the SER server (from what I can tell) as well. However, I don't know how to configure the routing correctly. All that I am trying to do right off the bat is to get that one SIP phone to work. My first attempt at writing a routing script looks like this (using a static route).. if (uri=~"sip:8646783188") { forward(10.10.1.134, 5060); }; I have tried many variations of this route but all to no avail.. :) Does anyone have a sample routing script that should work for this specific configuration? If I could see a sample then I think that I could build upon that. Any help would be very greatly appreciated! Regards, Darren Nay - dnay(a)libertyisp.com

20 years, 7 months

Re: [Serusers] sipsak

by Prince Arora

Hi On executing ldd sipsak libc.so.6 => /lib/i686/libc.so.6 (0x42000000) /lib/ld-linux.so.2 => /lib/ld-linux.so.2 (0x40000000) here i found that libc.so.6 is also in path /lib as well as in /lib/i686/. In both cases libc.so.6 maps to libc-2.2.5.so. In /lib size of libc-2.2.5.so is 1260480 in /lib/i686/ size of libc-2.2.5.so is 1401027. I copied libc-2.2.5.so from lib to i686 and take a backup of prev libc-2.2.5.so But problem remain same... no gethostname.... grep gethostbyname /lib/libc.so.6 gives Binary file /lib/libc.so.6 matches Shuld i deinstall glibc from my linux box and reinstall it ? what version of glibc should i used ? Any option to solve this problem. Regards Prince Nils Ohlmeier wrote: > Hello, > > this is very strange, because the tests from the configure script say, that > the gethostbyname function is present on your system. > > I can only imagine that the compiled sipsak binary uses a wrong lib. To verify > that please execute 'ldd sipsak'. Normaly the output should look like this: > libc.so.6 => /lib/libc.so.6 (0x4002c000) > /lib/ld-linux.so.2 => /lib/ld-linux.so.2 (0x40000000) > The command 'grep gethostbyname /lib/libc.so.6' (use the lib from the ldd > output above) should print the following: > Binary file /lib/libc.so.6 matches > If this does not match on your system, then you probably have a library > problem on your system, which i'm probably not able to solve from remove. > > Regards > Nils > > On Thursday 20 November 2003 04:50, you wrote: > > Hi Sir, > > > > netdb.h is present on my system in path /usr/include/ > > > > I am using sipsak-0.8.6 version. I am sending config.h and config .log as > > an attachment. > > > > I am using glibc-2.2.5-34. > > > > Regards > > Prince Arora > > > > Nils Ohlmeier wrote: > > > Hi, > > > > > > On Wednesday 19 November 2003 11:06, Prince Arora wrote: > > > > I have installed ser-0.8.11 on my redhat linux 7.3. SER is running > > > > fine. > > > > > > > > To check whether my server is responding to requests, i installed > > > > sipsack. > > > > > > > > My SIP_DOMAIN= "myserver.foobar.com". > > > > > > > > On giving command > > > > sipsak -T -s sip:admin@myserver.foobar.com. > > > > > > > > It shows following error. > > > > "no gethostbyname: No such file or directory" > > > > > > this is a little bit strange to me, because gethostbyname is a glibc > > > function. I'm wondering how you could compile it. Is the netdb.h header > > > file present on your system (usually at /usr/include)? > > > What version of sipsak do you use? > > > If you use the latest version (0.8.6) please send me (not to the list) > > > the config.h and config.log files. > > > > > > Greets > > > Nils

20 years, 7 months

RE: [Serusers] Record-Route Questions

by Garey, Tim

Thanks, Jan.... 1. When you say the BYE is missing the Route header field of ser, I'm not sure I follow. The BYE from our softswitch (packet 17) looks like this.. BYE sip:6037534033@10.16.1.218 SIP/2.0 Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1070017565 From: <sip:6032251114@10.16.1.147:5060>;tag=1069363200-4210481069784248-11 To: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b CSeq: 53 BYE Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 Max-Forwards: 10 Route: <sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr> Content-Length: 0 Is this part not correct? ==> Route: <sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr> -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Thursday, November 20, 2003 5:16 PM To: Garey, Tim Cc: 'serusers(a)lists.iptel.org' Subject: Re: [Serusers] Record-Route Questions Hello, there are two problems: 1) BYE sent by the softswitch is missing Route header field of ser. 2) You configuration script should contain t_relay() as the last command. Without that it will drop all requests that don't have IP or domain of SER server in the Request-URI which probably isn't what you want. BTW: The message below is an excellent problem report ! It contains all the necesarry information in clear form and it is quite easy to quickly locate the problem. Jan. On 20-11 16:52, Garey, Tim wrote: > Hello, > > Using ser 0.8.11 with redhat 9. I am not using mysql or authentication. > > The record-route call I'm making almost works, except for SER not passing > BYE's from our softswitch. I don't see why SER is blocking these. The ACK > is correctly following the record-route logic, the BYE looks like it is but > SER is > blocking passage. sip_scenario trace and ser.cfg routing logic are pasted > here > but the full sip_scenario file with packet contents are attached as a file. > > I don't see anything wrong with how the record-route and route headers are > being used. > Please help! > > Thank you VERY much !!! This group is terrific. > > Tim > > > Attached file has sip_scenario trace > > 10.16.1.218 6037534033 is SER registered phone who initiates call > 10.16.1.147 is SER > 172.17.4.102 is our softswitch acting as back2back user agent. > 10.16.1.214 6032251114 is softswitch registered phone answers/terminates > call. > > > 10.16.1.218 10.16.1.147 172.17.4.102 10.16.1.214 > | | | | > <Call><PFrame><Time> > | | | | > | F1 INVITE (sdp) | | | > |>---------------->| | | > | | | | > | ant to us 100 F2 | | | > |<----------------<| | | > | | | | > | | F3 INVITE (sdp) | | > | |>---------------->| | > | | | | > | | Trying 100 F4 | | > | |<----------------<| | > | | | | > | | | F5 INVITE (sdp) | > | | |>---------------->| > | | | | > | | | Ringing 180 F6 | > | | |<----------------<| > | | | | > | | Ringing 180 F7 | | > | |<----------------<| | > | | | | > | Ringing 180 F8 | | | > |<----------------<| | | > | | | | > | | | (sdp) OK 200 F9 | > | | |<----------------<| > | | | | > | | | F10 ACK | > | | |>---------------->| > | | | | > | | (sdp) OK 200 F11 | | > | |<----------------<| | > | | | | > | (sdp) OK 200 F12 | | | > |<----------------<| | | > | | | | > | F13 ACK | | | > |>---------------->| | | > | | | | > | | F14 ACK | | > | |>---------------->| | > | | | | > | | | BYE F15 | > | | |<----------------<| > | | | | > | | | F16 200 OK | > | | |>---------------->| > | | | | > | | BYE F17 | | > | |<----------------<| | > | | | | > | | BYE F18 | | > | |<----------------<| | > | | | | > | | BYE F19 | | > | |<----------------<| | > | | | | > | | BYE F20 | | > | |<----------------<| | > | | | | > | | BYE F21 | | > | |<----------------<| | > | | | | > | | BYE F22 | | > | |<----------------<| | > | | | | > | | BYE F23 | | > | |<----------------<| | > | | | | > | | BYE F24 | | > | |<----------------<| | > | | | | > | F25 BYE | | | > |>---------------->| | | > | | | | > | | F26 BYE | | > | |>---------------->| | > | | | | > | | OK 200 F27 | | > | |<----------------<| | > | | | | > | OK 200 F28 | | | > |<----------------<| | | > > > Here is routing logic: > > # main routing logic > > route{ > > # initial sanity checks -- messages with > # max_forwards==0, or excessively long requests > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > break; > }; > if (len_gt( max_len )) { > sl_send_reply("513", "Message too big"); > break; > }; > > if (method=="INVITE") { > rewritehostport("172.17.4.102"); > record_route(); > t_relay(); > # forward (172.17.4.102, 5060); > }; > > # we record-route all messages -- to make sure that > # subsequent messages will go through our proxy; that's > # particularly good if upstream and downstream entities > # use different transport protocol > #record_route(); > > # loose-route processing > if (loose_route()) { > t_relay(); > break; > }; > > # if the request is for other domain use UsrLoc > # (in case, it does not work, use the following command > # with proper names and addresses in it) > if (uri==myself) { > > if (method=="REGISTER") { > > # Uncomment this if you want to use digest authentication > # if (!www_authorize("iptel.org", "subscriber")) { > # www_challenge("iptel.org", "0"); > # break; > # }; > > save("location"); > break; > }; > > # native SIP destinations are handled using our USRLOC DB > if (!lookup("location")) { > sl_send_reply("404", "Not Found"); > break; > }; > }; > # forward to current uri now; use stateful forwarding; that > # works reliably even if we forward from TCP to UDP > #if (!t_relay()) { > # sl_reply_error(); > #}; > > } > > <<rr9.txt>> > > Sip Scenario Trace > > File: rr9.dump > Generated: Thu Nov 20 16:22:21 2003 > Traced on: Thu Nov 20 16:21:14 2003 > Created by:sip_scenario version=1.1.0 > > > 10.16.1.218 10.16.1.147 172.17.4.102 10.16.1.214 > | | | | <Call><PFrame><Time> > | | | | > | F1 INVITE (sdp) | | | > |>---------------->| | | > | | | | > | ant to us 100 F2 | | | > |<----------------<| | | > | | | | > | | F3 INVITE (sdp) | | > | |>---------------->| | > | | | | > | | Trying 100 F4 | | > | |<----------------<| | > | | | | > | | | F5 INVITE (sdp) | > | | |>---------------->| > | | | | > | | | Ringing 180 F6 | > | | |<----------------<| > | | | | > | | Ringing 180 F7 | | > | |<----------------<| | > | | | | > | Ringing 180 F8 | | | > |<----------------<| | | > | | | | > | | | (sdp) OK 200 F9 | > | | |<----------------<| > | | | | > | | | F10 ACK | > | | |>---------------->| > | | | | > | | (sdp) OK 200 F11 | | > | |<----------------<| | > | | | | > | (sdp) OK 200 F12 | | | > |<----------------<| | | > | | | | > | F13 ACK | | | > |>---------------->| | | > | | | | > | | F14 ACK | | > | |>---------------->| | > | | | | > | | | BYE F15 | > | | |<----------------<| > | | | | > | | | F16 200 OK | > | | |>---------------->| > | | | | > | | BYE F17 | | > | |<----------------<| | > | | | | > | | BYE F18 | | > | |<----------------<| | > | | | | > | | BYE F19 | | > | |<----------------<| | > | | | | > | | BYE F20 | | > | |<----------------<| | > | | | | > | | BYE F21 | | > | |<----------------<| | > | | | | > | | BYE F22 | | > | |<----------------<| | > | | | | > | | BYE F23 | | > | |<----------------<| | > | | | | > | | BYE F24 | | > | |<----------------<| | > | | | | > | F25 BYE | | | > |>---------------->| | | > | | | | > | | F26 BYE | | > | |>---------------->| | > | | | | > | | OK 200 F27 | | > | |<----------------<| | > | | | | > | OK 200 F28 | | | > |<----------------<| | | > > ============================================================================ ==== > > SIP MESSAGE 1 10.16.1.218:5060() -> 10.16.1.147:5060() > UDP Frame 1 20/Nov/03 16:21:14.8505 TimeFromPreviousSipFrame=0.0000 TimeFromStart=0.0000 > INVITE sip:6032251114@10.16.1.147:5060 SIP/2.0 > Via: SIP/2.0/UDP 10.16.1.218:5060 > To: sip:6032251114@10.16.1.147:5060 > From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > CSeq: 232267826 INVITE > Supported: timer > Min-SE: 5 > Content-Type: application/sdp > Content-Length: 219 > Contact: 6037534033 <sip:6037534033@10.16.1.218:5060> > Supported: replaces > > v=0 > o=MxSIP 0 758169102 IN IP4 10.16.1.218 > s=SIP Call > c=IN IP4 10.16.1.218 > t=0 0 > m=audio 5010 RTP/AVP 0 8 4 100 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:4 G723/8000 > a=rtpmap:100 telephone-event/8000 > > ============================================================================ ==== > > SIP MESSAGE 2 10.16.1.147:5060() -> 10.16.1.218:5060() > UDP Frame 2 20/Nov/03 16:21:14.8518 TimeFromPreviousSipFrame=0.0013 TimeFromStart=0.0013 > SIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/UDP 10.16.1.218:5060 > To: sip:6032251114@10.16.1.147:5060 > From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > CSeq: 232267826 INVITE > Server: Sip EXpress router (0.8.11 (i386/linux)) > Content-Length: 0 > Warning: 392 10.16.1.147:5060 "Noisy feedback tells: pid=32161 req_src_ip=10.16.1.218 req_src_port=5060 in_uri=sip:6032251114@10.16.1.147:5060 out_uri=sip:6032251114@172.17.4.102 via_cnt==1" > > > ============================================================================ ==== > > SIP MESSAGE 3 10.16.1.147:5060() -> 172.17.4.102:5060() > UDP Frame 3 20/Nov/03 16:21:14.8525 TimeFromPreviousSipFrame=0.0007 TimeFromStart=0.0020 > INVITE sip:6032251114@172.17.4.102 SIP/2.0 > Max-Forwards: 10 > Record-Route: <sip:6032251114@10.16.1.147;ftag=5c948a53eb3c31b;lr> > Via: SIP/2.0/UDP 10.16.1.147;branch=z9hG4bK369d.3afcc964.0 > Via: SIP/2.0/UDP 10.16.1.218:5060 > To: sip:6032251114@10.16.1.147:5060 > From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > CSeq: 232267826 INVITE > Supported: timer > Min-SE: 5 > Content-Type: application/sdp > Content-Length: 219 > Contact: 6037534033 <sip:6037534033@10.16.1.218:5060> > Supported: replaces > > v=0 > o=MxSIP 0 758169102 IN IP4 10.16.1.218 > s=SIP Call > c=IN IP4 10.16.1.218 > t=0 0 > m=audio 5010 RTP/AVP 0 8 4 100 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:4 G723/8000 > a=rtpmap:100 telephone-event/8000 > > ============================================================================ ==== > > SIP MESSAGE 4 172.17.4.102:5060() -> 10.16.1.147:5060() > UDP Frame 4 20/Nov/03 16:21:14.8606 TimeFromPreviousSipFrame=0.0081 TimeFromStart=0.0101 > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.16.1.147;branch=z9hG4bK369d.3afcc964.0 > Via: SIP/2.0/UDP 10.16.1.218:5060 > From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > To: <sip:6032251114@10.16.1.147:5060>;tag=1069363200-4210481069784248-11 > Record-Route: <sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr> > CSeq: 232267826 INVITE > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > > > ============================================================================ ==== > > SIP MESSAGE 5 172.17.4.102:5060() -> 10.16.1.214:5060() > UDP Frame 5 20/Nov/03 16:21:14.8667 TimeFromPreviousSipFrame=0.0062 TimeFromStart=0.0162 > INVITE sip:6032251114@10.16.1.214 SIP/2.0 > From: "6037534033" <sip:6037534033@172.17.4.102>;tag=1069363200-4276561069790856-11 > To: <sip:6032251114@10.16.1.214;user=phone> > Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1069790993 > CSeq: 52 INVITE > Contact: sip:6037534033@172.17.4.102:5060 > Call-ID: 0023639601-0837240850979601-11 > Max-Forwards: 70 > Supported: 100rel > Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER > Content-Type: application/sdp > Content-Length: 156 > > v=0 > o=MxSIP 0 758169102 IN IP4 10.16.1.218 > s=SIP Call > c=IN IP4 10.16.1.218 > t=0 0 > m=audio 5010 RTP/AVP 0 8 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > > ============================================================================ ==== > > SIP MESSAGE 6 10.16.1.214:5060() -> 172.17.4.102:5060() > UDP Frame 6 20/Nov/03 16:21:15.2762 TimeFromPreviousSipFrame=0.4095 TimeFromStart=0.4257 > SIP/2.0 180 Ringing > Call-ID: 0023639601-0837240850979601-11 > CSeq: 52 INVITE > From: "6037534033" <sip:6037534033@172.17.4.102>;tag=1069363200-4276561069790856-11 > To: <sip:6032251114@10.16.1.214;user=phone>;tag=c895ca6b2e9dc6f > Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1069790993 > Content-Length: 0 > Contact: 6032251114 <sip:6032251114@10.16.1.214:5060> > > > ============================================================================ ==== > > SIP MESSAGE 7 172.17.4.102:5060() -> 10.16.1.147:5060() > UDP Frame 7 20/Nov/03 16:21:15.2838 TimeFromPreviousSipFrame=0.0076 TimeFromStart=0.4333 > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 10.16.1.147;branch=z9hG4bK369d.3afcc964.0 > Via: SIP/2.0/UDP 10.16.1.218:5060 > From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > To: <sip:6032251114@10.16.1.147:5060>;tag=1069363200-4210481069784248-11 > Record-Route: <sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr> > CSeq: 232267826 INVITE > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > Contact: sip:6037534033@172.17.4.102:5060 > > > ============================================================================ ==== > > SIP MESSAGE 8 10.16.1.147:5060() -> 10.16.1.218:5060() > UDP Frame 8 20/Nov/03 16:21:15.2847 TimeFromPreviousSipFrame=0.0009 TimeFromStart=0.4342 > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 10.16.1.218:5060 > From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > To: <sip:6032251114@10.16.1.147:5060>;tag=1069363200-4210481069784248-11 > Record-Route: <sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr> > CSeq: 232267826 INVITE > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > Contact: sip:6037534033@172.17.4.102:5060 > > > ============================================================================ ==== > > SIP MESSAGE 9 10.16.1.214:5060() -> 172.17.4.102:5060() > UDP Frame 9 20/Nov/03 16:21:18.2880 TimeFromPreviousSipFrame=3.0033 TimeFromStart=3.4375 > SIP/2.0 200 OK > Call-ID: 0023639601-0837240850979601-11 > CSeq: 52 INVITE > From: "6037534033" <sip:6037534033@172.17.4.102>;tag=1069363200-4276561069790856-11 > To: <sip:6032251114@10.16.1.214;user=phone>;tag=c895ca6b2e9dc6f > Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1069790993 > Content-Length: 157 > Content-Type: application/sdp > Supported: replaces > Contact: 6032251114 <sip:6032251114@10.16.1.214:5060> > > v=0 > o=MxSIP 0 2025162741 IN IP4 10.16.1.214 > s=SIP Call > c=IN IP4 10.16.1.214 > t=0 0 > m=audio 5010 RTP/AVP 0 8 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > > ============================================================================ ==== > > SIP MESSAGE 10 172.17.4.102:5060() -> 10.16.1.214:5060() > UDP Frame 10 20/Nov/03 16:21:18.2935 TimeFromPreviousSipFrame=0.0055 TimeFromStart=3.4430 > ACK sip:6032251114@10.16.1.214:5060 SIP/2.0 > From: "6037534033" <sip:6037534033@172.17.4.102>;tag=1069363200-4276561069790856-11 > To: <sip:6032251114@10.16.1.214;user=phone>;tag=c895ca6b2e9dc6f > Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1069790993 > Max-Forwards: 70 > CSeq: 52 ACK > Call-ID: 0023639601-0837240850979601-11 > Contact: sip:6037534033@172.17.4.102:5060 > Content-Length: 0 > > > ============================================================================ ==== > > SIP MESSAGE 11 172.17.4.102:5060() -> 10.16.1.147:5060() > UDP Frame 11 20/Nov/03 16:21:18.2989 TimeFromPreviousSipFrame=0.0053 TimeFromStart=3.4484 > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.16.1.147;branch=z9hG4bK369d.3afcc964.0 > Via: SIP/2.0/UDP 10.16.1.218:5060 > From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > To: <sip:6032251114@10.16.1.147:5060>;tag=1069363200-4210481069784248-11 > Record-Route: <sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr> > CSeq: 232267826 INVITE > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > Contact: sip:6037534033@172.17.4.102:5060 > Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER > Content-Type: application/sdp > Content-Length: 157 > > v=0 > o=MxSIP 0 2025162741 IN IP4 10.16.1.214 > s=SIP Call > c=IN IP4 10.16.1.214 > t=0 0 > m=audio 5010 RTP/AVP 0 8 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > > ============================================================================ ==== > > SIP MESSAGE 12 10.16.1.147:5060() -> 10.16.1.218:5060() > UDP Frame 12 20/Nov/03 16:21:18.3000 TimeFromPreviousSipFrame=0.0012 TimeFromStart=3.4495 > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.16.1.218:5060 > From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > To: <sip:6032251114@10.16.1.147:5060>;tag=1069363200-4210481069784248-11 > Record-Route: <sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr> > CSeq: 232267826 INVITE > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > Contact: sip:6037534033@172.17.4.102:5060 > Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER > Content-Type: application/sdp > Content-Length: 157 > > v=0 > o=MxSIP 0 2025162741 IN IP4 10.16.1.214 > s=SIP Call > c=IN IP4 10.16.1.214 > t=0 0 > m=audio 5010 RTP/AVP 0 8 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > > ============================================================================ ==== > > SIP MESSAGE 13 10.16.1.218:5060() -> 10.16.1.147:5060() > UDP Frame 13 20/Nov/03 16:21:18.6079 TimeFromPreviousSipFrame=0.3079 TimeFromStart=3.7574 > ACK sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr SIP/2.0 > Via: SIP/2.0/UDP 10.16.1.218:5060 > To: sip:6032251114@10.16.1.147:5060;tag=1069363200-4210481069784248-11 > From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > CSeq: 232267826 ACK > Route: <sip:6037534033@172.17.4.102:5060> > Content-Length: 0 > Contact: 6037534033 <sip:6037534033@10.16.1.218:5060> > > > ============================================================================ ==== > > SIP MESSAGE 14 10.16.1.147:5060() -> 172.17.4.102:5060() > UDP Frame 14 20/Nov/03 16:21:18.6087 TimeFromPreviousSipFrame=0.0008 TimeFromStart=3.7582 > ACK sip:6037534033@172.17.4.102:5060 SIP/2.0 > Max-Forwards: 10 > Via: SIP/2.0/UDP 10.16.1.147;branch=0 > Via: SIP/2.0/UDP 10.16.1.218:5060 > To: sip:6032251114@10.16.1.147:5060;tag=1069363200-4210481069784248-11 > From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > CSeq: 232267826 ACK > Content-Length: 0 > Contact: 6037534033 <sip:6037534033@10.16.1.218:5060> > > > ============================================================================ ==== > > SIP MESSAGE 15 10.16.1.214:5060() -> 172.17.4.102:5060() > UDP Frame 17 20/Nov/03 16:21:32.0864 TimeFromPreviousSipFrame=13.4777 TimeFromStart=17.2359 > BYE sip:6037534033@172.17.4.102:5060 SIP/2.0 > Via: SIP/2.0/UDP 10.16.1.214:5060 > To: "6037534033" <sip:6037534033@172.17.4.102>;tag=1069363200-4276561069790856-11 > From: <sip:6032251114@10.16.1.214;user=phone>;tag=c895ca6b2e9dc6f > Call-ID: 0023639601-0837240850979601-11 > CSeq: 369997713 BYE > Supported: timer > Content-Length: 0 > Supported: replaces > > > ============================================================================ ==== > > SIP MESSAGE 16 172.17.4.102:5060() -> 10.16.1.214:5060() > UDP Frame 18 20/Nov/03 16:21:32.0922 TimeFromPreviousSipFrame=0.0058 TimeFromStart=17.2417 > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.16.1.214:5060 > From: <sip:6032251114@10.16.1.214;user=phone>;tag=c895ca6b2e9dc6f > To: "6037534033" <sip:6037534033@172.17.4.102>;tag=1069363200-4276561069790856-11 > CSeq: 369997713 BYE > Call-ID: 0023639601-0837240850979601-11 > > > ============================================================================ ==== > > SIP MESSAGE 17 172.17.4.102:5060() -> 10.16.1.147:5060() > UDP Frame 19 20/Nov/03 16:21:32.0926 TimeFromPreviousSipFrame=0.0004 TimeFromStart=17.2420 > BYE sip:6037534033@10.16.1.218 SIP/2.0 > Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1070017565 > From: <sip:6032251114@10.16.1.147:5060>;tag=1069363200-4210481069784248-11 > To: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > CSeq: 53 BYE > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > Max-Forwards: 10 > Route: <sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr> > Content-Length: 0 > > > ============================================================================ ==== > > SIP MESSAGE 18 172.17.4.102:5060() -> 10.16.1.147:5060() > UDP Frame 20 20/Nov/03 16:21:32.5992 TimeFromPreviousSipFrame=0.5067 TimeFromStart=17.7487 > BYE sip:6037534033@10.16.1.218 SIP/2.0 > Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1070017565 > From: <sip:6032251114@10.16.1.147:5060>;tag=1069363200-4210481069784248-11 > To: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > CSeq: 53 BYE > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > Max-Forwards: 10 > Route: <sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr> > Content-Length: 0 > > > ============================================================================ ==== > > SIP MESSAGE 19 172.17.4.102:5060() -> 10.16.1.147:5060() > UDP Frame 21 20/Nov/03 16:21:33.6090 TimeFromPreviousSipFrame=1.0098 TimeFromStart=18.7585 > BYE sip:6037534033@10.16.1.218 SIP/2.0 > Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1070017565 > From: <sip:6032251114@10.16.1.147:5060>;tag=1069363200-4210481069784248-11 > To: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > CSeq: 53 BYE > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > Max-Forwards: 10 > Route: <sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr> > Content-Length: 0 > > > ============================================================================ ==== > > SIP MESSAGE 20 172.17.4.102:5060() -> 10.16.1.147:5060() > UDP Frame 22 20/Nov/03 16:21:35.6190 TimeFromPreviousSipFrame=2.0100 TimeFromStart=20.7685 > BYE sip:6037534033@10.16.1.218 SIP/2.0 > Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1070017565 > From: <sip:6032251114@10.16.1.147:5060>;tag=1069363200-4210481069784248-11 > To: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > CSeq: 53 BYE > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > Max-Forwards: 10 > Route: <sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr> > Content-Length: 0 > > > ============================================================================ ==== > > SIP MESSAGE 21 172.17.4.102:5060() -> 10.16.1.147:5060() > UDP Frame 23 20/Nov/03 16:21:39.6292 TimeFromPreviousSipFrame=4.0102 TimeFromStart=24.7787 > BYE sip:6037534033@10.16.1.218 SIP/2.0 > Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1070017565 > From: <sip:6032251114@10.16.1.147:5060>;tag=1069363200-4210481069784248-11 > To: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > CSeq: 53 BYE > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > Max-Forwards: 10 > Route: <sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr> > Content-Length: 0 > > > ============================================================================ ==== > > SIP MESSAGE 22 172.17.4.102:5060() -> 10.16.1.147:5060() > UDP Frame 24 20/Nov/03 16:21:43.6395 TimeFromPreviousSipFrame=4.0104 TimeFromStart=28.7890 > BYE sip:6037534033@10.16.1.218 SIP/2.0 > Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1070017565 > From: <sip:6032251114@10.16.1.147:5060>;tag=1069363200-4210481069784248-11 > To: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > CSeq: 53 BYE > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > Max-Forwards: 10 > Route: <sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr> > Content-Length: 0 > > > ============================================================================ ==== > > SIP MESSAGE 23 172.17.4.102:5060() -> 10.16.1.147:5060() > UDP Frame 25 20/Nov/03 16:21:47.6496 TimeFromPreviousSipFrame=4.0101 TimeFromStart=32.7991 > BYE sip:6037534033@10.16.1.218 SIP/2.0 > Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1070017565 > From: <sip:6032251114@10.16.1.147:5060>;tag=1069363200-4210481069784248-11 > To: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > CSeq: 53 BYE > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > Max-Forwards: 10 > Route: <sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr> > Content-Length: 0 > > > ============================================================================ ==== > > SIP MESSAGE 24 172.17.4.102:5060() -> 10.16.1.147:5060() > UDP Frame 26 20/Nov/03 16:21:51.6595 TimeFromPreviousSipFrame=4.0099 TimeFromStart=36.8090 > BYE sip:6037534033@10.16.1.218 SIP/2.0 > Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1070017565 > From: <sip:6032251114@10.16.1.147:5060>;tag=1069363200-4210481069784248-11 > To: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > CSeq: 53 BYE > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > Max-Forwards: 10 > Route: <sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr> > Content-Length: 0 > > > ============================================================================ ==== > > SIP MESSAGE 25 10.16.1.218:5060() -> 10.16.1.147:5060() > UDP Frame 27 20/Nov/03 16:21:54.2541 TimeFromPreviousSipFrame=2.5946 TimeFromStart=39.4036 > BYE sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr SIP/2.0 > Via: SIP/2.0/UDP 10.16.1.218:5060 > To: sip:6032251114@10.16.1.147:5060;tag=1069363200-4210481069784248-11 > From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > CSeq: 232267827 BYE > Route: <sip:6037534033@172.17.4.102:5060> > Supported: timer > Content-Length: 0 > Supported: replaces > > > ============================================================================ ==== > > SIP MESSAGE 26 10.16.1.147:5060() -> 172.17.4.102:5060() > UDP Frame 28 20/Nov/03 16:21:54.2551 TimeFromPreviousSipFrame=0.0009 TimeFromStart=39.4046 > BYE sip:6037534033@172.17.4.102:5060 SIP/2.0 > Max-Forwards: 10 > Via: SIP/2.0/UDP 10.16.1.147;branch=z9hG4bK469d.a48a52a6.0 > Via: SIP/2.0/UDP 10.16.1.218:5060 > To: sip:6032251114@10.16.1.147:5060;tag=1069363200-4210481069784248-11 > From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > CSeq: 232267827 BYE > Supported: timer > Content-Length: 0 > Supported: replaces > > > ============================================================================ ==== > > SIP MESSAGE 27 172.17.4.102:5060() -> 10.16.1.147:5060() > UDP Frame 29 20/Nov/03 16:21:54.2611 TimeFromPreviousSipFrame=0.0061 TimeFromStart=39.4106 > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.16.1.147;branch=z9hG4bK469d.a48a52a6.0 > Via: SIP/2.0/UDP 10.16.1.218:5060 > From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > To: <sip:6032251114@10.16.1.147:5060>;tag=1069363200-4210481069784248-11 > Record-Route: <sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr> > CSeq: 232267827 BYE > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > > > ============================================================================ ==== > > SIP MESSAGE 28 10.16.1.147:5060() -> 10.16.1.218:5060() > UDP Frame 30 20/Nov/03 16:21:54.2617 TimeFromPreviousSipFrame=0.0005 TimeFromStart=39.4111 > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.16.1.218:5060 > From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=5c948a53eb3c31b > To: <sip:6032251114@10.16.1.147:5060>;tag=1069363200-4210481069784248-11 > Record-Route: <sip:6032251114@10.16.1.147:5060;ftag=5c948a53eb3c31b;lr> > CSeq: 232267827 BYE > Call-ID: e543c536ae93df9befd8b930046bbda3(a)10.16.1.147 > > > ============================================================================ ==== > > List of reasons 6 traced packets out of 34 in scenario were not included. 28 included. > [6] Arp Packets > ============================================================================ ==== > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

20 years, 7 months

[Serusers] Setting feilds for SIP User end

by Peter David

Hello all again, why is it that the docs http://iptel.org/ser/doc/ser-howto/ser-Howto.html is telling me to do a "/usr/local/sbin/ser_mysql.sh" but this is what I am getting: [root@bbcom-ser ser-0.8.11]# /usr/local/sbin/ser_mysql.sh usage: ser_mysql.sh create ser_mysql.sh drop (!!entirely deletes tables) ser_mysql.sh reinit (!!entirely deletes and than re-creates tables ser_mysql.sh backup (dumps current database to stdout) ser_mysql.sh restore <file> (restores tables from a file) ser_mysql.sh copy <new_db> (creates a new db from an existing one) ser_mysql.sh reinstall (updates to a new SER database) if you want to manipulate database as other MySql user than root, want to change database name from default value "ser", or want to use other values for users and password, edit the "config vars" section of the command ser_mysql.sh Shout I just do a "/usr/local/sbin/ser_mysql.sh create" Also I am trying to enter SIP users [Cisco 7940] I need some knowledge on the SER tables 1. which tables need to be populated to enable SIP devices to access the SER? 2. is just subsciber only? mysql> show tables; +-----------------+ | Tables_in_ser | +-----------------+ | acc | | active_sessions | | aliases | | config | | event | | grp | | location | | missed_calls | | pending | | phonebook | | reserved | | silo | | subscriber | | version | +-----------------+ 14 rows in set (0.00 sec) Peter David Network Engineer BBCOM, Inc. Office 213-489-2156 x228 pdavid(a)bbcominc.com

20 years, 7 months

[Serusers] Record-Route Questions

by Garey, Tim

Hello, Using ser 0.8.11 with redhat 9. I am not using mysql or authentication. The record-route call I'm making almost works, except for SER not passing BYE's from our softswitch. I don't see why SER is blocking these. The ACK is correctly following the record-route logic, the BYE looks like it is but SER is blocking passage. sip_scenario trace and ser.cfg routing logic are pasted here but the full sip_scenario file with packet contents are attached as a file. I don't see anything wrong with how the record-route and route headers are being used. Please help! Thank you VERY much !!! This group is terrific. Tim Attached file has sip_scenario trace 10.16.1.218 6037534033 is SER registered phone who initiates call 10.16.1.147 is SER 172.17.4.102 is our softswitch acting as back2back user agent. 10.16.1.214 6032251114 is softswitch registered phone answers/terminates call. 10.16.1.218 10.16.1.147 172.17.4.102 10.16.1.214 | | | | <Call><PFrame><Time> | | | | | F1 INVITE (sdp) | | | |>---------------->| | | | | | | | ant to us 100 F2 | | | |<----------------<| | | | | | | | | F3 INVITE (sdp) | | | |>---------------->| | | | | | | | Trying 100 F4 | | | |<----------------<| | | | | | | | | F5 INVITE (sdp) | | | |>---------------->| | | | | | | | Ringing 180 F6 | | | |<----------------<| | | | | | | Ringing 180 F7 | | | |<----------------<| | | | | | | Ringing 180 F8 | | | |<----------------<| | | | | | | | | | (sdp) OK 200 F9 | | | |<----------------<| | | | | | | | F10 ACK | | | |>---------------->| | | | | | | (sdp) OK 200 F11 | | | |<----------------<| | | | | | | (sdp) OK 200 F12 | | | |<----------------<| | | | | | | | F13 ACK | | | |>---------------->| | | | | | | | | F14 ACK | | | |>---------------->| | | | | | | | | BYE F15 | | | |<----------------<| | | | | | | | F16 200 OK | | | |>---------------->| | | | | | | BYE F17 | | | |<----------------<| | | | | | | | BYE F18 | | | |<----------------<| | | | | | | | BYE F19 | | | |<----------------<| | | | | | | | BYE F20 | | | |<----------------<| | | | | | | | BYE F21 | | | |<----------------<| | | | | | | | BYE F22 | | | |<----------------<| | | | | | | | BYE F23 | | | |<----------------<| | | | | | | | BYE F24 | | | |<----------------<| | | | | | | F25 BYE | | | |>---------------->| | | | | | | | | F26 BYE | | | |>---------------->| | | | | | | | OK 200 F27 | | | |<----------------<| | | | | | | OK 200 F28 | | | |<----------------<| | | Here is routing logic: # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; if (method=="INVITE") { rewritehostport("172.17.4.102"); record_route(); t_relay(); # forward (172.17.4.102, 5060); }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol #record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP #if (!t_relay()) { # sl_reply_error(); #}; } <<rr9.txt>>

20 years, 7 months

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sr-users November 2003