sr-users November 2003

sr-users@lists.kamailio.org

116 participants
267 discussions

by Gavin Bensom

Hello, I'm having trouble getting SEMS to play greetings that I have recorded. I've recorded a greeting and saved it as default_en.wav and SEMS will play that greeting just fine so I know I have the correct file format, but when I upload a greeting via SERWeb, SEMS still plays the default_en.wav greeting. SERWeb renames the file as <username.wav> (without the <> brackets of course) where "username" is the name of the user logged into SERWeb and stores the file in the same directory as default_en.wav. What could I be doing wrong? I'm running a fairly recent version (downloaded 4-5 weeks ago) of SER and SEMS from the stable CVS branch. Thanks, G. --------------------------------- Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard

20 years, 5 months

RE: [Serusers] udp error

by Stephen Miles

Hi Jiri, Thanks for that, makes a lot of sence. I did think thats what we were doing so here is my config file for you to have a look at. Could you go through it and tell me where I am going wrong? Thanks much, Stephen route{ loose_route(); force_rport(); fix_nated_contact(); # 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 (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # 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 if(method=="REGISTER") { setflag(1); # remember this is ATA force_rport(); fix_nated_contact(); log("LOG: In NAT clause\n"); #fix_nated_sdp("3"); }; if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (uri=~"portal.commverge.co.nz") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("portal.commverge.co.nz", "subscriber")) { www_challenge("portal.commverge.co.nz", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { forward(202.180.125.200,5060); break; }; # user on-line...forward to his current destination forward(uri:host,uri:port); break; }; if (uri=~"202.180.83.14") { log("LOG: In Portal clause!\n"); rewritehostport("portal.commverge.co.nz:5060"); log("LOG: Out of Portal clause!\n"); }; if (!lookup("location")) { forward(202.180.125.200,5060); break; }; # user on-line...forward to his current destination forward(uri:host,uri:port); break; } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } -----Original Message----- From: Jiri Kuthan [mailto:jiri@iptel.org] Sent: Friday, 7 November 2003 12:42 p.m. To: Stephen Miles Subject: RE: [Serusers] udp error At 12:15 AM 11/7/2003, Stephen Miles wrote: >Hi Jiri, > >SIP support is not turned on on our pix so it is just acting as a nat device, much the same as a linux box or standard router. Than you will need RTP relay. Otherwise the media from public internet to your natted client will be dropped by the nat box. Also, you will have to configure ser to send replies symmetricaly (as opposed to address advertised in Via). Use force_rport for that. >What is happening though is that the firewall (pix in this case, cisco 827 in others) is not forwarding/natting the same port as the original request. e.i 192.168.18.12 sends sip message to portal on port 5060, firewall gets that and nats it but uses the next avalible port to send out on so the portal (our sip proxy running ser) gets the request from firewall:next_avalible_port, not firewall:5060. It seems that ser doesn't realise this and thats why we get the error, if I set my softphone (X-Lite) to use the next port the firewall is going to use (from watching logs etc) is works. Sortphone sends register as 192.168.18.12:65389 and the firewall does the same firewall:65389 and it works against ser, phone registers and calls can be make. SIP servers are supposed to reply to port number advertised in rquest's topmost via header field. >I hope this makes more sence. If not please let me know as this could be a bit of a show stopper for us using ser and we would very much like to. Nevertheless, the rtp relay workaround along with SER configured in a nat-friendly manner should help you out. -jiri

20 years, 5 months

[Serusers] nathelper & rtpproxy forward call to other SER

by Glenn Dalgliesh

Problem: One-way audio when forwarding that must go thru another negociate SER server before reaching Endpoint. I have gotten calls from SER with nathelper and rtp proxy to work reliably btw phones registered with SER box but when I trie to forward the call to a EXT on another SER box not running nathelper with register client with public ip's I get one way audio. After doing some snooping it appears that one direction of the rtp stream end up directed at the rtpproxy and the other direction goes directly to the phone. I know the sip messages must not be getting rewriten properly but I can't seem to get it right. this is the route section of my ser.cfg Thanks 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 (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # things we think we want to do for every request; # this box is dedicated to NAT only behavior record_route(); force_rport(); fix_nated_contact(); # 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") { save("location"); break; }; if (uri=~"^sip:[7][0][0][1]@") { rewritehost("sipdemo.xxxx.net"); fix_nated_sdp("3"); force_rtp_proxy(); t_relay(); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; if (method=="INVITE") { # This appears to make the called party's audio work fix_nated_sdp("3"); force_rtp_proxy(); # Call the onreply_route? This appears to link calling and called party's audio together t_on_reply("1"); }; # 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(); }; } # This appears to make calling party's audio work onreply_route[1] { if (status=~"2[0-9][0-9]") { force_rport(); fix_nated_contact(); fix_nated_sdp("3"); force_rtp_proxy(); }; }

20 years, 5 months

[Serusers] (no subject)

by Thorsten Sasse

20 years, 5 months

[Serusers] On Reply Processing Question

by Ricardo Villa

Hi, The manual explains that we can use "t_on_failure" and "failure_route" primitives to route a SIP request to another destination if the original destination does not answer. It is very clear to me how this would work for an INVITE message which receives a confirmation from the far end (like "Trying"). But when the call is established, the originating party sends an ACK message which is not supposed to receive a reply. How does SER know where to send this ACK to?? Does it again try the first destination? If so then the message will be lost. And how about INFO messages (like those used to send DTMF tones). Lets say a call is established to a backup gateway using "failure_route". Then the user sends a DTMF tone as an INFO message (maybe he called an IVR). How does SER know that this INFO message must go to the backup gateway? If SER tries the main gateway first for every single DTMF digit pressed then this is not going to work properly. Thanks, Ricardo Villa

20 years, 5 months

Re: [Serusers] Multiple Invite Problem

by Jiri Kuthan

At 05:37 PM 11/6/2003, Garey, Tim wrote: >Greetings, > >I'm very new to the SIP protocol. I am a SQA test engineer with very limited >experience using Perl and C. Currently I am testing the SIP signaling >manager on our >companys softswitch. I would like to use the SER for testing some new >features, like Record-Route >and Route Header support amoung other things. But first I'd like to get past >a problem with >SER sending multiple invites. > >I have successfully installed ser 0.8.10. on a PC running Red Hat Linux. Better begin with 8.11. In particular, it includes loose routing. >Detailed OS and SER versions >are at the end of this message. >I am not using SQL or Digest/HTTP Authentication. I want to keep things as >simple as >possible until I gain more experience with ser. > >Here is a simple diagram of my test environment > >Phone A --------------- S E R ---------------- Softswitch >---------------Phone B >603-753-4033 sip signal mangr >603-225-1114 >10.16.1.218 10.16.1.147 172.17.4.102 >10.16.1.214 > > (note: attached media gateway not used for >sip & not shown) > >I believe have setup our softswitch with correct SIP profile, E164 and >destination route for the Sip Express Router. >I can successfully originate a call from a phone registered to SER to a >phone registered to our softswitch using >the following addition to the default ser.cfg file that comes with ser >0.8.10 > >if (method=="INVITE") { > rewriteuri("sip:6032251114@172.17.4.102"); > forward(172.17.4.102); > > > > >Plese refer to the attached network trace diagram. It was captured using >Ethereal, filtering on all packets to/from >the sip signaling manager. The trace dump is then manipulated by a sip >utility tool "sip_scenario" that creates >a sip call flow diagram with decoded sip packets....this allows ease of >viewing call flow & packets > >QUESTIONS: > >1. Why am I receiving multiple invites? I suppose that's because you first 'forward' and then 't_relay' the message. >Especially after Phone B answers and >it's 200OK is received by SER, I see >another 6 invites (in addition to several extra's at the beginning of the >message). If I hang up before SER sends >the last one phone B will ring. It takes about 30 seconds or so to stop. > >2. What routing logic can I insert to take any invite starting with 603-225- >and forward it >to the softswitch with the correct dialed digits. The above works great but >is hard coded >for just one number. use regular expressions as described in our doc. if (uri=~"sip:1234.*") { forward... break; } -jiri

20 years, 5 months

[Serusers] how to install ser and configure it with voicemail

by kelvin

I have install ser-0.8.11-0.i386.rpm, mysql and serweb. My sip phone are all window messenger. When someone contact a user who is not on-line, you get this message:- phone(a)192.168.1.3 has declined your request to have a voice conversation. When I login 'phone' to serwweb as a subscriber, under missed calls is always empty & message store too. How to make it ring first and after a period go to voicemail if phone(a)192.168.1.3 is not online? I went through the document but got confuse. Do you have a example ser.cfg script that works. 2. Through the web documents there are 2 vm softwares sems-0.1.0-0.i386.rpm & ortp-0.6.2.tar.gz. which is the right one? 3. if vm is required must you reinstall ser? If yes, what to change in the makefile? Help! Thanks!

20 years, 5 months

[Serusers] Multiple Invite Problem

by Garey, Tim

Greetings, I'm very new to the SIP protocol. I am a SQA test engineer with very limited experience using Perl and C. Currently I am testing the SIP signaling manager on our companys softswitch. I would like to use the SER for testing some new features, like Record-Route and Route Header support amoung other things. But first I'd like to get past a problem with SER sending multiple invites. I have successfully installed ser 0.8.10. on a PC running Red Hat Linux. Detailed OS and SER versions are at the end of this message. I am not using SQL or Digest/HTTP Authentication. I want to keep things as simple as possible until I gain more experience with ser. Here is a simple diagram of my test environment Phone A --------------- S E R ---------------- Softswitch ---------------Phone B 603-753-4033 sip signal mangr 603-225-1114 10.16.1.218 10.16.1.147 172.17.4.102 10.16.1.214 (note: attached media gateway not used for sip & not shown) I believe have setup our softswitch with correct SIP profile, E164 and destination route for the Sip Express Router. I can successfully originate a call from a phone registered to SER to a phone registered to our softswitch using the following addition to the default ser.cfg file that comes with ser 0.8.10 if (method=="INVITE") { rewriteuri("sip:6032251114@172.17.4.102"); forward(172.17.4.102); Plese refer to the attached network trace diagram. It was captured using Ethereal, filtering on all packets to/from the sip signaling manager. The trace dump is then manipulated by a sip utility tool "sip_scenario" that creates a sip call flow diagram with decoded sip packets....this allows ease of viewing call flow & packets QUESTIONS: 1. Why am I receiving multiple invites? Especially after Phone B answers and it's 200OK is received by SER, I see another 6 invites (in addition to several extra's at the beginning of the message). If I hang up before SER sends the last one phone B will ring. It takes about 30 seconds or so to stop. 2. What routing logic can I insert to take any invite starting with 603-225- and forward it to the softswitch with the correct dialed digits. The above works great but is hard coded for just one number. 3. What similar routing logic can I insert to take any invite starting with 603-753 and send it to to phones registered on the SER side? Thanks in advance for your great support and product!! Tim G ======================================= 10.16.1.147 172.17.4.102 10.16.1.214 | | | <Call><PFrame><Time> | | | | F1 INVITE (sdp) | | |>---------------->| | 1 PF:1 16:18:51.6300 | | | | F2 INVITE (sdp) | | |>---------------->| | 1 PF:2 16:18:51.6307 | | | | Trying 100 F3 | | |<----------------<| | 1 PF:3 16:18:51.6365 | | | | Trying 100 F4 | | |<----------------<| | 1 PF:4 16:18:51.6368 | | | | | F5 INVITE (sdp) | | |>---------------->| 2 PF:5 16:18:51.6417 | | | | | F6 INVITE (sdp) | | |>---------------->| 2 PF:6 16:18:52.1491 | | | | F7 INVITE (sdp) | | |>---------------->| | 1 PF:7 16:18:52.1534 | | | | Trying 100 F8 | | |<----------------<| | 1 PF:8 16:18:52.1572 | | | | | Trying 100 F9 | | |<----------------<| 2 PF:9 16:18:52.2087 | | | | | Trying 100 F10 | | |<----------------<| 2 PF:10 16:18:52.3263 | | | | | Ringing 180 F11 | | |<----------------<| 2 PF:11 16:18:52.3307 | | | | Ringing 180 F12 | | |<----------------<| | 1 PF:12 16:18:52.3388 | | | | F13 INVITE (sdp) | | |>---------------->| | 1 PF:13 16:18:54.1735 | | | | Ringing 180 F14 | | |<----------------<| | 1 PF:14 16:18:54.1776 | | | | | (sdp) OK 200 F15 | | |<----------------<| 2 PF:15 16:18:57.8652 | | | | | F16 ACK | | |>---------------->| 2 PF:16 16:18:57.8707 | | | | (sdp) OK 200 F17 | | |<----------------<| | 1 PF:17 16:18:57.8729 | | | | F18 INVITE (sdp) | | |>---------------->| | 1 PF:18 16:18:58.2135 | | | | F19 INVITE (sdp) | | |>---------------->| | 1 PF:19 16:19:2.2536 | | | | (sdp) OK 200 F20 | | |<----------------<| | 1 PF:20 16:19:2.2578 | | | | F21 INVITE (sdp) | | |>---------------->| | 1 PF:21 16:19:6.2937 | | | | (sdp) OK 200 F22 | | |<----------------<| | 1 PF:22 16:19:6.2980 | | | | F23 INVITE (sdp) | | |>---------------->| | 1 PF:23 16:19:10.3337 | | | | (sdp) OK 200 F24 | | |<----------------<| | 1 PF:24 16:19:10.3381 | | | | F25 INVITE (sdp) | | |>---------------->| | 1 PF:25 16:19:14.3738 | | | | (sdp) OK 200 F26 | | |<----------------<| | 1 PF:26 16:19:14.3781 | | | | F27 INVITE (sdp) | | |>---------------->| | 1 PF:27 16:19:18.4139 | | | | (sdp) OK 200 F28 | | |<----------------<| | 1 PF:28 16:19:18.4182 | | | | | BYE F29 | | |<----------------<| 2 PF:29 16:20:19.9704 | | | | | F30 200 OK | | |>---------------->| 2 PF:30 16:20:19.9758 ============================================================================ ==== SIP MESSAGE 1 10.16.1.147:5060() -> 172.17.4.102:5060() UDP Frame 1 5/Nov/03 16:18:51.6300 TimeFromPreviousSipFrame=0.0000 TimeFromStart=0.0000 INVITE sip:6032251114@172.17.4.102 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 From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=42585de30f5db0e Call-ID: 346de96b02374145cfe6f6785cefedd4(a)10.16.1.147 CSeq: 1369993920 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 841122468 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() -> 172.17.4.102:5060() UDP Frame 2 5/Nov/03 16:18:51.6307 TimeFromPreviousSipFrame=0.0007 TimeFromStart=0.0007 INVITE sip:6032251114@172.17.4.102 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.147;branch=z9hG4bK0385.803c6844.0 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=42585de30f5db0e Call-ID: 346de96b02374145cfe6f6785cefedd4(a)10.16.1.147 CSeq: 1369993920 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 841122468 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 3 172.17.4.102:5060() -> 10.16.1.147:5060() UDP Frame 3 5/Nov/03 16:18:51.6365 TimeFromPreviousSipFrame=0.0057 TimeFromStart=0.0065 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.218:5060 From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=42585de30f5db0e To: <sip:6032251114@10.16.1.147:5060>;tag=1068067102-2566051068323707-11 CSeq: 1369993920 INVITE Call-ID: 346de96b02374145cfe6f6785cefedd4(a)10.16.1.147 Content-Length: 0 ============================================================================ ==== SIP MESSAGE 4 172.17.4.102:5060() -> 10.16.1.147:5060() UDP Frame 4 5/Nov/03 16:18:51.6368 TimeFromPreviousSipFrame=0.0003 TimeFromStart=0.0068 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.218:5060 From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=42585de30f5db0e To: <sip:6032251114@10.16.1.147:5060>;tag=1068067102-2566051068323707-11 CSeq: 1369993920 INVITE Call-ID: 346de96b02374145cfe6f6785cefedd4(a)10.16.1.147 Content-Length: 0 ============================================================================ ==== SIP MESSAGE 5 172.17.4.102:5060() -> 10.16.1.214:5060() UDP Frame 5 5/Nov/03 16:18:51.6417 TimeFromPreviousSipFrame=0.0049 TimeFromStart=0.0116 INVITE sip:6032251114@10.16.1.214 SIP/2.0 From: "6037534033" <sip:6037534033@172.17.4.102>;tag=1068067102-2614791068328581-11 To: <sip:6032251114@10.16.1.214;user=phone> Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1068328713 CSeq: 28 INVITE Contact: sip:6037534033@172.17.4.102:5060 Call-ID: 2017608601-3521625538238601-11 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 841122468 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 6 172.17.4.102:5060() -> 10.16.1.214:5060() UDP Frame 6 5/Nov/03 16:18:52.1491 TimeFromPreviousSipFrame=0.5074 TimeFromStart=0.5191 INVITE sip:6032251114@10.16.1.214 SIP/2.0 From: "6037534033" <sip:6037534033@172.17.4.102>;tag=1068067102-2614791068328581-11 To: <sip:6032251114@10.16.1.214;user=phone> Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1068328713 CSeq: 28 INVITE Contact: sip:6037534033@172.17.4.102:5060 Call-ID: 2017608601-3521625538238601-11 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 841122468 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 7 10.16.1.147:5060() -> 172.17.4.102:5060() UDP Frame 7 5/Nov/03 16:18:52.1534 TimeFromPreviousSipFrame=0.0044 TimeFromStart=0.5234 INVITE sip:6032251114@172.17.4.102 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.147;branch=z9hG4bK0385.803c6844.0 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=42585de30f5db0e Call-ID: 346de96b02374145cfe6f6785cefedd4(a)10.16.1.147 CSeq: 1369993920 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 841122468 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 8 172.17.4.102:5060() -> 10.16.1.147:5060() UDP Frame 8 5/Nov/03 16:18:52.1572 TimeFromPreviousSipFrame=0.0038 TimeFromStart=0.5272 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.218:5060 From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=42585de30f5db0e To: <sip:6032251114@10.16.1.147:5060>;tag=1068067102-2566051068323707-11 CSeq: 1369993920 INVITE Call-ID: 346de96b02374145cfe6f6785cefedd4(a)10.16.1.147 Content-Length: 0 ============================================================================ ==== SIP MESSAGE 9 10.16.1.214:5060() -> 172.17.4.102:5060() UDP Frame 9 5/Nov/03 16:18:52.2087 TimeFromPreviousSipFrame=0.0515 TimeFromStart=0.5787 SIP/2.0 100 Trying Call-ID: 2017608601-3521625538238601-11 CSeq: 28 INVITE From: "6037534033" <sip:6037534033@172.17.4.102>;tag=1068067102-2614791068328581-11 To: <sip:6032251114@10.16.1.214;user=phone>;tag=f3ef4c08d69a086 Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1068328713 Content-Length: 0 ============================================================================ ==== SIP MESSAGE 10 10.16.1.214:5060() -> 172.17.4.102:5060() UDP Frame 10 5/Nov/03 16:18:52.3263 TimeFromPreviousSipFrame=0.1176 TimeFromStart=0.6963 SIP/2.0 100 Trying Call-ID: 2017608601-3521625538238601-11 CSeq: 28 INVITE From: "6037534033" <sip:6037534033@172.17.4.102>;tag=1068067102-2614791068328581-11 To: <sip:6032251114@10.16.1.214;user=phone>;tag=f3ef4c08d69a086 Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1068328713 Content-Length: 0 ============================================================================ ==== SIP MESSAGE 11 10.16.1.214:5060() -> 172.17.4.102:5060() UDP Frame 11 5/Nov/03 16:18:52.3307 TimeFromPreviousSipFrame=0.0044 TimeFromStart=0.7007 SIP/2.0 180 Ringing Call-ID: 2017608601-3521625538238601-11 CSeq: 28 INVITE From: "6037534033" <sip:6037534033@172.17.4.102>;tag=1068067102-2614791068328581-11 To: <sip:6032251114@10.16.1.214;user=phone>;tag=f3ef4c08d69a086 Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1068328713 Content-Length: 0 Contact: 6032251114 <sip:6032251114@10.16.1.214:5060> ============================================================================ ==== SIP MESSAGE 12 172.17.4.102:5060() -> 10.16.1.147:5060() UDP Frame 12 5/Nov/03 16:18:52.3388 TimeFromPreviousSipFrame=0.0081 TimeFromStart=0.7088 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.218:5060 From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=42585de30f5db0e To: <sip:6032251114@10.16.1.147:5060>;tag=1068067102-2566051068323707-11 CSeq: 1369993920 INVITE Call-ID: 346de96b02374145cfe6f6785cefedd4(a)10.16.1.147 Contact: sip:6037534033@172.17.4.102:5060 Content-Length: 0 ============================================================================ ==== SIP MESSAGE 13 10.16.1.147:5060() -> 172.17.4.102:5060() UDP Frame 13 5/Nov/03 16:18:54.1735 TimeFromPreviousSipFrame=1.8347 TimeFromStart=2.5434 INVITE sip:6032251114@172.17.4.102 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.147;branch=z9hG4bK0385.803c6844.0 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=42585de30f5db0e Call-ID: 346de96b02374145cfe6f6785cefedd4(a)10.16.1.147 CSeq: 1369993920 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 841122468 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 14 172.17.4.102:5060() -> 10.16.1.147:5060() UDP Frame 14 5/Nov/03 16:18:54.1776 TimeFromPreviousSipFrame=0.0041 TimeFromStart=2.5476 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.218:5060 From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=42585de30f5db0e To: <sip:6032251114@10.16.1.147:5060>;tag=1068067102-2566051068323707-11 CSeq: 1369993920 INVITE Call-ID: 346de96b02374145cfe6f6785cefedd4(a)10.16.1.147 Contact: sip:6037534033@172.17.4.102:5060 Content-Length: 0 ============================================================================ ==== SIP MESSAGE 15 10.16.1.214:5060() -> 172.17.4.102:5060() UDP Frame 15 5/Nov/03 16:18:57.8652 TimeFromPreviousSipFrame=3.6876 TimeFromStart=6.2352 SIP/2.0 200 OK Call-ID: 2017608601-3521625538238601-11 CSeq: 28 INVITE From: "6037534033" <sip:6037534033@172.17.4.102>;tag=1068067102-2614791068328581-11 To: <sip:6032251114@10.16.1.214;user=phone>;tag=f3ef4c08d69a086 Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1068328713 Content-Length: 156 Content-Type: application/sdp Supported: replaces Contact: 6032251114 <sip:6032251114@10.16.1.214:5060> v=0 o=MxSIP 0 431376550 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 16 172.17.4.102:5060() -> 10.16.1.214:5060() UDP Frame 16 5/Nov/03 16:18:57.8707 TimeFromPreviousSipFrame=0.0055 TimeFromStart=6.2407 ACK sip:6032251114@10.16.1.214:5060 SIP/2.0 From: "6037534033" <sip:6037534033@172.17.4.102>;tag=1068067102-2614791068328581-11 To: <sip:6032251114@10.16.1.214;user=phone>;tag=f3ef4c08d69a086 Via: SIP/2.0/UDP 172.17.4.102:5060;branch=z9hG4bKSNCLLC1068328713 Max-Forwards: 70 CSeq: 28 ACK Call-ID: 2017608601-3521625538238601-11 Contact: sip:6037534033@172.17.4.102:5060 Content-Length: 0 ============================================================================ ==== SIP MESSAGE 17 172.17.4.102:5060() -> 10.16.1.147:5060() UDP Frame 17 5/Nov/03 16:18:57.8729 TimeFromPreviousSipFrame=0.0022 TimeFromStart=6.2429 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.218:5060 From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=42585de30f5db0e To: <sip:6032251114@10.16.1.147:5060>;tag=1068067102-2566051068323707-11 CSeq: 1369993920 INVITE Call-ID: 346de96b02374145cfe6f6785cefedd4(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: 156 v=0 o=MxSIP 0 431376550 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 18 10.16.1.147:5060() -> 172.17.4.102:5060() UDP Frame 18 5/Nov/03 16:18:58.2135 TimeFromPreviousSipFrame=0.3406 TimeFromStart=6.5835 INVITE sip:6032251114@172.17.4.102 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.147;branch=z9hG4bK0385.803c6844.0 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=42585de30f5db0e Call-ID: 346de96b02374145cfe6f6785cefedd4(a)10.16.1.147 CSeq: 1369993920 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 841122468 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 19 10.16.1.147:5060() -> 172.17.4.102:5060() UDP Frame 19 5/Nov/03 16:19:2.2536 TimeFromPreviousSipFrame=4.0401 TimeFromStart=10.6236 INVITE sip:6032251114@172.17.4.102 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.147;branch=z9hG4bK0385.803c6844.0 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=42585de30f5db0e Call-ID: 346de96b02374145cfe6f6785cefedd4(a)10.16.1.147 CSeq: 1369993920 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 841122468 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 20 172.17.4.102:5060() -> 10.16.1.147:5060() UDP Frame 20 5/Nov/03 16:19:2.2578 TimeFromPreviousSipFrame=0.0042 TimeFromStart=10.6278 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.218:5060 From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=42585de30f5db0e To: <sip:6032251114@10.16.1.147:5060>;tag=1068067102-2566051068323707-11 CSeq: 1369993920 INVITE Call-ID: 346de96b02374145cfe6f6785cefedd4(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: 156 v=0 o=MxSIP 0 431376550 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 21 10.16.1.147:5060() -> 172.17.4.102:5060() UDP Frame 21 5/Nov/03 16:19:6.2937 TimeFromPreviousSipFrame=4.0358 TimeFromStart=14.6636 INVITE sip:6032251114@172.17.4.102 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.147;branch=z9hG4bK0385.803c6844.0 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=42585de30f5db0e Call-ID: 346de96b02374145cfe6f6785cefedd4(a)10.16.1.147 CSeq: 1369993920 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 841122468 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 22 172.17.4.102:5060() -> 10.16.1.147:5060() UDP Frame 22 5/Nov/03 16:19:6.2980 TimeFromPreviousSipFrame=0.0044 TimeFromStart=14.6680 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.218:5060 From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=42585de30f5db0e To: <sip:6032251114@10.16.1.147:5060>;tag=1068067102-2566051068323707-11 CSeq: 1369993920 INVITE Call-ID: 346de96b02374145cfe6f6785cefedd4(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: 156 v=0 o=MxSIP 0 431376550 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 23 10.16.1.147:5060() -> 172.17.4.102:5060() UDP Frame 23 5/Nov/03 16:19:10.3337 TimeFromPreviousSipFrame=4.0357 TimeFromStart=18.7037 INVITE sip:6032251114@172.17.4.102 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.147;branch=z9hG4bK0385.803c6844.0 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=42585de30f5db0e Call-ID: 346de96b02374145cfe6f6785cefedd4(a)10.16.1.147 CSeq: 1369993920 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 841122468 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 24 172.17.4.102:5060() -> 10.16.1.147:5060() UDP Frame 24 5/Nov/03 16:19:10.3381 TimeFromPreviousSipFrame=0.0044 TimeFromStart=18.7081 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.218:5060 From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=42585de30f5db0e To: <sip:6032251114@10.16.1.147:5060>;tag=1068067102-2566051068323707-11 CSeq: 1369993920 INVITE Call-ID: 346de96b02374145cfe6f6785cefedd4(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: 156 v=0 o=MxSIP 0 431376550 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 25 10.16.1.147:5060() -> 172.17.4.102:5060() UDP Frame 25 5/Nov/03 16:19:14.3738 TimeFromPreviousSipFrame=4.0357 TimeFromStart=22.7438 INVITE sip:6032251114@172.17.4.102 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.147;branch=z9hG4bK0385.803c6844.0 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=42585de30f5db0e Call-ID: 346de96b02374145cfe6f6785cefedd4(a)10.16.1.147 CSeq: 1369993920 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 841122468 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 26 172.17.4.102:5060() -> 10.16.1.147:5060() UDP Frame 26 5/Nov/03 16:19:14.3781 TimeFromPreviousSipFrame=0.0043 TimeFromStart=22.7481 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.218:5060 From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=42585de30f5db0e To: <sip:6032251114@10.16.1.147:5060>;tag=1068067102-2566051068323707-11 CSeq: 1369993920 INVITE Call-ID: 346de96b02374145cfe6f6785cefedd4(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: 156 v=0 o=MxSIP 0 431376550 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 27 10.16.1.147:5060() -> 172.17.4.102:5060() UDP Frame 27 5/Nov/03 16:19:18.4139 TimeFromPreviousSipFrame=4.0358 TimeFromStart=26.7839 INVITE sip:6032251114@172.17.4.102 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.147;branch=z9hG4bK0385.803c6844.0 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=42585de30f5db0e Call-ID: 346de96b02374145cfe6f6785cefedd4(a)10.16.1.147 CSeq: 1369993920 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 841122468 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 28 172.17.4.102:5060() -> 10.16.1.147:5060() UDP Frame 28 5/Nov/03 16:19:18.4182 TimeFromPreviousSipFrame=0.0043 TimeFromStart=26.7882 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.16.1.147;branch=0 Via: SIP/2.0/UDP 10.16.1.218:5060 From: 6037534033 <sip:6037534033@10.16.1.147:5060>;tag=42585de30f5db0e To: <sip:6032251114@10.16.1.147:5060>;tag=1068067102-2566051068323707-11 CSeq: 1369993920 INVITE Call-ID: 346de96b02374145cfe6f6785cefedd4(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: 156 v=0 o=MxSIP 0 431376550 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 29 10.16.1.214:5060() -> 172.17.4.102:5060() UDP Frame 29 5/Nov/03 16:20:19.9704 TimeFromPreviousSipFrame=61.5522 TimeFromStart=88.3404 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=1068067102-2614791068328581-11 From: <sip:6032251114@10.16.1.214;user=phone>;tag=f3ef4c08d69a086 Call-ID: 2017608601-3521625538238601-11 CSeq: 1122903123 BYE Supported: timer Content-Length: 0 Supported: replaces ============================================================================ ==== SIP MESSAGE 30 172.17.4.102:5060() -> 10.16.1.214:5060() UDP Frame 30 5/Nov/03 16:20:19.9758 TimeFromPreviousSipFrame=0.0054 TimeFromStart=88.3458 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=f3ef4c08d69a086 To: "6037534033" <sip:6037534033@172.17.4.102>;tag=1068067102-2614791068328581-11 CSeq: 1122903123 BYE Call-ID: 2017608601-3521625538238601-11 Content-Length: 0 ======================================= # ser2hiq.cfg # $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth", "secret", "alsdkhglaksdhfkloiwr") #modparam("auth", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth", "password_column", "password") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==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") { # addRecordRoute(); # forward(165.218.202.200, 5060); #}; if (method=="INVITE") { rewriteuri("sip:6032251114@172.17.4.102"); forward(172.17.4.102); #if (uri=~"^sip:[0-9]+@10.77.25.102") { # forward(10.77.18.102, 5060); #}; }; # Do strict routing if pre-loaded route headers present # rewriteFromRoute(); # 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; # }; #sl_send_reply("200","ok"); 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 if (!t_relay()) { sl_reply_error(); }; } ================================== Detailed O/S & SER versions Linux FatBoy 2.4.20-6 #1 Thu Feb 27 10:06:59 EST 2003 i686 i686 i386 GNU/Linux version: ser 0.8.10 (i386/linux) flags: STATS:Off, USE_IPV6, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 3040 @(#) $Id: main.c,v 1.126 2002/11/11 22:36:41 andrei Exp $ main.c compiled on 18:33:14 Nov 12 2002 with gcc 2.95

20 years, 5 months

[Serusers] Strange ATA-186 register behavior

by Mike

After playing around with the new firmware (and new SER cvs snapshot 0.8.12dev-22-tcp_aliases) I'm seeing some very strange behavior from SER (ngrep trace follows): U 1.1.19.48:1387 -> 1.1.16.55:5060 REGISTER sip:sip.test.net SIP/2.0..Via: SIP/2.0/UDP 1.1.19.48:1388..From: <sip:6526@sip.test.net;user=phone>;tag=50886574..T o: <sip:6526@sip.test.net;user=phone>..Call-ID: 4136319346@10.118.145.13..CSeq: 160 REGISTER..Contact: <sip:6526@1.1.19.48: 1388;user=phone;transport=udp>;expires=1800..User-Agent: Cisco ATA 186 v2.16.2 ata18x (030909a)..Content-Length: 0.... # U 1.1.16.55:5060 -> 1.1.19.48:1388 25017@0:1480 ...l....SIP/2.0 200 OK..Via: SIP/2.0/UDP 1.1.19.48:1388..From: <sip:6526@sip.test.net;user=phone>;tag=50886574..To: <sip:65 26(a)sip.test.net;user=phone>;tag=0596aef19d8edd571cbf9827f60616bb.97c5..Call-ID: 4136319346@10.118.145.13..CSeq: 160 REGISTER..Co ntact: <sip:6526@1.1.19.48:1376;user=phone;transport=udp>;q=0.00;expires=30..Contact: <sip:6526@1.1.19.48:1377;user=p hone;transport=udp>;q=0.00;expires=56..Contact: <sip:6526@1.1.19.48:1378;user=phone;transport=udp>;q=0.00;expires=517..Con tact: <sip:6526@1.1.19.48:1379;user=phone;transport=udp>;q=0.00;expires=527..Contact: <sip:6526@1.1.19.48:1380;user=p hone;transport=udp>;q=0.00;expires=652..Contact: <sip:6526@1.1.19.48:1381;user=phone;transport=udp>;q=0.00;expires=780..Co ntact: <sip:6526@1.1.19.48:1382;user=phone;transport=udp>;q=0.00;expires=911..Contact: <sip:6526@1.1.19.48:1383;user= phone;transport=udp>;q=0.00;expires=1040..Contact: <sip:6526@1.1.19.48:1385;user=phone;transport=udp>;q=0.00;expires=1303. .Contact: <sip:6526@1.1.19.48:1386;user=phone;transport=udp>;q=0.00;expires=1532..Contact: <sip:6526@1.1.19.48:1387;u ser=phone;transport=udp>;q=0.00;expires=1576..Contact: <sip:6526@1.1.19.48:1388;user=phone;transport=udp>;q=0.00;expires=1 800..Server: Sip EXpress router (0.8.12dev-22-tcp_aliases (i386/linux))..Content-Length: 0..Warning: 392 sip.test.net:5060 "Nois y feedback tells: pid=11260 req_src_ip=1.1.19.48 req_src_port=1387 in_uri=sip This is repeated over and over and over again. This seems very bizzare to me. ATAs with earlier firmware don't trigger this behavior. The only thing that I can see different in the register request is the addition of the tag in the From: line: From: <sip:6526@sip.test.net;user=phone>;tag=50886574 whereas the old ATA registers with: From: <sip:6526@sip.test.net;user=phone> Any clues?? Thanks!! - Mike

20 years, 5 months

[Serusers] Nathelper & rtpproxy - Poor audio with Xten clients

by Adrian Georgescu

I get pretty poor audio with Xten while using RTPproxy. The sound is choppy ever 5 seconds or so I miss half a second. ATA, Snom200, IPdialog and 7960 give all good results but these are all hardware phones. Could someone confirm this happens only with Xten? Anyone tried successfully another software phone with rtpproxy? Regards, Adrian Georgescu ag(a)ag-projects.com http://ag-projects.com Tel: +31-23-5458104 IP phone: sip:ag@ag-projects.com ------------------------------------------------ DNS, ENUM & IP telephony http://managed-dns.org/

20 years, 5 months

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sr-users November 2003