sr-users February 2006

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[Serusers] See you later at the SER- Pub today at 18!
by Greger V. Teigre 22 Feb '06

22 Feb '06

I'm about to head over to Beer Palace, Aker Brygge. See you guys there around 6pm! g-)

1 0

[Serusers] SER Dial plan
by Rakib Kanak 22 Feb '06

22 Feb '06

Hi, This is Rakib. I install SER on my Fedora core 2 server. Now i want to know how can i configure to dial on mobile (for Bangladesh e.g 88-0-152377618) and pstn. Greetings Rakib

2 2

[Serusers] Problem using SER with NAT
by Muhammad Asif Fayyaz 22 Feb '06

22 Feb '06

can anybody please help me. This is urgent. There is a continuous problem in using SER. SER is not forwarding the packet correctly to client behind NAT. Please have a look at this problem. My network scenerio is: 192.168.0.2 +----------+ | ClientA | +----------+ | | | eth1 eth0 | 192.168.0.1 | 172.25.25.83 +-------------|-------------+ | NAT1 | +-------------|-------------+ | | | 172.25.25.84 | +---------------------------+ |SER Proxy\Registrar, | | STUN Server | +---------------------------+ |172.25.25.85 | | | | 172.25.25.88 +----------+ | ClientC | +----------+ 172.25.25.X ---- LAN (External Network for client A and C) 192.168.0.X ---- Private Network 1 (Internal Network for Client A) ClientA and ClientC are sip clients with STUN functionality embedded. NAT1 is internal LAN NAT. I mean for testing purpose, i have used two machines with two network cards. -------------------------------------------------------- Here is the problem: Both clientA and clientB get registered successfully with Registrar. ClientA get registered with NAT1 address 172.25.25.83. When ClientA starts sip session by sending an invite to clientC, all works fine. But, when ClientC starts the session by inviting clientA, Proxy tries to forward the message invite directly to clientA. Although, clientA has already registered with the registrar and has indicated its contact address as of NAT1. My question is why proxy is not taking the contact address and sending invite there. below is the sequence of messages that I captured using ehtereal on Linux Plateform. source=172.25.25.88:5060 to=172.25.25.84:5060 REGISTER sip:172.25.25.84:5060 SIP/2.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport;branch=z9hG4bK1507969566 From: <sip:c@172.25.25.88:5060>;tag=1981102178 To: <sip:c@172.25.25.88:5060> Call-ID: 1003517860 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 1 REGISTER Contact: <sip:c%94@172.25.25.88:5060> Max-Forwards: 5 User-Agent: eXosip/0.1 Expires: 3600 Supported: 100rel Supported: path Content-Length: 0 source=172.25.25.84:5060 to=172.25.25.88:5060 IP/2.0 200 OK Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1507969566 From: <sip:c@172.25.25.88:5060>;tag=1981102178 To: <sip:c@172.25.25.88:5060>;tag=b27e1a1d33761e85846fc98f5f3a7e58.0a43 Call-ID: 1003517860 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 1 REGISTER Contact: <sip:c%94@172.25.25.88:5060>;expires=3600 Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 172.25.25.84:5060 "Noisy feedback tells: pid=21679 req_src_ip= 172.25.25.88 req_src_port=5060 in_uri=sip:172.25.25.84:5060 out_uri=sip: 172.25.25.84:5060 via_cnt==1" source=172.25.25.83:1025 to=172.25.25.84:5060 REGISTER sip:172.25.25.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5060;rport;branch=z9hG4bK600315716 From: <sip:a@192.168.0.2:5060>;tag=658770091 To: <sip:a@192.168.0.2:5060> Call-ID: 2091069424 [!at] 192.168.0.2 (replace the [!at] with a @) CSeq: 1 REGISTER Contact: <sip:a@172.25.25.83:1026> Max-Forwards: 5 User-Agent: eXosip/0.1 Expires: 3600 Supported: 100rel Supported: path Content-Length: 0 source=172.25.25.83:1025 to=172.25.25.84:5060 REGISTER sip:172.25.25.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5060;rport;branch=z9hG4bK600315716 From: <sip:a@192.168.0.2:5060>;tag=658770091 To: <sip:a@192.168.0.2:5060> Call-ID: 2091069424 [!at] 192.168.0.2 (replace the [!at] with a @) CSeq: 1 REGISTER Contact: <sip:a@172.25.25.83:1026> Max-Forwards: 5 User-Agent: eXosip/0.1 Expires: 3600 Supported: 100rel Supported: path Content-Length: 0 source=172.25.25.84:5060 to=172.25.25.83:1025 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.2:5060 ;rport=1025;branch=z9hG4bK600315716;received=172.25.25.83 From: <sip:a@192.168.0.2:5060>;tag=658770091 To: <sip:a@192.168.0.2:5060>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b319 Call-ID: 2091069424 [!at] 192.168.0.2 (replace the [!at] with a @) CSeq: 1 REGISTER Contact: <sip:a@172.25.25.83:1026>;expires=3600, <sip:a@172.25.25.83:5060>;expires=2868 Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 172.25.25.84:5060 "Noisy feedback tells: pid=21681 req_src_ip= 172.25.25.83 req_src_port=1025 in_uri=sip:172.25.25.84:5060 out_uri=sip: 172.25.25.84:5060 via_cnt==1" source=172.25.25.84:5060 to=172.25.25.83:1025 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.2:5060 ;rport=1025;branch=z9hG4bK600315716;received=172.25.25.83 From: <sip:a@192.168.0.2:5060>;tag=658770091 To: <sip:a@192.168.0.2:5060>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b319 Call-ID: 2091069424 [!at] 192.168.0.2 (replace the [!at] with a @) CSeq: 1 REGISTER Contact: <sip:a@172.25.25.83:1026>;expires=3600, <sip:a@172.25.25.83:5060>;expires=2868 Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 172.25.25.84:5060 "Noisy feedback tells: pid=21682 req_src_ip= 172.25.25.83 req_src_port=1025 in_uri=sip:172.25.25.84:5060 out_uri=sip: 172.25.25.84:5060 via_cnt==1" source=172.25.25.88:5060 to=172.25.25.84:5060 INVITE sip:172.25.25.84:5060 SIP/2.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport;branch=z9hG4bK1784363542 Route: <sip:a@192.168.0.2:5060> From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 5 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=172.25.25.88:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 172.25.25.84:5060 "Noisy feedback tells: pid=21677 req_src_ip= 172.25.25.88 req_src_port=5060 in_uri=sip:172.25.25.84:5060 out_uri= sip:a@192.168.0.2:5060 via_cnt==1" source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=172.25.25.88:5060 cSIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-cbaa Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 172.25.25.84:5060 "Noisy feedback tells: pid=21691 req_src_ip= 172.25.25.88 req_src_port=5060 in_uri=sip:172.25.25.84:5060 out_uri= sip:a@192.168.0.2:5060 via_cnt==0" source=172.25.25.88:5060 to=172.25.25.84:5060 ACK sip:172.25.25.84:5060 SIP/2.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport;branch=z9hG4bK1784363542 Route: <sip:a@192.168.0.2:5060> From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-cbaa Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 ACK Content-Length: 0 --------------------------------------------------------------------------------------------------------------------------------------------- the SER configuration file is: ----------------------------------------------------------------- # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ 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" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.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_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request 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 (msg:len >= 2048 ) { 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") 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==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); 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; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } --------------------------------------------------------------- Please help me in this regard. Asif

2 1

[Serusers] problems installing SER
by Davidson, Michael D 22 Feb '06

22 Feb '06

I was running an older version of SER and decided to upgrade my installation. I downloaded a solaris 10 installation and did the pkgadd -d to install it. Everything seems to have worked, but when I try and start ser using serctl start I get the following response: Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed I can't seem to get past this point. Why does it care if the ser.pid file exists since it is supposed to be starting the application? I would appreciate any suggestions?

2 1

[Serusers] undefined symbol: db_free_query
by jvr_78＠yahoo.com.ar 22 Feb '06

22 Feb '06

Hi, I have this error, undefined symbol: db_free_query I change the version of ser, 0.8.14 to ser-0.10.99-dev30-tm-timers-pa-3 what is the funcion to replace db_free_query ? best regards

2 1

[Serusers] Re: [Sems] Sems Thread Memory
by Nicky 22 Feb '06

22 Feb '06

Hi Don, Thank you for your reply. How can I check using commands that the thread is still in session? What are the commands in the scripts that close the rtp session? Please advice. Thanks Regards, Nicky ----- Original Message ----- From: don.fletcher(a)att.net To: Nicky Sent: Tuesday, February 21, 2006 2:47 AM Subject: Re: [Sems] Sems Thread Memory Nicky, Probably the RTP streams are not being closed down properly at the end of a call so that the threads remain in existence. Look at the media streams with ethereal. Look closely at call disconnect. Be sure all threads are closed. don -------------- Original message from "Nicky" <nicky(a)caliber.com.sg>: -------------- Dear All, I am having a serious problem, my thread memory size is growing to up to 2GB which applies to SEMS, SER and Mysqld thread. Please help. I am testing with about 10 people calling in at the same time. What shall I do? :( Nicky ----- Original Message ----- From: Nicky To: serusers(a)lists.iptel.org ; sems(a)lists.iptel.org Sent: Saturday, October 29, 2005 11:44 AM Subject: pthread create failed with code 11 Hi Support, I have experienced this error, please advice what can be the caused. Thanks. ERROR: start (AmThread.cpp:85): pthread create failed with code 11 Regards, Nicky

1 0

RE: [Serusers] Authentication of ReInvite
by Vitaly Nikolaev 21 Feb '06

21 Feb '06

What about loose_route ? ReINVITE must be catched by sloose_route and... for example in my config almost in the beginning, before all kind of auth checks: if (loose_route()) { if(method == "INVITE") { xlog("L_DBG", "%ci: route, loose_route said yes for INVITE msg\n"); route(1); break; } else { xlog("L_DBG", "%ci: route, loose_route said yes for non INVITE msg\n"); t_relay(); break; } }; Then on route(1) there is auth.. (it was before in main route) I do usual nat stuff and forward the call to whenever Route: field pinted: t_on_reply("1"); if (! loose_route()) { if ((!lookup("location") && method == "INVITE" && uri == myself) || uri == myself) { sl_send_reply("404", "Not Found"); break; }; } if(method == "INVITE") { /* Handle re-INVITEs */ if (force_rtp_proxy("l")) { t_on_reply("2"); } else { if(force_rtp_proxy("")) t_on_reply("2"); }; }; if (!t_relay()) { sl_reply_error(); }; So, ReINVITE will be catched in forst loose_route if and bypass auth section will be sent to route(1) where it will not do lookup(location) but send wherever route field pointed. I am wrong ? :) -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Pat wang Sent: Tuesday, February 21, 2006 10:36 AM To: klaus.mailinglists(a)pernau.at Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Authentication of ReInvite Hi Klaus, The /var/log/messages does not show any thing but here is the ngrep of the Re-Invite. I couldn't find anything wrong in the ACK for 407. Do you have any other thoughts? Thanks, Pat U 192.10.1.13:5060 -> 192.10.1.2:5060 INVITE sip:2101@192.10.1.11:5060 SIP/2.0. Via: SIP/2.0/UDP 192.10.1.13:5060;branch=z9hG4bK9523f072DDFFC805. From: "2103" <sip:2103@test.com:5060>;tag=9783CE70-D2E8B695. To: <sip:2101@test.com:5060>;tag=001280b9d20f80a2448c1c57-55a21cf1. Route: <sip:192.10.1.2;ftag=9783CE70-D2E8B695;lr=on>;ftag=9783CE70-D2E8B695;lr= on>. CSeq: 2 INVITE. Call-ID: 688c36b4-5bb67582-27524277(a)192.10.1.13. Contact: <sip:2103@192.10.1.13:5060>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0. Supported: 100rel,timer,replace. Allow-Events: talk,hold,conference. Max-Forwards: 70. Content-Type: application/sdp. Content-Length: 197. . v=0. o=- 1137681804 1137681804 IN IP4 192.10.1.13. s=Polycom IP Phone. c=IN IP4 192.10.1.13. t=0 0. m=audio 2252 RTP/AVP 0 101. a=sendonly. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. U 192.10.1.2:5060 -> 192.10.1.13:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.10.1.13:5060;branch=z9hG4bK9523f072DDFFC805. From: "2103" <sip:2103@test.com:5060>;tag=9783CE70-D2E8B695. To: <sip:2101@test.com:5060>;tag=001280b9d20f80a2448c1c57-55a21cf1. CSeq: 2 INVITE. Call-ID: 688c36b4-5bb67582-27524277(a)192.10.1.13. Proxy-Authenticate: Digest realm="test.com", nonce="43fb2e105733df37da780239bc94922da01a4177". Server: Sip EXpress router (0.9.6 (i386/linux)). Content-Length: 0. Warning: 392 192.10.1.2:5060 "Noisy feedback tells: pid=26457 req_src_ip=192.10.1.13 req_src_port=5060 in_uri=sip:2101@192.10.1.11:5060 out_uri=sip:2101@192.10.1.11:5060 via_cnt==1". . U 192.10.1.13:5060 -> 192.10.1.2:5060 ACK sip:2101@192.10.1.11:5060 SIP/2.0. Via: SIP/2.0/UDP 192.10.1.13:5060;branch=z9hG4bK9523f072DDFFC805. From: "2103" <sip:2103@test.com:5060>;tag=9783CE70-D2E8B695. To: <sip:2101@test.com:5060>;tag=001280b9d20f80a2448c1c57-55a21cf1. Route: <sip:192.10.1.2;ftag=9783CE70-D2E8B695;lr=on>;ftag=9783CE70-D2E8B695;lr= on>. CSeq: 2 ACK. Call-ID: 688c36b4-5bb67582-27524277(a)192.10.1.13. Contact: <sip:2103@192.10.1.13:5060>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0. Max-Forwards: 70. Content-Length: 0. . U 192.10.1.2:5060 -> 192.10.1.11:5060 ACK sip:2101@192.10.1.11:5060 SIP/2.0. Record-Route: <sip:192.10.1.2;ftag=9783CE70-D2E8B695;lr=on>. Via: SIP/2.0/UDP 192.10.1.2;branch=0. Via: SIP/2.0/UDP 192.10.1.13:5060;branch=z9hG4bK9523f072DDFFC805. From: "2103" <sip:2103@test.com:5060>;tag=9783CE70-D2E8B695. To: <sip:2101@test.com:5060>;tag=001280b9d20f80a2448c1c57-55a21cf1. CSeq: 2 ACK. Call-ID: 688c36b4-5bb67582-27524277(a)192.10.1.13. Contact: <sip:2103@192.10.1.13:5060>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0. Max-Forwards: 16. Content-Length: 0. P-hint: rr-enforced. . >From: Klaus Darilion <klaus.mailinglists(a)pernau.at> >To: Pat wang <wangyu39(a)hotmail.com> >CC: serusers(a)lists.iptel.org >Subject: Re: [Serusers] Authentication of ReInvite >Date: Tue, 21 Feb 2006 13:04:59 +0100 > >Pat wang wrote: > >>Morning everyone, >> >>Has anyone tried authentication of Re-INVITE on SER? When I add the >>proxy_challenge to the ReInvite in the SER configure file, the ACK for 407 >>from the client is porxied by SER to the other side. The signalling looks >>like: >> >>caller----------------------SER---------------------Callee >>------------------ normal call setup-------------------- >>--------------------------blha blha----------------------- >>------------------------------------------------------------- >>---call hold Re-Invite---> >><--- 407 challenge------ >>--------ACK-----------------> >> -------------ACK---------> *** This is >>not to be proxied! >>----ReInvite---------------> >>----------------------etc etc............ >> >>Anyone seen this while doing similar thing on SER? How can I stop SER >>proxing this ACK just like the way SER treat the original Invite with >>authentiacion? >> >>Here is the authentication part in the loose route block of my SER config >>file: >> >> if (method=="INVITE") { >> if (!proxy_authorize("test.com", "subscriber")) { >> proxy_challenge( "test.com", "0"); >> break; >> }; >> }; > >AFAIK, proxy challenge sends a stateless reply. ACK to stateless replies >should be absorbed by ser immediately without entering the ser.cfg routing >script. Thus, probably ser can't detect this ACK as stateless. > >Please watch syslog (debug=4 and "tail -f /var/log/syslog|grep -v qm_") >and verify why ser does nto detect a "stateless" ACK. > >Also verifiy the ACK if all the tags are correctly copied from 40x to the >ACK. Maybe this is cause by the branch=0 bug? > >regards >klaus _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Users] new textops functions
by Daniel-Constantin Mierla 21 Feb '06

21 Feb '06

Hello, the development version of textops module has a new set of functions that work on the body of the SIP messages. - search_body() - search a regular expression within the message's body - search_append_body() - search a regular expression within the message's body and append a text after it - replace_body() - replace a regular expression match with a text - replace_body_all() - replace all regular expression matches with a text - subst_body() - sed/perl like subst operation over the message's body More details about these functions at: http://openser.org/docs/modules/1.1.x/textops.html Cheers, Daniel

2 1

[Serusers] Authentication of ReInvite
by Pat wang 21 Feb '06

21 Feb '06

Morning everyone, Has anyone tried authentication of Re-INVITE on SER? When I add the proxy_challenge to the ReInvite in the SER configure file, the ACK for 407 from the client is porxied by SER to the other side. The signalling looks like: caller----------------------SER---------------------Callee ------------------ normal call setup-------------------- --------------------------blha blha----------------------- ------------------------------------------------------------- ---call hold Re-Invite---> <--- 407 challenge------ --------ACK-----------------> -------------ACK---------> *** This is not to be proxied! ----ReInvite---------------> ----------------------etc etc............ Anyone seen this while doing similar thing on SER? How can I stop SER proxing this ACK just like the way SER treat the original Invite with authentiacion? Here is the authentication part in the loose route block of my SER config file: if (method=="INVITE") { if (!proxy_authorize("test.com", "subscriber")) { proxy_challenge( "test.com", "0"); break; }; }; Thanks, Pat

2 3

[Serusers] avpops and wav file
by amellado＠uct.cl 21 Feb '06

21 Feb '06

Hi' I want to use voicemail in serweb. I've avpops working, but how I send the voicemail wav file to voice_silo table. Thanks.

1 0

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sr-users February 2006