SER gurus: I am an newbie to SER usuage. At present a dynamic redirect issue happens to me. But it is very pity for SER not to support this case!!
Problem Description: ==================== When configuring SER as only redirect server, config file regarding redirect routing section is partly writen as follows: #-----------------------request routing logic------------------------- #main routing logic if (method=="INVITE") { #rewrite current URI, which is always part of destination ser rewriteuri("sip:80000000@192.168.0.191:5060"); #redirect now sl_send_reply("301", "Redirect"); break; } #--------------------------------------------------------------------- where, redirecting destination URI is staticly configed into SER, but dynamic destination URI is what our IP-PBX product(RTCCP) want to hope. The following call flow diagram helps to understand above mentioned scenario.
GrandStream HandyTone486 SER Our B2BUA IP PBX(RTCCP) IP: 192.168.0.253 IP: 192.168.0.252 IP: 192.168.0.191 +++++++++++++++++++++++++++++++++++++ First Call+++++++++++++++++++++++++++++++++++++++++++++++++ |-----------------F1(INVITE)---------->| |<-----------------F2(302)------------>| |-----------------F3(ACK)------------->| |------------------------------------F4(INVITE)------------------------>| ( The subsequent call flow is omitted ) +++++++++++++++++++++++++++++++++++++ Second Call+++++++++++++++++++++++++++++++++++++++++++++++++ |-----------------F5(INVITE)---------->| |<-----------------F6(302)------------>| |-----------------F7(ACK)------------->| |------------------------------------F8(INVITE)------------------------>| ( The subsequent call flow is omitted ) F1: INVITE sip:8000000@192.168.0.252;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKb537ec0f1387845f From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=c526ecce44ec6d75 To: sip:8000000@192.168.0.252;user=phone Contact: sip:60000253@192.168.0.233;user=phone Supported: replaces Call-ID: d6169571fe4f59a2@192.168.0.233 CSeq: 50105 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330
v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20
F2: SIP/2.0 301 Redirect Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKb537ec0f1387845f From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=c526ecce44ec6d75 To: sip:8000000@192.168.0.252;user=phone;tag=b27e1a1d33761e85846fc98f5f3a7e58.f9ed Call-ID: d6169571fe4f59a2@192.168.0.233 CSeq: 50105 INVITE Contact: sip:80000000@192.168.0.191:5060 Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 192.168.0.252:5060 "Noisy feedback tells: pid=6155 req_src_ip=192.168.0.233 req_src_port=5060 in_uri=sip:8000000@192.168.0.252;user=phone out_uri=sip:80000000@192.168.0.191:5060 via_cnt==1"
F4: INVITE sip:80000000@192.168.0.191:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKc6905874d1906c58 From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=1be4a5d7d169d8f1 To: sip:80000000@192.168.0.191:5060 Contact: sip:60000253@192.168.0.233;user=phone Supported: replaces Call-ID: 0d325b0e778321ee@192.168.0.233 CSeq: 17809 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330
v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20
F5: INVITE sip:80000002@192.168.0.252;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKe966424e91ceac8e From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=466d9b06b51ca7b4 To: sip:80000002@192.168.0.252;user=phone Contact: sip:60000253@192.168.0.233;user=phone Supported: replaces Call-ID: c755c6b0908a8ce9@192.168.0.233 CSeq: 25248 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330
v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20
F6: SIP/2.0 301 Redirect Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKe966424e91ceac8e From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=466d9b06b51ca7b4 To: sip:80000002@192.168.0.252;user=phone;tag=b27e1a1d33761e85846fc98f5f3a7e58.7906 Call-ID: c755c6b0908a8ce9@192.168.0.233 CSeq: 25248 INVITE Contact: sip:80000000@192.168.0.191:5060 Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 192.168.0.252:5060 "Noisy feedback tells: pid=6151 req_src_ip=192.168.0.233 req_src_port=5060 in_uri=sip:80000002@192.168.0.252;user=phone out_uri=sip:80000000@192.168.0.191:5060 via_cnt==1"
F8: INVITE sip:80000000@192.168.0.191:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bK475fff37270b762d From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=58a90f4229e70a80 To: sip:80000000@192.168.0.191:5060 Contact: sip:60000253@192.168.0.233;user=phone Supported: replaces Call-ID: 2a79d9ac84e2f72e@192.168.0.233 CSeq: 62121 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330
v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20
In fact, I think SER should retrieve user portion of header "to" URI of Invite message(for above example it should be 80000000 for first call,whereas 80000002 for second call) to construct userinfo of destination redirect uri for header contact of 3xx response(but hostinfo maybe get from SER config file). Meanwhile I study into sl_send_reply source code, but no progress to issue. So any suggestions or solutions are appreciated!!
Thanks in advance!!
George Lee(ShenZhen, CHINA)
I am not sure if I understand the question. If you want to redirect the INVITE, then you need to get the new destination from somewhere -- either configure it statically in the configuration as you have done, or use the user location database.
To use the user location database, call lookup("location") and then 3xx back using sl_send reply:
lookup("location"); sl_send_reply("301", "Redirect");
That way SER would put the real destinations (registered by the called user agent) into Contact and send it back to the calling user agent.
Jan.
On 03-02 15:14, George Lee wrote:
SER gurus: I am an newbie to SER usuage. At present a dynamic redirect issue happens to me. But it is very pity for SER not to support this case!!
Problem Description:
When configuring SER as only redirect server, config file regarding redirect routing section is partly writen as follows:#-----------------------request routing logic------------------------- #main routing logic if (method=="INVITE") { #rewrite current URI, which is always part of destination ser rewriteuri("sip:80000000@192.168.0.191:5060"); #redirect now sl_send_reply("301", "Redirect"); break; } #--------------------------------------------------------------------- where, redirecting destination URI is staticly configed into SER, but dynamic destination URI is what our IP-PBX product(RTCCP) want to hope. The following call flow diagram helps to understand above mentioned scenario.
GrandStream HandyTone486 SER Our B2BUA IP PBX(RTCCP) IP: 192.168.0.253 IP: 192.168.0.252 IP: 192.168.0.191+++++++++++++++++++++++++++++++++++++ First Call+++++++++++++++++++++++++++++++++++++++++++++++++ |-----------------F1(INVITE)---------->| |<-----------------F2(302)------------>| |-----------------F3(ACK)------------->| |------------------------------------F4(INVITE)------------------------>| ( The subsequent call flow is omitted ) +++++++++++++++++++++++++++++++++++++ Second Call+++++++++++++++++++++++++++++++++++++++++++++++++ |-----------------F5(INVITE)---------->| |<-----------------F6(302)------------>| |-----------------F7(ACK)------------->| |------------------------------------F8(INVITE)------------------------>| ( The subsequent call flow is omitted ) F1: INVITE sip:8000000@192.168.0.252;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKb537ec0f1387845f From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=c526ecce44ec6d75 To: sip:8000000@192.168.0.252;user=phone Contact: sip:60000253@192.168.0.233;user=phone Supported: replaces Call-ID: d6169571fe4f59a2@192.168.0.233 CSeq: 50105 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330
v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20
F2: SIP/2.0 301 Redirect Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKb537ec0f1387845f From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=c526ecce44ec6d75 To: sip:8000000@192.168.0.252;user=phone;tag=b27e1a1d33761e85846fc98f5f3a7e58.f9ed Call-ID: d6169571fe4f59a2@192.168.0.233 CSeq: 50105 INVITE Contact: sip:80000000@192.168.0.191:5060 Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 192.168.0.252:5060 "Noisy feedback tells: pid=6155 req_src_ip=192.168.0.233 req_src_port=5060 in_uri=sip:8000000@192.168.0.252;user=phone out_uri=sip:80000000@192.168.0.191:5060 via_cnt==1"
F4: INVITE sip:80000000@192.168.0.191:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKc6905874d1906c58 From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=1be4a5d7d169d8f1 To: sip:80000000@192.168.0.191:5060 Contact: sip:60000253@192.168.0.233;user=phone Supported: replaces Call-ID: 0d325b0e778321ee@192.168.0.233 CSeq: 17809 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330
v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20
F5: INVITE sip:80000002@192.168.0.252;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKe966424e91ceac8e From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=466d9b06b51ca7b4 To: sip:80000002@192.168.0.252;user=phone Contact: sip:60000253@192.168.0.233;user=phone Supported: replaces Call-ID: c755c6b0908a8ce9@192.168.0.233 CSeq: 25248 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330
v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20
F6: SIP/2.0 301 Redirect Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKe966424e91ceac8e From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=466d9b06b51ca7b4 To: sip:80000002@192.168.0.252;user=phone;tag=b27e1a1d33761e85846fc98f5f3a7e58.7906 Call-ID: c755c6b0908a8ce9@192.168.0.233 CSeq: 25248 INVITE Contact: sip:80000000@192.168.0.191:5060 Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 192.168.0.252:5060 "Noisy feedback tells: pid=6151 req_src_ip=192.168.0.233 req_src_port=5060 in_uri=sip:80000002@192.168.0.252;user=phone out_uri=sip:80000000@192.168.0.191:5060 via_cnt==1"
F8: INVITE sip:80000000@192.168.0.191:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bK475fff37270b762d From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=58a90f4229e70a80 To: sip:80000000@192.168.0.191:5060 Contact: sip:60000253@192.168.0.233;user=phone Supported: replaces Call-ID: 2a79d9ac84e2f72e@192.168.0.233 CSeq: 62121 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330
v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20
In fact, I think SER should retrieve user portion of header "to" URI of Invite message(for above example it should be 80000000 for first call,whereas 80000002 for second call) to construct userinfo of destination redirect uri for header contact of 3xx response(but hostinfo maybe get from SER config file). Meanwhile I study into sl_send_reply source code, but no progress to issue. So any suggestions or solutions are appreciated!!Thanks in advance!!
George Lee(ShenZhen, CHINA)
Serdev mailing list serdev@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serdev
Hello Jan : In fact, I need SER to implement a speific request-redirect routing logic that currently not supported by SER. At present, we can call this logic "ip pbx redirect logic" described as follows in detail: ============================================================================= When incoming "Invite" request arrives, SER extracts username of "To" header field of "Invite" request as username of "Contact" header field in 3xx response whereas hostname or hostip of "Contact" header is retrieved from external configuration such as ser.cfg file. Simply speaking:
redirect-generated "Contact" = [username of "To" header field of "Invite" request]@[hostname or hostip get from external configuration such as ser.cfg file]
Thanks a Lot! George Lee
----- Original Message ----- From: "Jan Janak" jan@iptel.org To: "George Lee" jlsnmp@126.com Cc: serusers@iptel.org; serdev@iptel.org Sent: Thursday, February 03, 2005 9:05 PM Subject: Re: [Serdev] Dynamic Redirect Issue about SER!
I am not sure if I understand the question. If you want to redirect the INVITE, then you need to get the new destination from somewhere -- either configure it statically in the configuration as you have done, or use the user location database.
To use the user location database, call lookup("location") and then 3xx back using sl_send reply:
lookup("location"); sl_send_reply("301", "Redirect");
That way SER would put the real destinations (registered by the called user agent) into Contact and send it back to the calling user agent.
Jan.
On 03-02 15:14, George Lee wrote:
SER gurus: I am an newbie to SER usuage. At present a dynamic redirect issue happens to me. But it is very pity for SER not to support this case!!
Problem Description:
When configuring SER as only redirect server, config file regarding redirect routing section is partly writen as follows:#-----------------------request routing logic------------------------- #main routing logic if (method=="INVITE") { #rewrite current URI, which is always part of destination ser rewriteuri("sip:80000000@192.168.0.191:5060"); #redirect now sl_send_reply("301", "Redirect"); break; } #--------------------------------------------------------------------- where, redirecting destination URI is staticly configed into SER, but dynamic destination URI is what our IP-PBX product(RTCCP) want to hope. The following call flow diagram helps to understand above mentioned scenario.
GrandStream HandyTone486 SER Our B2BUA IP PBX(RTCCP) IP: 192.168.0.253 IP: 192.168.0.252 IP: 192.168.0.191+++++++++++++++++++++++++++++++++++++ First Call+++++++++++++++++++++++++++++++++++++++++++++++++ |-----------------F1(INVITE)---------->| |<-----------------F2(302)------------>| |-----------------F3(ACK)------------->| |------------------------------------F4(INVITE)------------------------>| ( The subsequent call flow is omitted ) +++++++++++++++++++++++++++++++++++++ Second Call+++++++++++++++++++++++++++++++++++++++++++++++++ |-----------------F5(INVITE)---------->| |<-----------------F6(302)------------>| |-----------------F7(ACK)------------->| |------------------------------------F8(INVITE)------------------------>| ( The subsequent call flow is omitted ) F1: INVITE sip:8000000@192.168.0.252;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKb537ec0f1387845f From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=c526ecce44ec6d75 To: sip:8000000@192.168.0.252;user=phone Contact: sip:60000253@192.168.0.233;user=phone Supported: replaces Call-ID: d6169571fe4f59a2@192.168.0.233 CSeq: 50105 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330
v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20
F2: SIP/2.0 301 Redirect Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKb537ec0f1387845f From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=c526ecce44ec6d75 To: sip:8000000@192.168.0.252;user=phone;tag=b27e1a1d33761e85846fc98f5f3a7e58.f9ed Call-ID: d6169571fe4f59a2@192.168.0.233 CSeq: 50105 INVITE Contact: sip:80000000@192.168.0.191:5060 Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 192.168.0.252:5060 "Noisy feedback tells: pid=6155 req_src_ip=192.168.0.233 req_src_port=5060 in_uri=sip:8000000@192.168.0.252;user=phone out_uri=sip:80000000@192.168.0.191:5060 via_cnt==1"
F4: INVITE sip:80000000@192.168.0.191:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKc6905874d1906c58 From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=1be4a5d7d169d8f1 To: sip:80000000@192.168.0.191:5060 Contact: sip:60000253@192.168.0.233;user=phone Supported: replaces Call-ID: 0d325b0e778321ee@192.168.0.233 CSeq: 17809 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330
v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20
F5: INVITE sip:80000002@192.168.0.252;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKe966424e91ceac8e From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=466d9b06b51ca7b4 To: sip:80000002@192.168.0.252;user=phone Contact: sip:60000253@192.168.0.233;user=phone Supported: replaces Call-ID: c755c6b0908a8ce9@192.168.0.233 CSeq: 25248 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330
v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20
F6: SIP/2.0 301 Redirect Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKe966424e91ceac8e From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=466d9b06b51ca7b4 To: sip:80000002@192.168.0.252;user=phone;tag=b27e1a1d33761e85846fc98f5f3a7e58.7906 Call-ID: c755c6b0908a8ce9@192.168.0.233 CSeq: 25248 INVITE Contact: sip:80000000@192.168.0.191:5060 Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 192.168.0.252:5060 "Noisy feedback tells: pid=6151 req_src_ip=192.168.0.233 req_src_port=5060 in_uri=sip:80000002@192.168.0.252;user=phone out_uri=sip:80000000@192.168.0.191:5060 via_cnt==1"
F8: INVITE sip:80000000@192.168.0.191:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bK475fff37270b762d From: "60000253" sip:60000253@192.168.0.252;user=phone;tag=58a90f4229e70a80 To: sip:80000000@192.168.0.191:5060 Contact: sip:60000253@192.168.0.233;user=phone Supported: replaces Call-ID: 2a79d9ac84e2f72e@192.168.0.233 CSeq: 62121 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330
v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20
In fact, I think SER should retrieve user portion of header "to" URI of Invite message(for above example it should be 80000000 for first call,whereas 80000002 for second call) to construct userinfo of destination redirect uri for header contact of 3xx response(but hostinfo maybe get from SER config file). Meanwhile I study into sl_send_reply source code, but no progress to issue. So any suggestions or solutions are appreciated!!Thanks in advance!!
George Lee(ShenZhen, CHINA)
Serdev mailing list Serdev@iptel.org http://mail.iptel.org/mailman/listinfo/serdev
IMHO you should use username from the Request-URI, not from To. To may contain a completely different URI.
Jan.
On 04-02 15:18, George Lee wrote:
Hello Jan : In fact, I need SER to implement a speific request-redirect routing logic that currently not supported by SER. At present, we can call this logic "ip pbx redirect logic" described as follows in detail: ============================================================================= When incoming "Invite" request arrives, SER extracts username of "To" header field of "Invite" request as username of "Contact" header field in 3xx response whereas hostname or hostip of "Contact" header is retrieved from external configuration such as ser.cfg file. Simply speaking:
redirect-generated "Contact" = [username of "To" header field of "Invite" request]@[hostname or hostip get from external configuration such as ser.cfg file]Thanks a Lot! George Lee
hello jan: For my need for specific ip-pbx request routing logic, can you give me a more detailed step description to issue?
Thanks a lot!
----- Original Message ----- From: "Jan Janak" jan@iptel.org To: "George Lee" jlsnmp@126.com Cc: serusers@iptel.org; serdev@iptel.org Sent: Friday, February 04, 2005 6:38 PM Subject: Re: [Serdev] Dynamic Redirect Issue about SER!
IMHO you should use username from the Request-URI, not from To. To may contain a completely different URI.
Jan.
On 04-02 15:18, George Lee wrote:
Hello Jan : In fact, I need SER to implement a speific request-redirect routing logic that currently not supported by SER. At present, we can call this logic "ip pbx redirect logic" described as follows in detail: ============================================================================= When incoming "Invite" request arrives, SER extracts username of "To" header field of "Invite" request as username of "Contact" header field in 3xx response whereas hostname or hostip of "Contact" header is retrieved from external configuration such as ser.cfg file. Simply speaking:
redirect-generated "Contact" = [username of "To" header field of "Invite" request]@[hostname or hostip get from external configuration such as ser.cfg file]Thanks a Lot! George Lee
Hi George,
you can implement the logic you require very easily with the avpops module: rewritehost("new_host"); avp_write("$to/username","i:13"); avp_pushto("$ruri/username","i:13");
Best regards, Marian
George Lee wrote:
Hello Jan : In fact, I need SER to implement a speific request-redirect routing logic that currently not supported by SER. At present, we can call this logic "ip pbx redirect logic" described as follows in detail: ============================================================================= When incoming "Invite" request arrives, SER extracts username of "To" header field of "Invite" request as username of "Contact" header field in 3xx response whereas hostname or hostip of "Contact" header is retrieved from external configuration such as ser.cfg file. Simply speaking:
redirect-generated "Contact" = [username of "To" header field of "Invite" request]@[hostname or hostip get from external configuration such as ser.cfg file]Thanks a Lot! George Lee
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George Lee -
Jan Janak -
Marian Dumitru