Hello all!
I finally have something that I could call a working config (almost that is), the proxy works fine with voice and fax sent over g.711 (but I guess re-invites will cause problems in this scenario also). The problem occurs when I try to switch to T.38. Since it works with voice my guess is that the problem is caused by the re-invite to t.38. I guess it has to be solved by a if (loose_route()) section, but I am little bit clueless right now, I have tried multiple variants but with no success.
RTP proxy is started with the -l flag to distinguish between local and public traffic. -l PUBLIC.IP.ETH0/LOCAL.IP.ETH1
The setup: Sonus UAC/Trunk <-private ip-> (eth1)Kamailio/RTPProxy(eth0) <-public ip-> Asterisk
IP.TO.TRUNK = Sonus IP, public address range, but only reachable by private net (from the range of LOCAL.IP.ETH1).
LOCAL.IP.ETH1 = Local net, static route to IP.TO.TRUNK. PUBLIC.IP.ETH0 = Public routable address. IP.TO.ASTERISK = Public routable address.
The purpose of Kamailio/RTPProxy are to proxy everything between the routable and the private net.
Does anyone here have any idea of how I'm going to solve my issue? It would also be nice if anyone here have better solutions to my approach, and there could also be some other errors in my config that causes this behaviour.
####### Routing ########
route {
if (msg:len >= 2048) { sl_send_reply("513", "Message too big"); exit; }
if (method != "INVITE" && method != "ACK" && method != "BYE" && method != "CANCEL") { sl_send_reply("503", "Service Unavailable"); exit; }
if (!has_totag()) { record_route(); route("1"); }
if (has_totag()) { if (!loose_route()) { if (t_check_trans()) { t_relay(); exit; } exit; } }
if (has_body("application/sdp")) { if (method == "INVITE") { route("2"); t_on_reply("2"); } else if (method == "ACK") { rtpproxy_answer(); } } else if (method == "INVITE") { t_on_reply("2"); }
if (method == "BYE" || method == "CANCEL") { unforce_rtp_proxy(); }
t_relay(); }
#RELAY BY SOURCE ROUTE route[1] { if (src_ip == "IP.TO.TRUNK" || dst_ip == "LOCAL.IP.ETH1") { force_send_socket("PUBLIC.IP.ETH0"); rewritehost("IP.TO.ASTERISK"); } else { force_send_socket("LOCAL.IP.ETH1"); rewritehost("IP.TO.TRUNK"); } }
# RTP OFFER ROUTE route[2] { if (src_ip == "IP.TO.TRUNK" || dst_ip == "LOCAL.IP.ETH1") force_rtp_proxy("eir"); else force_rtp_proxy("ier"); }
# RTP OFFER ONREPLY onreply_route[1] { if (has_body("application/sdp")) route("2"); }
# RTP ANSWER ONREPLY onreply_route[2] { if (has_body("application/sdp")) route("2");
}
## END OF CONFIG
Sorry for the long post, but I hope someone outthere are able to help med with this issue.
Hi!
1. it is useful to find out if the problem happens due to a T.38 reINVITE or with all reINVITEs. You could easily test this by just putting a call on-hold, and back off-hold (e.g. by pressing the line button in Xlite) and verify if audio works again.
more inline ...
Am 14.04.2010 11:28, schrieb Espen Berg:
Hello all!
I finally have something that I could call a working config (almost that is), the proxy works fine with voice and fax sent over g.711 (but I guess re-invites will cause problems in this scenario also). The problem occurs when I try to switch to T.38. Since it works with voice my guess is that the problem is caused by the re-invite to t.38. I guess it has to be solved by a if (loose_route()) section, but I am little bit clueless right now, I have tried multiple variants but with no success.
RTP proxy is started with the -l flag to distinguish between local and public traffic. -l PUBLIC.IP.ETH0/LOCAL.IP.ETH1
I would reverse the interfaces as the first interface corresponds with the "internal" interface and the second with the "external". See 'ie' flags at http://sip-router.org/docbook/sip-router/branch/master/modules_k/nathelper/n...
When a call is routed from local IP to public IP I would like to name it internal-to-external, hence 'ie' flags instead of 'ei', but that s just a matter of naming and shouldn't cause problems.
The setup: Sonus UAC/Trunk <-private ip-> (eth1)Kamailio/RTPProxy(eth0) <-public ip-> Asterisk
IP.TO.TRUNK = Sonus IP, public address range, but only reachable by private net (from the range of LOCAL.IP.ETH1).
LOCAL.IP.ETH1 = Local net, static route to IP.TO.TRUNK. PUBLIC.IP.ETH0 = Public routable address. IP.TO.ASTERISK = Public routable address.
The purpose of Kamailio/RTPProxy are to proxy everything between the routable and the private net.
Does anyone here have any idea of how I'm going to solve my issue? It would also be nice if anyone here have better solutions to my approach, and there could also be some other errors in my config that causes this behaviour.
####### Routing ########
route {
if (msg:len >= 2048) { sl_send_reply("513", "Message too big"); exit; }
if (method != "INVITE" && method != "ACK" && method != "BYE" && method != "CANCEL") { sl_send_reply("503", "Service Unavailable"); exit; }
if (!has_totag()) { record_route(); route("1"); }
if (has_totag()) { if (!loose_route()) { if (t_check_trans()) { t_relay(); exit; } exit; } }
That's strange. You shouldn't allow requests with totag but without loose-routing. But that's not related to your problem.
if (has_body("application/sdp")) { if (method == "INVITE") { route("2"); t_on_reply("2"); } else if (method == "ACK") { rtpproxy_answer(); } } else if (method == "INVITE") { t_on_reply("2"); }
if (method == "BYE" || method == "CANCEL") { unforce_rtp_proxy(); }
t_relay(); }
#RELAY BY SOURCE ROUTE route[1] { if (src_ip == "IP.TO.TRUNK" || dst_ip == "LOCAL.IP.ETH1") { force_send_socket("PUBLIC.IP.ETH0"); rewritehost("IP.TO.ASTERISK"); } else { force_send_socket("LOCAL.IP.ETH1"); rewritehost("IP.TO.TRUNK"); } }
# RTP OFFER ROUTE route[2] { if (src_ip == "IP.TO.TRUNK" || dst_ip == "LOCAL.IP.ETH1") force_rtp_proxy("eir"); else force_rtp_proxy("ier"); }
# RTP OFFER ONREPLY onreply_route[1] { if (has_body("application/sdp")) route("2"); }
# RTP ANSWER ONREPLY onreply_route[2] { if (has_body("application/sdp")) route("2");
}
## END OF CONFIG
Sorry for the long post, but I hope someone outthere are able to help med with this issue.
IMO the config is a bit strange but I couldn't spot an error.
Trace the scenario with ngrep and take a look at the SDPs if they are rewritten properly.
regards klaus
Den 14.04.2010 14:27, skrev Klaus Darilion:
- it is useful to find out if the problem happens due to a T.38
reINVITE or with all reINVITEs. You could easily test this by just putting a call on-hold, and back off-hold (e.g. by pressing the line button in Xlite) and verify if audio works again.
more inline ...
Seems to be no problems there, xlite are able to pickup the call again after "on-hold". But I have to do some more testing here.
I finally have something that I could call a working config (almost that is), the proxy works fine with voice and fax sent over g.711 (but I guess re-invites will cause problems in this scenario also). The problem occurs when I try to switch to T.38. Since it works with voice my guess is that the problem is caused by the re-invite to t.38. I guess it has to be solved by a if (loose_route()) section, but I am little bit clueless right now, I have tried multiple variants but with no success. RTP proxy is started with the -l flag to distinguish between local and public traffic. -l PUBLIC.IP.ETH0/LOCAL.IP.ETH1
I would reverse the interfaces as the first interface corresponds with the "internal" interface and the second with the "external". See 'ie' flags at http://sip-router.org/docbook/sip-router/branch/master/modules_k/nathelper/n... When a call is routed from local IP to public IP I would like to name it internal-to-external, hence 'ie' flags instead of 'ei', but that s just a matter of naming and shouldn't cause problems.
As long as I use the ie/ei flags correct in the "RTP OFFER ROUTE"-section, everything should be fine? The RTP offer and RTP answer onreply route both refer to route 2.
route[2] { if (src_ip == "IP.TO.TRUNK" || dst_ip == "LOCAL.IP.ETH1") force_rtp_proxy("eir"); else force_rtp_proxy("ier"); }
But in my head that should be correct since i filter on the IP-address. Correct?
if (has_totag()) { if (!loose_route()) { if (t_check_trans()) { t_relay(); exit; } exit; } }
That's strange. You shouldn't allow requests with totag but without loose-routing. But that's not related to your problem.
Could that cause some problems for me? If it has totag and have a route it should exit, else it should check if the current request is associated to a request, if ok it should relay else exit.
How would you have written it?
IMO the config is a bit strange but I couldn't spot an error.
I'd really hoped for an error here. :\
Trace the scenario with ngrep and take a look at the SDPs if they are rewritten properly.
I'm not able to spot any errors, the only error I'm able to see is that it seems to work one way.
If A sends a fax to B
This works with T.38: asterisk A <-> sipprovider <-> our trunk <-> kamailio/rtpp <-> asterisk <-> asterisk B (our asterisk)
This fails with T.38, but works with G.711: asterisk B -> asterisk -> kamailio/rtpp -> our trunk -> asterisk -> asterisk A
Could also be an Asterisk (A: 1.6.2.5/tested with 1.6.2.7rc, B: 1.6.2.6) or FFA bug. But if I try to send the FAX directly via SIP from A <-> B without the proxy everything works OK, so therefore I believe the problem are related to my kamailio configuration.
Appreciate all the help I can get.
Espen.
Am 15.04.2010 13:05, schrieb Espen Berg:
That's strange. You shouldn't allow requests with totag but without loose-routing. But that's not related to your problem.
Could that cause some problems for me? If it has totag and have a route it should exit, else it should check if the current request is associated to a request, if ok it should relay else exit.
How would you have written it?
I prefer this logic:
... if (loose_route()) { if (!has_totag()) { xlog("L_WARN","$ci loose_route request without to-tag, 403...\n"); sl_send_reply("403", "out-of-dialog loose_route not allowed"); exit; } ... loose..route..processing ... NAT traversal...rtpproxy.... t_relay(); } ... initial call setup ... ...
IMO the config is a bit strange but I couldn't spot an error.
I'd really hoped for an error here. :\
Trace the scenario with ngrep and take a look at the SDPs if they are rewritten properly.
I'm not able to spot any errors, the only error I'm able to see is that it seems to work one way.
If A sends a fax to B
This works with T.38: asterisk A <-> sipprovider <-> our trunk <-> kamailio/rtpp <-> asterisk <-> asterisk B (our asterisk)
This fails with T.38, but works with G.711: asterisk B -> asterisk -> kamailio/rtpp -> our trunk -> asterisk -> asterisk A
Could also be an Asterisk (A: 1.6.2.5/tested with 1.6.2.7rc, B: 1.6.2.6) or FFA bug. But if I try to send the FAX directly via SIP from A <-> B without the proxy everything works OK, so therefore I believe the problem are related to my kamailio configuration.
Post the ngrep dump of the 2 different processes.
Furter use "ngrep udp" on the Kamailio/rtpproxy server to verify RTP/UDPTL packet flows.
regards klaus
Appreciate all the help I can get.
Espen.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Den 15.04.2010 13:27, skrev Klaus Darilion:
How would you have written it?
I prefer this logic:
... if (loose_route()) { if (!has_totag()) { xlog("L_WARN","$ci loose_route request without to-tag, 403...\n"); sl_send_reply("403", "out-of-dialog loose_route not allowed"); exit; } ... loose..route..processing ... NAT traversal...rtpproxy.... t_relay(); } ... initial call setup ... ...
Nice :)
Trace the scenario with ngrep and take a look at the SDPs if they are rewritten properly.
I'm not able to spot any errors, the only error I'm able to see is that it seems to work one way.
If A sends a fax to B
This works with T.38: asterisk A <-> sipprovider <-> our trunk <-> kamailio/rtpp <-> asterisk <-> asterisk B (our asterisk)
This fails with T.38, but works with G.711: asterisk B -> asterisk -> kamailio/rtpp -> our trunk -> asterisk -> asterisk A
Could also be an Asterisk (A: 1.6.2.5/tested with 1.6.2.7rc, B: 1.6.2.6) or FFA bug. But if I try to send the FAX directly via SIP from A <-> B without the proxy everything works OK, so therefore I believe the problem are related to my kamailio configuration.
Post the ngrep dump of the 2 different processes.
I have been debugging this problem little bit more. The re-invite test showed precisely the same with x-lite on hold as with t.38 re-invite.
If I call out and place a call on hold, I'm able to continue the call for about 20 seconds until the call was aborted. If I call into the same extension there are no problems to continue the call after "on hold". So just as the fax, the problem only occurs for outgoing calls, not inbound. I also tested with a cellular phone, just to eliminate the other end.
Here are some ncat captures of the t.38 scenario that fails.
ASTERISK B:
#############################################################################################################################$ U PUBLIC-IP-TO-PROXY:5060 -> IP-TO-ASTERISK:5060 INVITE sip:FAXNUMBER@IP-TO-ASTERISK:5060 SIP/2.0..Via: SIP/2.0/UDP PUBLIC-IP-TO-PROXY;branch=z9hG4bK61a.7e6809b3.0..Vi a: SIP/2.0/UDP IP-TO-SONUS-TRUNK:5060;branch=z9hG4bK02B5afc10d34f983f6e..From: sip:SENDTONUMBER@PUBLIC-IP-TO-PROXY;tag=gK02b1 6fe7..To: "FAX" sip:FAXNUMBER@PUBLIC-IP-TO-PROXY;tag=as56fa57a3..Call-ID: 15180894487146a33b3f3bac366bc0bb@PUBLIC-IP-TO-PROXY ..CSeq: 17055 INVITE..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDAT E,OPTIONS,MESSAGE,PUBLISH..Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed..Contact: sip:SENDTONUMBER@IP-TO-SONUS-TRUNK:5060..Supported: timer..Session-Expires: 1800;refresher=uas. .Min-SE: 90..Content-Length: 323..Content-Disposition: session; handling=required..Content-Type: application/sdp ....v=0..o=Sonus_UAC 23586 20272 IN IP4 IP-TO-SONUS-TRUNK..s=SIP Media Capabilities..c=IN IP4 PUBLIC-IP-TO-PROXY..t=0 0..m= image 35298 udptl t38..a=T38FaxVersion:0..a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxM
axBuffer:262..a=T38FaxMaxDatagram:176..a=T38FaxUdpEC:t38UDPRedundancy..a=sendrecv..a=nortpproxy:yes.. ### U IP-TO-ASTERISK:5060 -> PUBLIC-IP-TO-PROXY:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP PUBLIC-IP-TO-PROXY;branch=z9hG4bK61a.7e6809b3.0;received=PUBLIC-IP-TO-PROXY..Via: SIP/2. 0/UDP IP-TO-SONUS-TRUNK:5060;branch=z9hG4bK02B5afc10d34f983f6e..From: sip:SENDTONUMBER@PUBLIC-IP-TO-PROXY;tag=gK02b16fe7..To: "FAX" sip:FAXNUMBER@PUBLIC-IP-TO-PROXY;tag=as56fa57a3..Call-ID: 15180894487146a33b3f3bac366bc0bb@PUBLIC-IP-TO-PROXY..CS eq: 17055 INVITE..Server: Asterisk PBX 1.6.2.6..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIF Y, INFO..Supported: replaces, timer..Require: timer..Session-Expires: 1800;refresher=uas..Contact: <sip:FAXNUMBER@ IP-TO-ASTERISK>..Content-Type: application/sdp..Content-Length: 276....v=0..o=root 1138957066 1138957067 IN IP4 IP-TO-ASTERISK ..s=Asterisk PBX 1.6.2.6..c=IN IP4 IP-TO-ASTERISK..t=0 0..m=image 4400 udptl t38..a=T38FaxVersion:0.
.a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxDatagram:1400..a=T38FaxUdpEC:t38UDPRedu ndancy.. #############################################################################################################################$
KAMAILIO PROXY ETH0:
U PUBLIC-IP-TO-PROXY:5060 -> IP-TO-ASTERISK:5060 INVITE sip:FAXNUMBER@IP-TO-ASTERISK:5060 SIP/2.0..Via: SIP/2.0/UDP PUBLIC-IP-TO-PROXY;branch=z9hG4bK6d41.d083efb2.0..Via: SIP/2.0/UDP IP-TO-SONUS-TRUNK:5060;branch=z9hG4bK02B44d12570046e38f1..From: sip:SENDTONUMBER@PUBLIC-IP-TO-PROXY;tag=gK02ec0fad. .To: "FAX" sip:FAXNUMBER@PUBLIC-IP-TO-PROXY;tag=as6196c897..Call-ID: 58b69aef10209bf30e38c5371ebb29cf@PUBLIC-IP-TO-PROXY..CSeq: 15775 INVITE..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,M ESSAGE,PUBLISH..Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed..Contact: sip:SENDTONUMBER@IP-TO-SONUS-TRUNK:5060..Supported: timer..Session-Expires: 1800;refresher=uas..Min-SE: 90..Cont ent-Length: 322..Content-Disposition: session; handling=required..Content-Type: application/sdp....v=0..o=Sonus_UAC 9775 15421 IN IP4 IP-TO-SONUS-TRUNK..s=SIP Media Capabilities..c=IN IP4 PUBLIC-IP-TO-PROXY..t=0 0..m=image 35292 udptl t38..a
=T38FaxVersion:0..a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxBuffer:262..a=T38FaxMaxDatagram:176..a=T38FaxUdpEC:t38UDPRedundancy..a=sendrecv..a=nortpproxy:yes..
### U IP-TO-ASTERISK:5060 -> PUBLIC-IP-TO-PROXY:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP PUBLIC-IP-TO-PROXY;branch=z9hG4bK6d41.d083efb2.0;received=PUBLIC-IP-TO-PROXY..Via: SIP/2.0/UDP IP-TO-SONUS-TRUNK:5060;branch=z9hG4bK02B44d12570046e38f1..From: sip:SENDTONUMBER@PUBLIC-IP-TO-PROXY;tag=gK02ec0fad..To: "FAX " sip:FAXNUMBER@PUBLIC-IP-TO-PROXY;tag=as6196c897..Call-ID: 58b69aef10209bf30e38c5371ebb29cf@PUBLIC-IP-TO-PROXY..CSeq: 15775 INVITE..Server: Asterisk PBX 1.6.2.6..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Su pported: replaces, timer..Require: timer..Session-Expires: 1800;refresher=uas..Contact: sip:FAXNUMBER@IP-TO-ASTERISK..Content-Type: application/sdp..Content-Length: 276....v=0..o=root 1675681328 1675681329 IN IP4 IP-TO-ASTERISK..s= Asterisk PBX 1.6.2.6..c=IN IP4 IP-TO-ASTERISK..t=0 0..m=image 4852 udptl t38..a=T38FaxVersion:0..a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxDatagram:1400..a=T38FaxUdpEC:t38UDPRedundancy..
KAMAILIO PROXY ETH1:
U IP-TO-SONUS-TRUNK:5060 -> LOCAL-IP-TO-PROXY:5060 INVITE sip:FAXNUMBER@IP-TO-ASTERISK:5060 SIP/2.0..Via: SIP/2.0/UDP IP-TO-SONUS-TRUNK:5060;branch=z9hG4bK02B48cc1116e608d4ed..From: sip:SENDTONUMBER@PUBLIC-IP-TO-PROXY;tag=gK02f86d27..To: "FAX" sip:FAXNUMBER@PUBLIC-IP-TO-PROXY;tag=as279d4a05..Call-I D: 101766785455b9020d350d300cb15082@PUBLIC-IP-TO-PROXY..CSeq: 24873 INVITE..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH..Accept: application/sdp, application/isup, appl ication/dtmf, application/dtmf-relay, multipart/mixed..Contact: sip:SENDTONUMBER@IP-TO-SONUS-TRUNK:5060..Route: sip:LOCAL-IP-TO-PROXY:5060;r2=on;lr;ftag=as279d4a05..Route: sip:PUBLIC-IP-TO-PROXY:5060;r2=on;lr;ftag=as279d4a05..Supported: timer.. Session-Expires: 1800;refresher=uas..Min-SE: 90..Content-Length: 301..Content-Disposition: session; handling=required..Content-Type: application/sdp....v=0..o=Sonus_UAC 18853 20363 IN IP4 IP-TO-SONUS-TRUNK..s=SIP Media Capabilities..c=IN IP4 RTP-IP-TO-SONUSTRUNK..t=0 0..m=image 7994 udptl t38..a=T38FaxVersion:0..a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxBuffer:262..a=T38FaxMaxDatagram:176..a=T38FaxUdpEC:t38UDPRedundancy..a=sendrecv.. #### U LOCAL-IP-TO-PROXY:5060 -> IP-TO-SONUS-TRUNK:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP IP-TO-SONUS-TRUNK:5060;branch=z9hG4bK02B48cc1116e608d4ed..From: sip:SENDTONUMBER@PUBLIC-IP-TO-PROXY;tag=gK02f86d27..To: "FAX" sip:FAXNUMBER@PUBLIC-IP-TO-PROXY;tag=as279d4a05..Call-ID: 101766785455b9020d350d300cb150 82@PUBLIC-IP-TO-PROXY..CSeq: 24873 INVITE..Server: Asterisk PBX 1.6.2.6..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces, timer..Require: timer..Session-Expires: 1800;refresher=uas..Contact: sip:FAXNUMBER@IP-TO-ASTERISK..Content-Type: application/sdp..Content-Length: 293....v=0..o=root 1135418376 1135418377 IN IP4 IP-TO-ASTERISK..s=Asterisk PBX 1.6.2.6..c=IN IP4 LOCAL-IP-TO-PROXY..t=0 0..m=image 35324 udptl t38..a=T38FaxVe
rsion:0..a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxDatagram:1400..a=T38FaxUdpEC:t38UDPRedundancy..a=nortpproxy:yes..
U IP-TO-SONUS-TRUNK:5060 -> LOCAL-IP-TO-PROXY:5060 INVITE sip:FAXNUMBER@IP-TO-ASTERISK:5060 SIP/2.0..Via: SIP/2.0/UDP IP-TO-SONUS-TRUNK:5060;branch=z9hG4bK02B48cc1116e608d4ed..From: sip:SENDTONUMBER@PUBLIC-IP-TO-PROXY;tag=gK02f86d27..To: "FAX" sip:FAXNUMBER@PUBLIC-IP-TO-PROXY;tag=as279d4a05..Call-I D: 101766785455b9020d350d300cb15082@PUBLIC-IP-TO-PROXY..CSeq: 24873 INVITE..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH..Accept: application/sdp, application/isup, appl ication/dtmf, application/dtmf-relay, multipart/mixed..Contact: sip:SENDTONUMBER@IP-TO-SONUS-TRUNK:5060..Route: sip:LOCAL-IP-TO-PROXY:5060;r2=on;lr;ftag=as279d4a05..Route: sip:PUBLIC-IP-TO-PROXY:5060;r2=on;lr;ftag=as279d4a05..Supported: timer.. Session-Expires: 1800;refresher=uas..Min-SE: 90..Content-Length: 301..Content-Disposition: session; handling=required..Content-Type: application/sdp....v=0..o=Sonus_UAC 18853 20363 IN IP4 IP-TO-SONUS-TRUNK..s=SIP Media Capabilities..c=IN IP4 RTP-IP-TO-SONUSTRUNK..t=0 0..m=image 7994 udptl t38..a=T38FaxVersion:0..a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxBuffer:262..a=T38FaxMaxDatagram:176..a=T38FaxUdpEC:t38UDPRedundancy..a=sendrecv.. #### U LOCAL-IP-TO-PROXY:5060 -> IP-TO-SONUS-TRUNK:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP IP-TO-SONUS-TRUNK:5060;branch=z9hG4bK02B48cc1116e608d4ed..From: sip:SENDTONUMBER@PUBLIC-IP-TO-PROXY;tag=gK02f86d27..To: "FAX" sip:FAXNUMBER@PUBLIC-IP-TO-PROXY;tag=as279d4a05..Call-ID: 101766785455b9020d350d300cb150 82@PUBLIC-IP-TO-PROXY..CSeq: 24873 INVITE..Server: Asterisk PBX 1.6.2.6..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces, timer..Require: timer..Session-Expires: 1800;refresher=uas..Contact: sip:FAXNUMBER@IP-TO-ASTERISK..Content-Type: application/sdp..Content-Length: 293....v=0..o=root 1135418376 1135418377 IN IP4 IP-TO-ASTERISK..s=Asterisk PBX 1.6.2.6..c=IN IP4 LOCAL-IP-TO-PROXY..t=0 0..m=image 35324 udptl t38..a=T38FaxVe
rsion:0..a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxDatagram:1400..a=T38FaxUdpEC:t38UDPRedundancy..a=nortpproxy:yes..
ASTERISK A: (REMOTE TESTSITE CONNECTED TO PHONZO, ONE PROVIDER KNOWN TO PROVIDE FUNCTIONING T.38 SERVICES)
######################## U ASTERISK-TESTSITE-IP:5060 -> 80.232.37.178:5060 INVITE sip:80.232.37.178:5061 SIP/2.0..Via: SIP/2.0/UDP 85.165.146.221:5060;branch=z9hG4bK7726391b;rport..Route: <sip:80.232.37 .178;ftag=b1ae9fe1eb7f13837782c9061d653c57;lr>..Max-Forwards: 70..From: sip:47SENDTONUMBER@80.232.37.178;tag=as61e4a63c..To: 6970 8347 sip:FAXNUMBER@80.232.37.178;tag=b1ae9fe1eb7f13837782c9061d653c57..Contact: sip:SENDTONUMBER@85.165.146.221..Call-ID: 3D1965 63-47C711DF-AC2A9E0C-AB1DB137@80.232.37.179..CSeq: 102 INVITE..User-Agent: none..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFE R, SUBSCRIBE, NOTIFY, INFO..Supported: replaces, timer..Content-Type: application/sdp..Content-Length: 268....v=0..o=root 10588 12562 1058812563 IN IP4 85.165.146.221..s=Asterisk PBX 1.6.2.5..c=IN IP4 85.165.146.221..t=0 0..m=image 4810 udptl t38..a=T38Fa
xVersion:0..a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxDatagram:849..a=T38FaxUdpEC:t38UDPFEC..
###### U 80.232.37.178:5060 -> ASTERISK-TESTSITE-IP:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 85.165.146.221:5060;branch=z9hG4bK7726391b;rport=5060..Record-Route: <sip:80.232.37.178;ftag=a s61e4a63c;lr>..From: sip:47SENDTONUMBER@80.232.37.178;tag=as61e4a63c..To: FAXNUMBER sip:FAXNUMBER@80.232.37.178;tag=b1ae9fe1eb7f1 3837782c9061d653c57..Call-ID: 3D196563-47C711DF-AC2A9E0C-AB1DB137@80.232.37.179..CSeq: 102 INVITE..Server: Sippy..Contact: Anon ymous sip:80.232.37.178:5061..Content-Length: 220..Content-Type: application/sdp....v=0..o=Sippy 144528972 1 IN IP4 80.232.37 .178..s=SIP Call..t=0 0..m=image 17772 udptl t38..c=IN IP4 80.232.37.179..a=T38FaxVersion:0..a=T38MaxBitRate:14400..a=T38FaxRat eManagement:transferredTCF..a=T38FaxMaxDatagram:72..
Hi Espen, can you provide the traces with proper line break?
E.g. I always use
"ngrep -W byline -t -q -P "" any-pattery port 5060"
regards klaus
Am 17.04.2010 21:39, schrieb Espen Berg:
Den 15.04.2010 13:27, skrev Klaus Darilion:
How would you have written it?
I prefer this logic:
... if (loose_route()) { if (!has_totag()) { xlog("L_WARN","$ci loose_route request without to-tag, 403...\n"); sl_send_reply("403", "out-of-dialog loose_route not allowed"); exit; } ... loose..route..processing ... NAT traversal...rtpproxy.... t_relay(); } ... initial call setup ... ...
Nice :)
Trace the scenario with ngrep and take a look at the SDPs if they are rewritten properly.
I'm not able to spot any errors, the only error I'm able to see is that it seems to work one way.
If A sends a fax to B
This works with T.38: asterisk A <-> sipprovider <-> our trunk <-> kamailio/rtpp <-> asterisk <-> asterisk B (our asterisk)
This fails with T.38, but works with G.711: asterisk B -> asterisk -> kamailio/rtpp -> our trunk -> asterisk -> asterisk A
Could also be an Asterisk (A: 1.6.2.5/tested with 1.6.2.7rc, B: 1.6.2.6) or FFA bug. But if I try to send the FAX directly via SIP from A <-> B without the proxy everything works OK, so therefore I believe the problem are related to my kamailio configuration.
Post the ngrep dump of the 2 different processes.
I have been debugging this problem little bit more. The re-invite test showed precisely the same with x-lite on hold as with t.38 re-invite.
If I call out and place a call on hold, I'm able to continue the call for about 20 seconds until the call was aborted. If I call into the same extension there are no problems to continue the call after "on hold". So just as the fax, the problem only occurs for outgoing calls, not inbound. I also tested with a cellular phone, just to eliminate the other end.
Here are some ncat captures of the t.38 scenario that fails.
ASTERISK B:
#############################################################################################################################$
U PUBLIC-IP-TO-PROXY:5060 -> IP-TO-ASTERISK:5060 INVITE sip:FAXNUMBER@IP-TO-ASTERISK:5060 SIP/2.0..Via: SIP/2.0/UDP PUBLIC-IP-TO-PROXY;branch=z9hG4bK61a.7e6809b3.0..Vi a: SIP/2.0/UDP IP-TO-SONUS-TRUNK:5060;branch=z9hG4bK02B5afc10d34f983f6e..From: sip:SENDTONUMBER@PUBLIC-IP-TO-PROXY;tag=gK02b1 6fe7..To: "FAX" sip:FAXNUMBER@PUBLIC-IP-TO-PROXY;tag=as56fa57a3..Call-ID: 15180894487146a33b3f3bac366bc0bb@PUBLIC-IP-TO-PROXY ..CSeq: 17055 INVITE..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDAT E,OPTIONS,MESSAGE,PUBLISH..Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed..Contact: sip:SENDTONUMBER@IP-TO-SONUS-TRUNK:5060..Supported: timer..Session-Expires: 1800;refresher=uas. .Min-SE: 90..Content-Length: 323..Content-Disposition: session; handling=required..Content-Type: application/sdp ....v=0..o=Sonus_UAC 23586 20272 IN IP4 IP-TO-SONUS-TRUNK..s=SIP Media Capabilities..c=IN IP4 PUBLIC-IP-TO-PROXY..t=0 0..m= image 35298 udptl t38..a=T38FaxVersion:0..a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxM
axBuffer:262..a=T38FaxMaxDatagram:176..a=T38FaxUdpEC:t38UDPRedundancy..a=sendrecv..a=nortpproxy:yes..
### U IP-TO-ASTERISK:5060 -> PUBLIC-IP-TO-PROXY:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP PUBLIC-IP-TO-PROXY;branch=z9hG4bK61a.7e6809b3.0;received=PUBLIC-IP-TO-PROXY..Via: SIP/2. 0/UDP IP-TO-SONUS-TRUNK:5060;branch=z9hG4bK02B5afc10d34f983f6e..From: sip:SENDTONUMBER@PUBLIC-IP-TO-PROXY;tag=gK02b16fe7..To: "FAX" sip:FAXNUMBER@PUBLIC-IP-TO-PROXY;tag=as56fa57a3..Call-ID: 15180894487146a33b3f3bac366bc0bb@PUBLIC-IP-TO-PROXY..CS eq: 17055 INVITE..Server: Asterisk PBX 1.6.2.6..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIF Y, INFO..Supported: replaces, timer..Require: timer..Session-Expires: 1800;refresher=uas..Contact: sip:FAXNUMBER@ IP-TO-ASTERISK..Content-Type: application/sdp..Content-Length: 276....v=0..o=root 1138957066 1138957067 IN IP4 IP-TO-ASTERISK ..s=Asterisk PBX 1.6.2.6..c=IN IP4 IP-TO-ASTERISK..t=0 0..m=image 4400 udptl t38..a=T38FaxVersion:0.
.a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxDatagram:1400..a=T38FaxUdpEC:t38UDPRedu
ndancy.. #############################################################################################################################$
KAMAILIO PROXY ETH0:
U PUBLIC-IP-TO-PROXY:5060 -> IP-TO-ASTERISK:5060 INVITE sip:FAXNUMBER@IP-TO-ASTERISK:5060 SIP/2.0..Via: SIP/2.0/UDP PUBLIC-IP-TO-PROXY;branch=z9hG4bK6d41.d083efb2.0..Via: SIP/2.0/UDP IP-TO-SONUS-TRUNK:5060;branch=z9hG4bK02B44d12570046e38f1..From: sip:SENDTONUMBER@PUBLIC-IP-TO-PROXY;tag=gK02ec0fad. .To: "FAX" sip:FAXNUMBER@PUBLIC-IP-TO-PROXY;tag=as6196c897..Call-ID: 58b69aef10209bf30e38c5371ebb29cf@PUBLIC-IP-TO-PROXY..CSeq: 15775 INVITE..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,M ESSAGE,PUBLISH..Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed..Contact: sip:SENDTONUMBER@IP-TO-SONUS-TRUNK:5060..Supported: timer..Session-Expires: 1800;refresher=uas..Min-SE: 90..Cont ent-Length: 322..Content-Disposition: session; handling=required..Content-Type: application/sdp....v=0..o=Sonus_UAC 9775 15421 IN IP4 IP-TO-SONUS-TRUNK..s=SIP Media Capabilities..c=IN IP4 PUBLIC-IP-TO-PROXY..t=0 0..m=image 35292 udptl t38..a
=T38FaxVersion:0..a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxBuffer:262..a=T38FaxMaxDatagram:176..a=T38FaxUdpEC:t38UDPRedundancy..a=sendrecv..a=nortpproxy:yes..
### U IP-TO-ASTERISK:5060 -> PUBLIC-IP-TO-PROXY:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP PUBLIC-IP-TO-PROXY;branch=z9hG4bK6d41.d083efb2.0;received=PUBLIC-IP-TO-PROXY..Via: SIP/2.0/UDP IP-TO-SONUS-TRUNK:5060;branch=z9hG4bK02B44d12570046e38f1..From: sip:SENDTONUMBER@PUBLIC-IP-TO-PROXY;tag=gK02ec0fad..To: "FAX " sip:FAXNUMBER@PUBLIC-IP-TO-PROXY;tag=as6196c897..Call-ID: 58b69aef10209bf30e38c5371ebb29cf@PUBLIC-IP-TO-PROXY..CSeq: 15775 INVITE..Server: Asterisk PBX 1.6.2.6..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Su pported: replaces, timer..Require: timer..Session-Expires: 1800;refresher=uas..Contact: sip:FAXNUMBER@IP-TO-ASTERISK..Content-Type: application/sdp..Content-Length: 276....v=0..o=root 1675681328 1675681329 IN IP4 IP-TO-ASTERISK..s= Asterisk PBX 1.6.2.6..c=IN IP4 IP-TO-ASTERISK..t=0 0..m=image 4852 udptl t38..a=T38FaxVersion:0..a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxDatagram:1400..a=T38FaxUdpEC:t38UDPRedundancy..
KAMAILIO PROXY ETH1:
U IP-TO-SONUS-TRUNK:5060 -> LOCAL-IP-TO-PROXY:5060 INVITE sip:FAXNUMBER@IP-TO-ASTERISK:5060 SIP/2.0..Via: SIP/2.0/UDP IP-TO-SONUS-TRUNK:5060;branch=z9hG4bK02B48cc1116e608d4ed..From: sip:SENDTONUMBER@PUBLIC-IP-TO-PROXY;tag=gK02f86d27..To: "FAX" sip:FAXNUMBER@PUBLIC-IP-TO-PROXY;tag=as279d4a05..Call-I D: 101766785455b9020d350d300cb15082@PUBLIC-IP-TO-PROXY..CSeq: 24873 INVITE..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH..Accept: application/sdp, application/isup, appl ication/dtmf, application/dtmf-relay, multipart/mixed..Contact: sip:SENDTONUMBER@IP-TO-SONUS-TRUNK:5060..Route: sip:LOCAL-IP-TO-PROXY:5060;r2=on;lr;ftag=as279d4a05..Route: sip:PUBLIC-IP-TO-PROXY:5060;r2=on;lr;ftag=as279d4a05..Supported: timer.. Session-Expires: 1800;refresher=uas..Min-SE: 90..Content-Length: 301..Content-Disposition: session; handling=required..Content-Type: application/sdp....v=0..o=Sonus_UAC 18853 20363 IN IP4 IP-TO-SONUS-TRUNK..s=SIP Media Capabilities..c=IN IP4 RTP-IP-TO-SONUSTRUNK..t=0 0..m=image 7994 udptl t38..a=T38FaxVersion:0..a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxBuffer:262..a=T38FaxMaxDatagram:176..a=T38FaxUdpEC:t38UDPRedundancy..a=sendrecv..
#### U LOCAL-IP-TO-PROXY:5060 -> IP-TO-SONUS-TRUNK:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP IP-TO-SONUS-TRUNK:5060;branch=z9hG4bK02B48cc1116e608d4ed..From: sip:SENDTONUMBER@PUBLIC-IP-TO-PROXY;tag=gK02f86d27..To: "FAX" sip:FAXNUMBER@PUBLIC-IP-TO-PROXY;tag=as279d4a05..Call-ID: 101766785455b9020d350d300cb150 82@PUBLIC-IP-TO-PROXY..CSeq: 24873 INVITE..Server: Asterisk PBX 1.6.2.6..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces, timer..Require: timer..Session-Expires: 1800;refresher=uas..Contact: sip:FAXNUMBER@IP-TO-ASTERISK..Content-Type: application/sdp..Content-Length: 293....v=0..o=root 1135418376 1135418377 IN IP4 IP-TO-ASTERISK..s=Asterisk PBX 1.6.2.6..c=IN IP4 LOCAL-IP-TO-PROXY..t=0 0..m=image 35324 udptl t38..a=T38FaxVe
rsion:0..a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxDatagram:1400..a=T38FaxUdpEC:t38UDPRedundancy..a=nortpproxy:yes..
U IP-TO-SONUS-TRUNK:5060 -> LOCAL-IP-TO-PROXY:5060 INVITE sip:FAXNUMBER@IP-TO-ASTERISK:5060 SIP/2.0..Via: SIP/2.0/UDP IP-TO-SONUS-TRUNK:5060;branch=z9hG4bK02B48cc1116e608d4ed..From: sip:SENDTONUMBER@PUBLIC-IP-TO-PROXY;tag=gK02f86d27..To: "FAX" sip:FAXNUMBER@PUBLIC-IP-TO-PROXY;tag=as279d4a05..Call-I D: 101766785455b9020d350d300cb15082@PUBLIC-IP-TO-PROXY..CSeq: 24873 INVITE..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH..Accept: application/sdp, application/isup, appl ication/dtmf, application/dtmf-relay, multipart/mixed..Contact: sip:SENDTONUMBER@IP-TO-SONUS-TRUNK:5060..Route: sip:LOCAL-IP-TO-PROXY:5060;r2=on;lr;ftag=as279d4a05..Route: sip:PUBLIC-IP-TO-PROXY:5060;r2=on;lr;ftag=as279d4a05..Supported: timer.. Session-Expires: 1800;refresher=uas..Min-SE: 90..Content-Length: 301..Content-Disposition: session; handling=required..Content-Type: application/sdp....v=0..o=Sonus_UAC 18853 20363 IN IP4 IP-TO-SONUS-TRUNK..s=SIP Media Capabilities..c=IN IP4 RTP-IP-TO-SONUSTRUNK..t=0 0..m=image 7994 udptl t38..a=T38FaxVersion:0..a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxBuffer:262..a=T38FaxMaxDatagram:176..a=T38FaxUdpEC:t38UDPRedundancy..a=sendrecv..
#### U LOCAL-IP-TO-PROXY:5060 -> IP-TO-SONUS-TRUNK:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP IP-TO-SONUS-TRUNK:5060;branch=z9hG4bK02B48cc1116e608d4ed..From: sip:SENDTONUMBER@PUBLIC-IP-TO-PROXY;tag=gK02f86d27..To: "FAX" sip:FAXNUMBER@PUBLIC-IP-TO-PROXY;tag=as279d4a05..Call-ID: 101766785455b9020d350d300cb150 82@PUBLIC-IP-TO-PROXY..CSeq: 24873 INVITE..Server: Asterisk PBX 1.6.2.6..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces, timer..Require: timer..Session-Expires: 1800;refresher=uas..Contact: sip:FAXNUMBER@IP-TO-ASTERISK..Content-Type: application/sdp..Content-Length: 293....v=0..o=root 1135418376 1135418377 IN IP4 IP-TO-ASTERISK..s=Asterisk PBX 1.6.2.6..c=IN IP4 LOCAL-IP-TO-PROXY..t=0 0..m=image 35324 udptl t38..a=T38FaxVe
rsion:0..a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxDatagram:1400..a=T38FaxUdpEC:t38UDPRedundancy..a=nortpproxy:yes..
ASTERISK A: (REMOTE TESTSITE CONNECTED TO PHONZO, ONE PROVIDER KNOWN TO PROVIDE FUNCTIONING T.38 SERVICES)
######################## U ASTERISK-TESTSITE-IP:5060 -> 80.232.37.178:5060 INVITE sip:80.232.37.178:5061 SIP/2.0..Via: SIP/2.0/UDP 85.165.146.221:5060;branch=z9hG4bK7726391b;rport..Route: sip:80.232.37 .178;ftag=b1ae9fe1eb7f13837782c9061d653c57;lr..Max-Forwards: 70..From: sip:47SENDTONUMBER@80.232.37.178;tag=as61e4a63c..To: 6970 8347 sip:FAXNUMBER@80.232.37.178;tag=b1ae9fe1eb7f13837782c9061d653c57..Contact: sip:SENDTONUMBER@85.165.146.221..Call-ID: 3D1965 63-47C711DF-AC2A9E0C-AB1DB137@80.232.37.179..CSeq: 102 INVITE..User-Agent: none..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFE R, SUBSCRIBE, NOTIFY, INFO..Supported: replaces, timer..Content-Type: application/sdp..Content-Length: 268....v=0..o=root 10588 12562 1058812563 IN IP4 85.165.146.221..s=Asterisk PBX 1.6.2.5..c=IN IP4 85.165.146.221..t=0 0..m=image 4810 udptl t38..a=T38Fa
xVersion:0..a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxDatagram:849..a=T38FaxUdpEC:t38UDPFEC..
###### U 80.232.37.178:5060 -> ASTERISK-TESTSITE-IP:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 85.165.146.221:5060;branch=z9hG4bK7726391b;rport=5060..Record-Route: sip:80.232.37.178;ftag=a s61e4a63c;lr..From: sip:47SENDTONUMBER@80.232.37.178;tag=as61e4a63c..To: FAXNUMBER sip:FAXNUMBER@80.232.37.178;tag=b1ae9fe1eb7f1 3837782c9061d653c57..Call-ID: 3D196563-47C711DF-AC2A9E0C-AB1DB137@80.232.37.179..CSeq: 102 INVITE..Server: Sippy..Contact: Anon ymous sip:80.232.37.178:5061..Content-Length: 220..Content-Type: application/sdp....v=0..o=Sippy 144528972 1 IN IP4 80.232.37 .178..s=SIP Call..t=0 0..m=image 17772 udptl t38..c=IN IP4 80.232.37.179..a=T38FaxVersion:0..a=T38MaxBitRate:14400..a=T38FaxRat eManagement:transferredTCF..a=T38FaxMaxDatagram:72..
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Den 20.04.2010 10:52, skrev Klaus Darilion:
Hi Espen, can you provide the traces with proper line break?
E.g. I always use
"ngrep -W byline -t -q -P "" any-pattery port 5060"
It seems that Kamailio/RDP-proxy where not the root of my problems, the new FFA from Digium partly solved the issue.
Thanks for all help!
Espen
-
Espen Berg -
Klaus Darilion