Dear Alexandr,

You can connect Kamailio to RTPproxy via socket as well, use modparam like this:

modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:12221")

Then if your rtprpoxy is started in bridged mode you should use the "i" and "e" flags while you call the rtpproxy-manage() function in the kamailio.cfg file.

The placement of both the flags sets the SDP c= param , so if you use "ie" combination of flag then that is not equal to "ei" combination of the flag.

I also suggest that you turn on sip debug on the call receiving asterisk and observe the SDP for an incoming call from Kamailio. that will help you figure out the situation in SDP.

Best Regards, Sammy

On Tue, Aug 6, 2013 at 2:02 AM, Alexandr Usov blessendor@gmail.com wrote:

Thank you for response! A little difficult for me to find the same logic in my case with tutorial of ipv4/ipv6 bridgin...

When I started /usr/sbin/rtpproxy -u kamailio -l 1.1.1.1/2.2.2.2 -s udp:127.0.0.1 12221

There is no sound.

Is this a major to connect via unix sock?:

modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock")

2013/8/6 Daniel-Constantin Mierla miconda@gmail.com

...

Hello,

you have to use rtpproxy in bridge mode, to route packets between the two local network interfaces. There are many examples out there, one shows even how to bridge between ipv4 and ipv4 networks -- you can use it as reference:

• http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6

Cheers, Daniel

On 8/5/13 7:12 PM, Alexandr Usov wrote:

I have Kamailio on OpenSUSE with static real Public IP (WAN), for ex. 1.1.1.1. I have LAN IP 2.2.2.2. Asterisk as KVM virtual machine with LAN IP 2.2.2.101 and default GW not the SuSe (2.2.2.2), but 2.2.2.1 pfsense LAN with PUB IP 1.1.1.2) I am configured Registration of UA on Kamailio DB, and on Asterisk side create a static peers with Kamailio LAN ip (host=2.2.2.2).

RTP Proxy question.

/usr/sbin/rtpproxy -u daemon -l 1.1.1.1 -s udp:127.0.0.1 12221

.... kamailio.cfg ...

# RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return;

    rtpproxy_manage();

if (is_request()) {
    if (!has_totag()) {
        add_rr_param(";nat=yes");
    }
}
if (is_reply()) {
    if(isbflagset(FLB_NATB)) {
        fix_nated_contact();
    }
}

#!endif return; }

Testing call:

Whe User 1-100 calling User 1-101, on Asterisk side I see:

-- Called SIP/1-100@sip1.somedomain.com.ua
-- SIP/sip1.somedomain.com.ua-000004cf is ringing
-- SIP/sip1.somedomain.com.ua-000004cf answered SIP/1-101-000004ce
   > 0x15bc370 -- Probation passed - setting RTP source address to

1.1.1.1:50868 > 0x7f2b6044bd10 -- Probation passed - setting RTP source address to 1.1.1.1:35082

Got RTP packet from 1.1.1.1:50868 (type 00, seq 027109, ts 000160, len 000160) Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037469, ts 000160, len 000160) Got RTP packet from 1.1.1.1:50868 (type 00, seq 027110, ts 000320, len 000160) Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037470, ts 000320, len 000160) Got RTP packet from 1.1.1.1:50868 (type 00, seq 027111, ts 000480, len 000160) Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037471, ts 000480, len 000160) Got RTP packet from 1.1.1.1:50868 (type 00, seq 027112, ts 000640, len 000160) Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037472, ts 000640, len 000160)

Voice transfers OK.

But why not Kamailio LAN ip I receiving on the Asterisk side with the same LAN?

And Kamailio log grep:

skynet:~ # tail -f /var/log/messages | grep rtpproxy 2013-08-05T19:18:17.508760+03:00 skynet kamailio[25462]: 3(25481) DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type <application/sdp> found valid 2013-08-05T19:18:17.508875+03:00 skynet kamailio[25462]: 3(25481) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 63566 1.1.1.1 2013-08-05T19:18:17.530765+03:00 skynet kamailio[25462]: 6(25484) DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type <application/sdp> found valid 2013-08-05T19:18:17.530876+03:00 skynet kamailio[25462]: 6(25484) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 41958 1.1.1.1 2013-08-05T19:18:18.625815+03:00 skynet kamailio[25462]: 4(25482) DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type <application/sdp> found valid 2013-08-05T19:18:18.627131+03:00 skynet kamailio[25462]: 4(25482) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 39876 1.1.1.1 2013-08-05T19:18:18.632649+03:00 skynet kamailio[25462]: a=nortpproxy:yes 2013-08-05T19:18:18.648075+03:00 skynet kamailio[25462]: 6(25484) DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type <application/sdp> found valid 2013-08-05T19:18:18.649615+03:00 skynet kamailio[25462]: 6(25484) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 43500 1.1.1.1 2013-08-05T19:18:18.653948+03:00 skynet kamailio[25462]: a=nortpproxy:yes 2013-08-05T19:18:18.688603+03:00 skynet kamailio[25462]: 4(25482) DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type <application/sdp> found valid 2013-08-05T19:18:18.689762+03:00 skynet kamailio[25462]: 4(25482) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 63566 1.1.1.1 2013-08-05T19:18:18.701062+03:00 skynet kamailio[25462]: 6(25484) DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type <application/sdp> found valid 2013-08-05T19:18:18.701405+03:00 skynet kamailio[25462]: 6(25484) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 43500 1.1.1.1 2013-08-05T19:18:18.705506+03:00 skynet kamailio[25462]: a=nortpproxy:yes

My goal is using Asterisk boxes behind Kamailio with the same LAN or even OpeVPN Lan2Lan, with Public IP on Kamailio WAN for users registration and RTP routing. So is strange to my, why RTPproxy not rewrite source of RTP traffic from PUBLIC Kamailio IP to LAN Kamailio IP when user A calls B via Asterisk?

---

SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

---

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

---

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

You should not change the kamailio.cfg for nat=yes param, that works the way it is. Yes you're right changing the NAT param in asterisk won't change anything.

Please enable sip debug on asterisk and paste the complete INVITE/200OK packets for the established call with no audio.

-- Sammy

On Tue, Aug 6, 2013 at 3:06 AM, Alexandr Usov blessendor@gmail.com wrote:

It seems I am undesrtand whereis problem can be found. Original tutorial of Kamailio+Asterisk realtime integration (by Asipto) containse settings for cheking if the "nat=yes" presents, but in Asterisk 11 I am using nat=force_rport,comedia.

# RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return;

    rtpproxy_manage();

if (is_request()) {
    if (!has_totag()) {
        add_rr_param(";nat=yes");
    }
}
if (is_reply()) {
    if(isbflagset(FLB_NATB)) {
        fix_nated_contact();
    }
}

#!endif return; }

BTW, we don't need NAT for asterisks peers, if we use Asterisk behinde Kanailio LAN interface (2.2.2.2).

ToHost : 2.2.2.2 Addr->IP : 2.2.2.2:5060

If I cahnge to nat=no in the NATMANAGE - RTP debug still showing from 1.1.1.1 (public) kamailio IP. Rtpproxy started in the bridge mode.

set_destination: Parsing sip:2.2.2.2;r2=on;lr=on;nat=no for address/port to send to set_destination: set destination to 2.2.2.2:5060 Reliably Transmitting (no NAT) to 2.2.2.2:5060: BYE sip:1-101@PUBLIC.CLIENT.PEER.IP:17303;ob SIP/2.0 Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK1597848e Route: sip:2.2.2.2;r2=on;lr=on;nat=no,sip:1.1.1.1;r2=on;lr=on;nat=no

Sent RTP packet to 1.1.1.1:63232 (type 00, seq 003837, ts 016800, len 000160) Got RTP packet from 2.2.2.2:42346 (type 00, seq 000986, ts 016960, len 000160) Sent RTP packet to 1.1.1.1:63232 (type 00, seq 003838, ts 016960, len 000160) Got RTP packet from 2.2.2.2:42346 (type 00, seq 000987, ts 017120, len 000160) Sent RTP packet to 1.1.1.1:63232 (type 00, seq 003839, ts 017120, len 000160)

But no voicing)

2013/8/6 SamyGo govoiper@gmail.com

...

Dear Alexandr,

You can connect Kamailio to RTPproxy via socket as well, use modparam like this:

modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:12221")

Then if your rtprpoxy is started in bridged mode you should use the "i" and "e" flags while you call the rtpproxy-manage() function in the kamailio.cfg file.

The placement of both the flags sets the SDP c= param , so if you use "ie" combination of flag then that is not equal to "ei" combination of the flag.

I also suggest that you turn on sip debug on the call receiving asterisk and observe the SDP for an incoming call from Kamailio. that will help you figure out the situation in SDP.

Best Regards, Sammy

On Tue, Aug 6, 2013 at 2:02 AM, Alexandr Usov blessendor@gmail.comwrote:

...

Thank you for response! A little difficult for me to find the same logic in my case with tutorial of ipv4/ipv6 bridgin...

When I started /usr/sbin/rtpproxy -u kamailio -l 1.1.1.1/2.2.2.2 -s udp:127.0.0.1 12221

There is no sound.

Is this a major to connect via unix sock?:

modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock")

2013/8/6 Daniel-Constantin Mierla miconda@gmail.com

...

Hello,

you have to use rtpproxy in bridge mode, to route packets between the two local network interfaces. There are many examples out there, one shows even how to bridge between ipv4 and ipv4 networks -- you can use it as reference:

• http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6

Cheers, Daniel

On 8/5/13 7:12 PM, Alexandr Usov wrote:

I have Kamailio on OpenSUSE with static real Public IP (WAN), for ex. 1.1.1.1. I have LAN IP 2.2.2.2. Asterisk as KVM virtual machine with LAN IP 2.2.2.101 and default GW not the SuSe (2.2.2.2), but 2.2.2.1 pfsense LAN with PUB IP 1.1.1.2) I am configured Registration of UA on Kamailio DB, and on Asterisk side create a static peers with Kamailio LAN ip (host=2.2.2.2).

RTP Proxy question.

/usr/sbin/rtpproxy -u daemon -l 1.1.1.1 -s udp:127.0.0.1 12221

.... kamailio.cfg ...

# RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return;

    rtpproxy_manage();

if (is_request()) {
    if (!has_totag()) {
        add_rr_param(";nat=yes");
    }
}
if (is_reply()) {
    if(isbflagset(FLB_NATB)) {
        fix_nated_contact();
    }
}

#!endif return; }

Testing call:

Whe User 1-100 calling User 1-101, on Asterisk side I see:

-- Called SIP/1-100@sip1.somedomain.com.ua
-- SIP/sip1.somedomain.com.ua-000004cf is ringing
-- SIP/sip1.somedomain.com.ua-000004cf answered SIP/1-101-000004ce
   > 0x15bc370 -- Probation passed - setting RTP source address to

1.1.1.1:50868 > 0x7f2b6044bd10 -- Probation passed - setting RTP source address to 1.1.1.1:35082

Got RTP packet from 1.1.1.1:50868 (type 00, seq 027109, ts 000160, len 000160) Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037469, ts 000160, len 000160) Got RTP packet from 1.1.1.1:50868 (type 00, seq 027110, ts 000320, len 000160) Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037470, ts 000320, len 000160) Got RTP packet from 1.1.1.1:50868 (type 00, seq 027111, ts 000480, len 000160) Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037471, ts 000480, len 000160) Got RTP packet from 1.1.1.1:50868 (type 00, seq 027112, ts 000640, len 000160) Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037472, ts 000640, len 000160)

Voice transfers OK.

But why not Kamailio LAN ip I receiving on the Asterisk side with the same LAN?

And Kamailio log grep:

skynet:~ # tail -f /var/log/messages | grep rtpproxy 2013-08-05T19:18:17.508760+03:00 skynet kamailio[25462]: 3(25481) DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type <application/sdp> found valid 2013-08-05T19:18:17.508875+03:00 skynet kamailio[25462]: 3(25481) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 63566 1.1.1.1 2013-08-05T19:18:17.530765+03:00 skynet kamailio[25462]: 6(25484) DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type <application/sdp> found valid 2013-08-05T19:18:17.530876+03:00 skynet kamailio[25462]: 6(25484) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 41958 1.1.1.1 2013-08-05T19:18:18.625815+03:00 skynet kamailio[25462]: 4(25482) DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type <application/sdp> found valid 2013-08-05T19:18:18.627131+03:00 skynet kamailio[25462]: 4(25482) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 39876 1.1.1.1 2013-08-05T19:18:18.632649+03:00 skynet kamailio[25462]: a=nortpproxy:yes 2013-08-05T19:18:18.648075+03:00 skynet kamailio[25462]: 6(25484) DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type <application/sdp> found valid 2013-08-05T19:18:18.649615+03:00 skynet kamailio[25462]: 6(25484) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 43500 1.1.1.1 2013-08-05T19:18:18.653948+03:00 skynet kamailio[25462]: a=nortpproxy:yes 2013-08-05T19:18:18.688603+03:00 skynet kamailio[25462]: 4(25482) DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type <application/sdp> found valid 2013-08-05T19:18:18.689762+03:00 skynet kamailio[25462]: 4(25482) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 63566 1.1.1.1 2013-08-05T19:18:18.701062+03:00 skynet kamailio[25462]: 6(25484) DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type <application/sdp> found valid 2013-08-05T19:18:18.701405+03:00 skynet kamailio[25462]: 6(25484) DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 43500 1.1.1.1 2013-08-05T19:18:18.705506+03:00 skynet kamailio[25462]: a=nortpproxy:yes

My goal is using Asterisk boxes behind Kamailio with the same LAN or even OpeVPN Lan2Lan, with Public IP on Kamailio WAN for users registration and RTP routing. So is strange to my, why RTPproxy not rewrite source of RTP traffic from PUBLIC Kamailio IP to LAN Kamailio IP when user A calls B via Asterisk?

---

SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

---

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

---

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

---

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

---

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

Hi again,

Still Missing 200OK for this call. It'll be helpful to send a complete trace for the call coming in to the Asterisk at first place and then Dialing out to the B-leg whose trace which you've just shared.

On Tue, Aug 6, 2013 at 3:22 AM, Alexandr Usov blessendor@gmail.com wrote:

<------------> Dial (.......) in new stack

== Using SIP RTP CoS mark 5 Audio is at 19614 Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100004 (alaw) to SDP Adding codec 100017 (testlaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP

Reliably Transmitting (no NAT) to 2.2.2.2:5060: INVITE sip:101@sip1.domain.com.ua SIP/2.0 Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299 Max-Forwards: 70 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as1b8070ba To: sip:101@sip1.domain.com.ua Contact: sip:101@2.2.2.101:5080 Call-ID: 2ac37537499c919f01683582349522d6@sip1.domain.com.ua CSeq: 102 INVITE User-Agent: Asterisk Date: Tue, 06 Aug 2013 10:18:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 319

v=0 o=root 1885227245 1885227245 IN IP4 2.2.2.101 s=Asterisk c=IN IP4 2.2.2.101 t=0 0 m=audio 19614 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

---

-- Called SIP/101@sip1.domain.com.ua

<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5- Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes From: "101" <sip:101@sip1.domain.com.ua

...

;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as1bd39f9d Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9 CSeq: 10050 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@2.2.2.101:5080 Content-Length: 0

<------------>

<--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299;rport=5080 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as1b8070ba To: sip:101@sip1.domain.com.ua Call-ID: 2ac37537499c919f01683582349522d6@sip1.domain.com.ua CSeq: 102 INVITE Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0

<-------------> --- (8 headers 0 lines) ---

<--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2.2.2.101:5080;rport=5080;branch=z9hG4bK3c47a299 Record-Route: sip:1.1.1.1;lr;r2=on;nat=yes Record-Route: sip:2.2.2.2;lr;r2=on;nat=yes Call-ID: 2ac37537499c919f01683582349522d6@sip1.domain.com.ua From: "Bomber" sip:101@sip1.domain.com.ua;tag=as1b8070ba To: sip:101@sip1.domain.com.ua;tag=JPsPQ1pCVKkH8Y1yAjFWmZk5c5PfyLNV CSeq: 102 INVITE Contact: "101" sip:101@CLIENT.GW.PUB.IP:17303;ob Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0

<-------------> --- (11 headers 0 lines) --- list_route: hop: sip:2.2.2.2;lr;r2=on;nat=yes list_route: hop: sip:1.1.1.1;lr;r2=on;nat=yes -- SIP/sip1.domain.com.ua-0000050f is ringing

<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5- Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes From: "101" <sip:101@sip1.domain.com.ua

...

;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as1bd39f9d Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9 CSeq: 10050 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@2.2.2.101:5080 Content-Length: 0

<------------>

---

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

Note: Asrterisk LAN IP real 192.168.144.101 but must be 2.2.2.101 in this described network (I am missed to change before copy-pasting here).

2013/8/6 Alexandr Usov blessendor@gmail.com

Sorry, It was call wothout answering.

I'm disable rtp debug and got full sip trace on asterisk side.

<--- SIP read from UDP:2.2.2.2:5060 ---> INVITE sip:101@sip1.domain.com.ua SIP/2.0 Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5 Max-Forwards: 16 From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua Contact: "101" sip:101@CLIENT.GW.PUB.IP:17303;ob Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12894 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: Telephone 1.0.4 Content-Type: application/sdp Content-Length: 461

v=0 o=- 3584774018 3584774018 IN IP4 1.1.1.1 s=pjmedia c=IN IP4 1.1.1.1 t=0 0 a=X-nat:0 m=audio 45032 RTP/AVP 103 102 104 109 3 0 8 9 101 a=rtcp:45033 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:109 iLBC/8000 a=fmtp:109 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes <-------------> --- (18 headers 21 lines) --- Sending to 2.2.2.2:5060 (no NAT) Sending to 2.2.2.2:5060 (no NAT) Using INVITE request as basis request - NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii Found peer '101' for '101' from 2.2.2.2:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 103 Found RTP audio format 102 Found RTP audio format 104 Found RTP audio format 109 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 101 Found audio description format speex for ID 103 Found audio description format speex for ID 102 Found audio description format speex for ID 104 Found audio description format iLBC for ID 109 Found audio description format GSM for ID 3 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G722 for ID 9 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw), peer - audio=(gsm|ulaw|alaw|speex|speex16|ilbc|g722|speex32)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 1.1.1.1:45032 Looking for 101 in 1-internal (domain sip1.domain.com.ua) list_route: hop: sip:2.2.2.2;r2=on;lr=on;nat=yes list_route: hop: sip:1.1.1.1;r2=on;lr=on;nat=yes

<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5 Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12894 INVITE Server: Asterisk Cloud PBX 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@192.168.144.101:5080 Content-Length: 0

<------------> -- Executing [101@1-internal:1] Macro("SIP/101-00000510", "1-internal,101,60,rTt,0637679232,30,vm,Broker,101") in new stack -- Executing [s@macro-1-internal:1] NoOp("SIP/101-00000510", "") in new stack -- Executing [s@macro-1-internal:2] Dial("SIP/101-00000510", "SIP/ 101@sip1.domain.com.ua,60,rTt") in new stack == Using SIP RTP CoS mark 5 Audio is at 14084 Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100004 (alaw) to SDP Adding codec 100017 (testlaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 2.2.2.2:5060: INVITE sip:101@sip1.domain.com.ua SIP/2.0 Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK055f6241 Max-Forwards: 70 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e To: sip:101@sip1.domain.com.ua Contact: sip:101@192.168.144.101:5080 Call-ID: 025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua CSeq: 102 INVITE User-Agent: Asterisk Cloud PBX 1.0 Date: Tue, 06 Aug 2013 10:33:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 319

v=0 o=root 2136064201 2136064201 IN IP4 192.168.144.101 s=Asterisk Cloud PBX 1.0 c=IN IP4 192.168.144.101 t=0 0 m=audio 14084 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

---

-- Called SIP/101@sip1.domain.com.ua

<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5 Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12894 INVITE Server: Asterisk Cloud PBX 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@192.168.144.101:5080 Content-Length: 0

<------------>

<--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK055f6241;rport=5080 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e To: sip:101@sip1.domain.com.ua Call-ID: 025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua CSeq: 102 INVITE Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0

<-------------> --- (8 headers 0 lines) ---

<--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241 Record-Route: sip:1.1.1.1;lr;r2=on;nat=yes Record-Route: sip:2.2.2.2;lr;r2=on;nat=yes Call-ID: 025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua From: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e To: sip:101@sip1.domain.com.ua;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO CSeq: 102 INVITE Contact: "101" sip:101@CLIENT.GW.PUB.IP:17303;ob Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0

<-------------> --- (11 headers 0 lines) --- list_route: hop: sip:2.2.2.2;lr;r2=on;nat=yes list_route: hop: sip:1.1.1.1;lr;r2=on;nat=yes -- SIP/sip1.domain.com.ua-00000511 is ringing

<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5 Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12894 INVITE Server: Asterisk Cloud PBX 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@192.168.144.101:5080 Content-Length: 0

<------------>

<--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241 Record-Route: sip:1.1.1.1;lr;r2=on;nat=yes Record-Route: sip:2.2.2.2;lr;r2=on;nat=yes Call-ID: 025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua From: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e To: sip:101@sip1.domain.com.ua;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO CSeq: 102 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: "101" sip:101@CLIENT.GW.PUB.IP:17303;ob Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 253

v=0 o=- 3584774018 3584774019 IN IP4 1.1.1.1 s=pjmedia c=IN IP4 1.1.1.1 t=0 0 a=X-nat:0 m=audio 57312 RTP/AVP 0 101 a=rtcp:57313 a=rtpmap:0 PCMU/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes <-------------> --- (13 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 1.1.1.1:57312 list_route: hop: sip:2.2.2.2;lr;r2=on;nat=yes list_route: hop: sip:1.1.1.1;lr;r2=on;nat=yes set_destination: Parsing sip:2.2.2.2;lr;r2=on;nat=yes for address/port to send to set_destination: set destination to 2.2.2.2:5060 Transmitting (no NAT) to 2.2.2.2:5060: ACK sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK67e4a7a4 Route: sip:2.2.2.2;lr;r2=on;nat=yes,sip:1.1.1.1;lr;r2=on;nat=yes Max-Forwards: 70 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e To: sip:101@sip1.domain.com.ua;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO Contact: sip:101@192.168.144.101:5080 Call-ID: 025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua CSeq: 102 ACK User-Agent: Asterisk Cloud PBX 1.0 Content-Length: 0

---

-- SIP/sip1.domain.com.ua-00000511 answered SIP/101-00000510

Audio is at 18570 Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5 Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12894 INVITE Server: Asterisk Cloud PBX 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@192.168.144.101:5080 Content-Type: application/sdp Require: timer Content-Length: 294

v=0 o=root 794877266 794877266 IN IP4 192.168.144.101 s=Asterisk Cloud PBX 1.0 c=IN IP4 192.168.144.101 t=0 0 m=audio 18570 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

<------------> > 0x7f2b60530ea0 -- Probation passed - setting RTP source address to 1.1.1.1:57312

<--- SIP read from UDP:2.2.2.2:5060 ---> ACK sip:101@192.168.144.101:5080 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjogmI976-nkPYObMh8FDEf-ji4fnFUiCU Max-Forwards: 16 From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12894 ACK Content-Length: 0

<-------------> --- (9 headers 0 lines) ---

<--- SIP read from UDP:2.2.2.2:5060 ---> INVITE sip:101@192.168.144.101:5080 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4 Max-Forwards: 16 From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 Contact: "101" sip:101@CLIENT.GW.PUB.IP:17303;ob Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12895 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 253

v=0 o=- 3584774018 3584774019 IN IP4 1.1.1.1 s=pjmedia c=IN IP4 1.1.1.1 t=0 0 a=X-nat:0 m=audio 45032 RTP/AVP 0 101 a=rtcp:45033 a=rtpmap:0 PCMU/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes <-------------> --- (15 headers 13 lines) --- Sending to 2.2.2.2:5060 (no NAT) Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 1.1.1.1:45032

<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4 From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12895 INVITE Server: Asterisk Cloud PBX 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@192.168.144.101:5080 Content-Length: 0

<------------> Audio is at 18570 Adding codec 100003 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4 From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12895 INVITE Server: Asterisk Cloud PBX 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@192.168.144.101:5080 Content-Type: application/sdp Require: timer Content-Length: 270

v=0 o=root 794877266 794877267 IN IP4 192.168.144.101 s=Asterisk Cloud PBX 1.0 c=IN IP4 192.168.144.101 t=0 0 m=audio 18570 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

<------------> > 0x7f2b60141b30 -- Probation passed - setting RTP source address to 1.1.1.1:45032

<--- SIP read from UDP:2.2.2.2:5060 ---> ACK sip:101@192.168.144.101:5080 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjyTfyg.XujqDRFK4QXdHXiQAydv.OoY6i Max-Forwards: 16 From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12895 ACK Content-Length: 0

<-------------> --- (9 headers 0 lines) ---

<--- SIP read from UDP:2.2.2.2:5060 ---> BYE sip:101@192.168.144.101:5080 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKc32d.777a2532.0 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjHZ9R98C4pO9lP3OOvaTjVAe8rP7oPXnq Max-Forwards: 16 From: sip:101@sip1.domain.com.ua;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO To: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e Call-ID: 025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua CSeq: 18603 BYE User-Agent: Telephone 1.0.4 Content-Length: 0

<-------------> --- (10 headers 0 lines) --- Sending to 2.2.2.2:5060 (no NAT) Scheduling destruction of SIP dialog ' 025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKc32d.777a2532.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjHZ9R98C4pO9lP3OOvaTjVAe8rP7oPXnq From: sip:101@sip1.domain.com.ua;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO To: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e Call-ID: 025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua CSeq: 18603 BYE Server: Asterisk Cloud PBX 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0

<------------> == Spawn extension (macro-1-internal, s, 2) exited non-zero on 'SIP/101-00000510' in macro '1-internal' == Spawn extension (1-internal, 101, 1) exited non-zero on 'SIP/101-00000510' Scheduling destruction of SIP dialog 'NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii' in 6400 ms (Method: ACK) set_destination: Parsing sip:2.2.2.2;r2=on;lr=on;nat=yes for address/port to send to set_destination: set destination to 2.2.2.2:5060 Reliably Transmitting (no NAT) to 2.2.2.2:5060: BYE sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK5c0a5754 Route: sip:2.2.2.2;r2=on;lr=on;nat=yes,sip:1.1.1.1;r2=on;lr=on;nat=yes Max-Forwards: 70 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 To: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 102 BYE User-Agent: Asterisk Cloud PBX 1.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0

---

Retransmitting #1 (no NAT) to 2.2.2.2:5060: BYE sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK5c0a5754 Route: sip:2.2.2.2;r2=on;lr=on;nat=yes,sip:1.1.1.1;r2=on;lr=on;nat=yes Max-Forwards: 70 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 To: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 102 BYE User-Agent: Asterisk Cloud PBX 1.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0

---

<--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK5c0a5754 Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii From: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 To: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

CSeq: 102 BYE Content-Length: 0

<-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii' Method: ACK

2013/8/6 SamyGo govoiper@gmail.com

...

Hi again,

Still Missing 200OK for this call. It'll be helpful to send a complete trace for the call coming in to the Asterisk at first place and then Dialing out to the B-leg whose trace which you've just shared.

On Tue, Aug 6, 2013 at 3:22 AM, Alexandr Usov blessendor@gmail.comwrote:

...

<------------> Dial (.......) in new stack

== Using SIP RTP CoS mark 5 Audio is at 19614 Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100004 (alaw) to SDP Adding codec 100017 (testlaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP

Reliably Transmitting (no NAT) to 2.2.2.2:5060: INVITE sip:101@sip1.domain.com.ua SIP/2.0 Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299 Max-Forwards: 70 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as1b8070ba To: sip:101@sip1.domain.com.ua Contact: sip:101@2.2.2.101:5080 Call-ID: 2ac37537499c919f01683582349522d6@sip1.domain.com.ua CSeq: 102 INVITE User-Agent: Asterisk Date: Tue, 06 Aug 2013 10:18:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 319

v=0 o=root 1885227245 1885227245 IN IP4 2.2.2.101 s=Asterisk c=IN IP4 2.2.2.101 t=0 0 m=audio 19614 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

---

-- Called SIP/101@sip1.domain.com.ua

<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5- Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes From: "101" <sip:101@sip1.domain.com.ua

...

;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as1bd39f9d Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9 CSeq: 10050 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@2.2.2.101:5080 Content-Length: 0

<------------>

<--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299;rport=5080 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as1b8070ba To: sip:101@sip1.domain.com.ua Call-ID: 2ac37537499c919f01683582349522d6@sip1.domain.com.ua CSeq: 102 INVITE Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0

<-------------> --- (8 headers 0 lines) ---

<--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2.2.2.101:5080;rport=5080;branch=z9hG4bK3c47a299 Record-Route: sip:1.1.1.1;lr;r2=on;nat=yes Record-Route: sip:2.2.2.2;lr;r2=on;nat=yes Call-ID: 2ac37537499c919f01683582349522d6@sip1.domain.com.ua From: "Bomber" sip:101@sip1.domain.com.ua;tag=as1b8070ba To: sip:101@sip1.domain.com.ua;tag=JPsPQ1pCVKkH8Y1yAjFWmZk5c5PfyLNV CSeq: 102 INVITE Contact: "101" sip:101@CLIENT.GW.PUB.IP:17303;ob Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0

<-------------> --- (11 headers 0 lines) --- list_route: hop: sip:2.2.2.2;lr;r2=on;nat=yes list_route: hop: sip:1.1.1.1;lr;r2=on;nat=yes -- SIP/sip1.domain.com.ua-0000050f is ringing

<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5- Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes From: "101" <sip:101@sip1.domain.com.ua

...

;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as1bd39f9d Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9 CSeq: 10050 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@2.2.2.101:5080 Content-Length: 0

<------------>

---

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

---

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

As I posted resently:

Note: Asrterisk LAN IP real 192.168.144.101 but must be 2.2.2.101 in this described network (I am missed to change before copy-pasting here).

2013/8/6 SamyGo govoiper@gmail.com

I wonder who this belongs to : c=IN IP4 192.168.144.101

Also your Kamailio just sends the c=IN IP4 1.1.1.1 for the very first incoming call that tells me that RTP proxy function is either running on 1.1.1.1 only or if its in bridging mode then you're not using the right flag combination to use the 2.2.2.2 IP.

Is your kamailio set to have "mhomed=yes"; just wanted to know.

---

Sammy

On Tue, Aug 6, 2013 at 3:36 AM, Alexandr Usov blessendor@gmail.comwrote:

...

Sorry, It was call wothout answering.

I'm disable rtp debug and got full sip trace on asterisk side.

<--- SIP read from UDP:2.2.2.2:5060 ---> INVITE sip:101@sip1.domain.com.ua SIP/2.0

Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5 Max-Forwards: 16 From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua

Contact: "101" sip:101@CLIENT.GW.PUB.IP:17303;ob Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12894 INVITE

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: Telephone 1.0.4 Content-Type: application/sdp Content-Length: 461

v=0 o=- 3584774018 3584774018 IN IP4 1.1.1.1 s=pjmedia c=IN IP4 1.1.1.1 t=0 0 a=X-nat:0 m=audio 45032 RTP/AVP 103 102 104 109 3 0 8 9 101 a=rtcp:45033 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:109 iLBC/8000 a=fmtp:109 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes <-------------> --- (18 headers 21 lines) --- Sending to 2.2.2.2:5060 (no NAT) Sending to 2.2.2.2:5060 (no NAT) Using INVITE request as basis request - NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii Found peer '101' for '101' from 2.2.2.2:5060

== Using SIP RTP CoS mark 5 Found RTP audio format 103 Found RTP audio format 102 Found RTP audio format 104 Found RTP audio format 109 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 101 Found audio description format speex for ID 103 Found audio description format speex for ID 102 Found audio description format speex for ID 104 Found audio description format iLBC for ID 109 Found audio description format GSM for ID 3 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G722 for ID 9 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw), peer - audio=(gsm|ulaw|alaw|speex|speex16|ilbc|g722|speex32)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 1.1.1.1:45032 Looking for 101 in 1-internal (domain sip1.domain.com.ua) list_route: hop: sip:2.2.2.2;r2=on;lr=on;nat=yes list_route: hop: sip:1.1.1.1;r2=on;lr=on;nat=yes

<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5

Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12894 INVITE Server: Asterisk Cloud PBX 1.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@192.168.144.101:5080 Content-Length: 0

<------------> -- Executing [101@1-internal:1] Macro("SIP/101-00000510", "1-internal,101,60,rTt,0637679232,30,vm,Broker,101") in new stack -- Executing [s@macro-1-internal:1] NoOp("SIP/101-00000510", "") in new stack -- Executing [s@macro-1-internal:2] Dial("SIP/101-00000510", "SIP/ 101@sip1.domain.com.ua,60,rTt") in new stack

== Using SIP RTP CoS mark 5 Audio is at 14084

Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100004 (alaw) to SDP Adding codec 100017 (testlaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 2.2.2.2:5060: INVITE sip:101@sip1.domain.com.ua SIP/2.0 Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK055f6241 Max-Forwards: 70 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e To: sip:101@sip1.domain.com.ua Contact: sip:101@192.168.144.101:5080 Call-ID: 025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua CSeq: 102 INVITE User-Agent: Asterisk Cloud PBX 1.0 Date: Tue, 06 Aug 2013 10:33:43 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 319

v=0 o=root 2136064201 2136064201 IN IP4 192.168.144.101 s=Asterisk Cloud PBX 1.0 c=IN IP4 192.168.144.101 t=0 0 m=audio 14084 RTP/AVP 0 3 8 101

a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

---

-- Called SIP/101@sip1.domain.com.ua

<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5

Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12894 INVITE Server: Asterisk Cloud PBX 1.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@192.168.144.101:5080

Content-Length: 0

<------------>

<--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK055f6241;rport=5080 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e To: sip:101@sip1.domain.com.ua Call-ID: 025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua

CSeq: 102 INVITE Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0

<-------------> --- (8 headers 0 lines) ---

<--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241

Record-Route: sip:1.1.1.1;lr;r2=on;nat=yes Record-Route: sip:2.2.2.2;lr;r2=on;nat=yes Call-ID: 025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua From: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e To: sip:101@sip1.domain.com.ua;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO

CSeq: 102 INVITE Contact: "101" sip:101@CLIENT.GW.PUB.IP:17303;ob Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0

<-------------> --- (11 headers 0 lines) --- list_route: hop: sip:2.2.2.2;lr;r2=on;nat=yes list_route: hop: sip:1.1.1.1;lr;r2=on;nat=yes -- SIP/sip1.domain.com.ua-00000511 is ringing

<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5

Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12894 INVITE Server: Asterisk Cloud PBX 1.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@192.168.144.101:5080

Content-Length: 0

<------------>

<--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241

Record-Route: sip:1.1.1.1;lr;r2=on;nat=yes Record-Route: sip:2.2.2.2;lr;r2=on;nat=yes Call-ID: 025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua From: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e To: sip:101@sip1.domain.com.ua;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO CSeq: 102 INVITE

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: "101" sip:101@CLIENT.GW.PUB.IP:17303;ob Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 253

v=0 o=- 3584774018 3584774019 IN IP4 1.1.1.1 s=pjmedia c=IN IP4 1.1.1.1 t=0 0 a=X-nat:0 m=audio 57312 RTP/AVP 0 101 a=rtcp:57313 a=rtpmap:0 PCMU/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes <-------------> --- (13 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 1.1.1.1:57312

list_route: hop: sip:2.2.2.2;lr;r2=on;nat=yes list_route: hop: sip:1.1.1.1;lr;r2=on;nat=yes set_destination: Parsing sip:2.2.2.2;lr;r2=on;nat=yes for address/port to send to

set_destination: set destination to 2.2.2.2:5060 Transmitting (no NAT) to 2.2.2.2:5060: ACK sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK67e4a7a4 Route: sip:2.2.2.2;lr;r2=on;nat=yes,sip:1.1.1.1;lr;r2=on;nat=yes Max-Forwards: 70 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e To: sip:101@sip1.domain.com.ua;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO Contact: sip:101@192.168.144.101:5080 Call-ID: 025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua CSeq: 102 ACK User-Agent: Asterisk Cloud PBX 1.0 Content-Length: 0

---

-- SIP/sip1.domain.com.ua-00000511 answered SIP/101-00000510

Audio is at 18570

Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5

Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12894 INVITE Server: Asterisk Cloud PBX 1.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@192.168.144.101:5080 Content-Type: application/sdp Require: timer Content-Length: 294

v=0 o=root 794877266 794877266 IN IP4 192.168.144.101 s=Asterisk Cloud PBX 1.0 c=IN IP4 192.168.144.101 t=0 0 m=audio 18570 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

<------------> > 0x7f2b60530ea0 -- Probation passed - setting RTP source address to 1.1.1.1:57312

<--- SIP read from UDP:2.2.2.2:5060 ---> ACK sip:101@192.168.144.101:5080 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjogmI976-nkPYObMh8FDEf-ji4fnFUiCU Max-Forwards: 16 From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12894 ACK Content-Length: 0

<-------------> --- (9 headers 0 lines) ---

<--- SIP read from UDP:2.2.2.2:5060 ---> INVITE sip:101@192.168.144.101:5080 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4 Max-Forwards: 16 From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42

Contact: "101" sip:101@CLIENT.GW.PUB.IP:17303;ob Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12895 INVITE

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 253

v=0 o=- 3584774018 3584774019 IN IP4 1.1.1.1 s=pjmedia c=IN IP4 1.1.1.1 t=0 0 a=X-nat:0 m=audio 45032 RTP/AVP 0 101 a=rtcp:45033 a=rtpmap:0 PCMU/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes <-------------> --- (15 headers 13 lines) --- Sending to 2.2.2.2:5060 (no NAT) Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 1.1.1.1:45032

<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4 From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12895 INVITE Server: Asterisk Cloud PBX 1.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@192.168.144.101:5080 Content-Length: 0

<------------> Audio is at 18570

Adding codec 100003 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4 From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12895 INVITE Server: Asterisk Cloud PBX 1.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@192.168.144.101:5080 Content-Type: application/sdp Require: timer Content-Length: 270

v=0 o=root 794877266 794877267 IN IP4 192.168.144.101 s=Asterisk Cloud PBX 1.0 c=IN IP4 192.168.144.101 t=0 0 m=audio 18570 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

<------------> > 0x7f2b60141b30 -- Probation passed - setting RTP source address to 1.1.1.1:45032

<--- SIP read from UDP:2.2.2.2:5060 ---> ACK sip:101@192.168.144.101:5080 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjyTfyg.XujqDRFK4QXdHXiQAydv.OoY6i Max-Forwards: 16 From: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 12895 ACK Content-Length: 0

<-------------> --- (9 headers 0 lines) ---

<--- SIP read from UDP:2.2.2.2:5060 ---> BYE sip:101@192.168.144.101:5080 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKc32d.777a2532.0 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjHZ9R98C4pO9lP3OOvaTjVAe8rP7oPXnq Max-Forwards: 16 From: sip:101@sip1.domain.com.ua;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO To: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e Call-ID: 025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua CSeq: 18603 BYE User-Agent: Telephone 1.0.4 Content-Length: 0

<-------------> --- (10 headers 0 lines) --- Sending to 2.2.2.2:5060 (no NAT) Scheduling destruction of SIP dialog ' 025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKc32d.777a2532.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjHZ9R98C4pO9lP3OOvaTjVAe8rP7oPXnq From: sip:101@sip1.domain.com.ua;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO To: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e Call-ID: 025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua CSeq: 18603 BYE Server: Asterisk Cloud PBX 1.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0

<------------> == Spawn extension (macro-1-internal, s, 2) exited non-zero on 'SIP/101-00000510' in macro '1-internal' == Spawn extension (1-internal, 101, 1) exited non-zero on 'SIP/101-00000510' Scheduling destruction of SIP dialog 'NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii' in 6400 ms (Method: ACK) set_destination: Parsing sip:2.2.2.2;r2=on;lr=on;nat=yes for address/port to send to

set_destination: set destination to 2.2.2.2:5060 Reliably Transmitting (no NAT) to 2.2.2.2:5060: BYE sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK5c0a5754 Route: sip:2.2.2.2;r2=on;lr=on;nat=yes,sip:1.1.1.1;r2=on;lr=on;nat=yes Max-Forwards: 70 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 To: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 102 BYE User-Agent: Asterisk Cloud PBX 1.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0

---

Retransmitting #1 (no NAT) to 2.2.2.2:5060: BYE sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK5c0a5754 Route: sip:2.2.2.2;r2=on;lr=on;nat=yes,sip:1.1.1.1;r2=on;lr=on;nat=yes Max-Forwards: 70 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 To: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii CSeq: 102 BYE User-Agent: Asterisk Cloud PBX 1.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0

---

<--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK5c0a5754 Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii From: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 To: "101" <sip:101@sip1.domain.com.ua

...

;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC

CSeq: 102 BYE Content-Length: 0

<-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii' Method: ACK

2013/8/6 SamyGo govoiper@gmail.com

...

Hi again,

Still Missing 200OK for this call. It'll be helpful to send a complete trace for the call coming in to the Asterisk at first place and then Dialing out to the B-leg whose trace which you've just shared.

On Tue, Aug 6, 2013 at 3:22 AM, Alexandr Usov blessendor@gmail.comwrote:

...

<------------> Dial (.......) in new stack

== Using SIP RTP CoS mark 5 Audio is at 19614 Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100004 (alaw) to SDP Adding codec 100017 (testlaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP

Reliably Transmitting (no NAT) to 2.2.2.2:5060: INVITE sip:101@sip1.domain.com.ua SIP/2.0 Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299 Max-Forwards: 70 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as1b8070ba To: sip:101@sip1.domain.com.ua Contact: sip:101@2.2.2.101:5080 Call-ID: 2ac37537499c919f01683582349522d6@sip1.domain.com.ua CSeq: 102 INVITE User-Agent: Asterisk Date: Tue, 06 Aug 2013 10:18:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 319

v=0 o=root 1885227245 1885227245 IN IP4 2.2.2.101 s=Asterisk c=IN IP4 2.2.2.101 t=0 0 m=audio 19614 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

---

-- Called SIP/101@sip1.domain.com.ua

<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5- Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes From: "101" <sip:101@sip1.domain.com.ua

...

;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as1bd39f9d Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9 CSeq: 10050 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@2.2.2.101:5080 Content-Length: 0

<------------>

<--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299;rport=5080 From: "Bomber" sip:101@sip1.domain.com.ua;tag=as1b8070ba To: sip:101@sip1.domain.com.ua Call-ID: 2ac37537499c919f01683582349522d6@sip1.domain.com.ua CSeq: 102 INVITE Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0

<-------------> --- (8 headers 0 lines) ---

<--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2.2.2.101:5080;rport=5080;branch=z9hG4bK3c47a299 Record-Route: sip:1.1.1.1;lr;r2=on;nat=yes Record-Route: sip:2.2.2.2;lr;r2=on;nat=yes Call-ID: 2ac37537499c919f01683582349522d6@sip1.domain.com.ua From: "Bomber" sip:101@sip1.domain.com.ua;tag=as1b8070ba To: sip:101@sip1.domain.com.ua;tag=JPsPQ1pCVKkH8Y1yAjFWmZk5c5PfyLNV CSeq: 102 INVITE Contact: "101" sip:101@CLIENT.GW.PUB.IP:17303;ob Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0

<-------------> --- (11 headers 0 lines) --- list_route: hop: sip:2.2.2.2;lr;r2=on;nat=yes list_route: hop: sip:1.1.1.1;lr;r2=on;nat=yes -- SIP/sip1.domain.com.ua-0000050f is ringing

<--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2 Via: SIP/2.0/UDP 192.168.10.240:52396 ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5- Record-Route: sip:2.2.2.2;r2=on;lr=on;nat=yes Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes From: "101" <sip:101@sip1.domain.com.ua

...

;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD

To: "Bomber" sip:101@sip1.domain.com.ua;tag=as1bd39f9d Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9 CSeq: 10050 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:101@2.2.2.101:5080 Content-Length: 0

<------------>

---

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

---

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

---

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

---

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

I am gound flags for rtpproxy_manage(which help external (behind NAT) UA registerd on Kamailio call to Echo() test extension on Asterisk - it is "cwie". But for Peer-to-Peerm, registered on Kamailio and working though Asterisk dialplan, it must be rtpproxy_manage("cwii").

Thanks to author of this example of kamailio.cfg https://github.com/xlab1/sipfe_kamailio/blob/master/kamailio.cfg

And from Kamailio documentation for rtpproxy_offer([flags [, ip_address]]):

-

*c* - flags to change the session-level SDP connection (c=) IP if media-description also includes connection information. -

*w* - flags that for the UA from which message is received, support symmetric RTP must be forced.

Hope I am understand what I do =) Thank you all for speaking with me! I'll be back... ASAP

Finally it's working:

#!define ASTERISK_LAN1 2.2.2.0/24 # End second LAN for PBXs - tunneling with LAN1 over OpenVPN: #!define ASTERISK_LAN2 3.3.3.0.0/24

rtpproxy in bridge mode.

And route[NATMANAGE] from Asipto tutorial ( http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb) changed for my needs :

# RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return;

if((src_ip==ASTERISK_LAN1) || (src_ip==ASTERISK_LAN2)) { rtpproxy_manage("cwei"); } else { rtpproxy_manage("cwie"); }

if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; }

Tested and worked on both tested Asterisk (LAN1 and LAN2). Kamailio just one - in the LAN1.

Please check the rtpproxy function and paste the way it is written in your configuration file. Share the output of "ps -ef | grep rtpproxy" and "netstat -pln|grep rtpproxy"

-- Sammy

On Tue, Aug 6, 2013 at 3:55 AM, Alexandr Usov blessendor@gmail.com wrote:

Note: Asrterisk LAN IP real 192.168.144.101 but must be 2.2.2.101 in this described network (I am missed to change before copy-pasting here).

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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

Ok thanks,

All fine except the rtpproxy_manage function is just out in the open and since you're in bridging mode you need to realize that your kamailio may receive calls from the WAN Interface or from the Asterisks on LAN interface. So if you're to bridge the RTPs at the proxy then how can Kamailio detect that which combination of flags to use in rtpproxy_manage() function?

As an example if you replace your rtpproxy_manage(); with rtpproxy_manage("ei"); and make a call from SIP user to 777 in asterisk dialplan where 777 is echo()

exten => 777,1,Answer() same => n,Echo()

Take a sip trace now and see that you get the desired LAN IP of kamailio coming in on Asterisk ? revert the "ei" to "ie" if not. This simple test should get you a two way audio established for any call from Internet to the Asterisk via Kamailio.

The Next step is to figure out that your call is coming from Asterisk and wants to go out from kamailio's WAN interface. use inverse flag combination here and let the call go out.

I hope it works for you now.

-- Sammy

On Tue, Aug 6, 2013 at 4:06 AM, Alexandr Usov blessendor@gmail.com wrote:

ps -ef | grep rtpproxy kamailio 15853 1 0 12:41 ? 00:00:01 /usr/sbin/rtpproxy -u kamailio -l 1.1.1.1 2.2.2.2 -s unix:/var/run/rtpproxy.sock

netstat -pln|grep rtpproxy unix 2 [ ACC ] STREAM LISTENING 238561268 15853/rtpproxy /var/run/rtpproxy.sock

I'm trying rtpproxy_manage("ei"); rtpproxy_manage("ie"); rtpproxy_manage(); - the same if rtpproxy in bridged mode.

Config cuts:

#!define WITH_NAT ... #!ifdef WITH_NAT # ----- rtpproxy params ----- # modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:12221") modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock")

# ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@sip.myrealhostdomain.com ")

# params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif

....

# Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error();

    exit;
}

}

...

# Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return;

}

# RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB);

        }
    }
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
    return;

    rtpproxy_manage();

if (is_request()) {
    if (!has_totag()) {
        add_rr_param(";nat=yes");
    }
}
if (is_reply()) {
    if(isbflagset(FLB_NATB)) {
        fix_nated_contact();
    }
}

#!endif return; }

2013/8/6 SamyGo govoiper@gmail.com

...

Please check the rtpproxy function and paste the way it is written in your configuration file. Share the output of "ps -ef | grep rtpproxy" and "netstat -pln|grep rtpproxy"

-- Sammy

On Tue, Aug 6, 2013 at 3:55 AM, Alexandr Usov blessendor@gmail.comwrote:

...

Note: Asrterisk LAN IP real 192.168.144.101 but must be 2.2.2.101 in this described network (I am missed to change before copy-pasting here).

---

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

---

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

---

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