Also, since a picture is worth a thousand words, it can be seen here that the INVITE for the new branch is sent before the CANCEL of the earlier branch. Again, I’m not sure why this would make any difference, but it does seem to be the current behavior.
-----Original Message-----
From: Ben Kaufman via sr-users <sr-users@lists.kamailio.org>
Sent: Thursday, September 28, 2023 5:05 PM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Cc: James Lipski <jameslipski@protonmail.com>; Ben Kaufman <bkaufman@bcmone.com>
Subject: [SR-Users] Re: CANCEL first, then INVITE.
CAUTION: This email originated from outside the organization. Do not click links or open attachments unless you recognize the sender and know the content is safe.
I can reproduce this behavior easily with the config below, but I'm not quite sure how/why it matters.
request_route {
xinfo("[$ru] recieved\n");
$rd = "target2";
append_branch("sip:$rU@target1", "1.0");
t_load_contacts();
t_next_contacts();
t_on_failure("MANAGE_FAILURE");
t_on_branch("MANAGE_BRANCH");
t_set_fr(5000);
t_relay();
exit;
}
branch_route[MANAGE_BRANCH] {
xinfo("Sending [$ru]\n");
}
failure_route[MANAGE_FAILURE] {
if ( t_branch_timeout() ) {
xinfo("Branch timed out.\n");
}
if ( !t_next_contacts() ) {
send_reply("503", "Server Error");
exit;
}
t_on_failure("MANAGE_FAILURE");
t_relay();
}
-----Original Message-----
From: James Lipski via sr-users <sr-users@lists.kamailio.org>
Sent: Thursday, September 28, 2023 3:08 PM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Cc: James Lipski <jameslipski@protonmail.com>
Subject: [SR-Users] Re: CANCEL first, then INVITE.
CAUTION: This email originated from outside the organization. Do not click links or open attachments unless you recognize the sender and know the content is safe.
Hello,
As with my previous update, I'm using the skeleton configurations (with minor changes, such as defining an external voicemail destination) just for testing purposes. With the skeleton configurations, I'm still seeing that the INVITE packet to the voicemail server sent first, and then the CANCEL to the second endpoint (please see SIP trace below). In this call example the second endpoint is a PBX, and is configured to route unanswered calls towards it's own local voicemail system. Is there a way to switch the behaviour so that the CANCEL packet towards endpoint 2 is sent FIRST; and then the INVITE to the voicemail system second? Thank you.
10.0.0.117 --- snom (endpoint 1)
10.0.0.177 --- kamailio
10.0.0.188 --- PBX (endpoint 2)
10.0.0.26 --- external voicemail system.
2023/09/28 15:41:39.122897 10.0.0.117:40274 -> 10.0.0.177:5060 INVITE sip:10000010@10.0.0.177 SIP/2.0
v: SIP/2.0/UDP 10.0.0.117:40274;branch=z9hG4bK-30wsteb4v1cl;rport
f: <sip:10000006@10.0.0.177>;tag=pqxa29jt8v
i: f1d61565275f-tqgd64ng8qmq
CSeq: 2 INVITE
Max-Forwards: 70
User-Agent: snomD785/10.1.159.12
m: <sip:10000006@10.0.0.117:40274;line=y3jwlgtb>;reg-id=1
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, replaces, from-change
Session-Expires: 3600
Min-SE: 90
Proxy-Authorization: Digest username="10000006",realm="10.0.0.177",nonce="ZRXYH2UV1vNTJ01Zmy97xZsd43iZmRNW",uri="sip:10000010@10.0.0.177",response="4a1d80ae375ae8d56d7b7c602e5bbcd4
",algorithm=MD5
c: application/sdp
l: 315
v=0
o=root 1457342879 1457342879 IN IP4 10.0.0.117 s=call c=IN IP4 10.0.0.117
t=0 0
m=audio 61370 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
2023/09/28 15:41:39.124438 10.0.0.177:5060 -> 10.0.0.117:40274
SIP/2.0 100 trying -- your call is important to us
v: SIP/2.0/UDP 10.0.0.117:40274;branch=z9hG4bK-30wsteb4v1cl;rport=40274;received=10.0.0.117
f: <sip:10000006@10.0.0.177>;tag=pqxa29jt8v
i: f1d61565275f-tqgd64ng8qmq
CSeq: 2 INVITE
Content-Length: 0
2023/09/28 15:41:39.125338 10.0.0.177:5060 -> 10.0.0.188:5678 INVITE sip:10000010@10.0.0.188:5678;transport=udp;line=aab32389;alias=10.0.0.188~5678~1 SIP/2.0
Record-Route: <sip:10.0.0.177;lr;nat=yes>
Via: SIP/2.0/UDP 10.0.0.177;branch=z9hG4bK87d3.36c49e4bbd41f8c5625007842e35ce3a.0
v: SIP/2.0/UDP 10.0.0.117:40274;received=10.0.0.117;branch=z9hG4bK-30wsteb4v1cl;rport=40274
f: <sip:10000006@10.0.0.177>;tag=pqxa29jt8v
i: f1d61565275f-tqgd64ng8qmq
CSeq: 2 INVITE
Max-Forwards: 69
User-Agent: snomD785/10.1.159.12
m: <sip:10000006@10.0.0.117:40274;line=y3jwlgtb;alias=10.0.0.117~40274~1>;reg-id=1
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, replaces, from-change
Session-Expires: 3600
Min-SE: 90
c: application/sdp
l: 315
v=0
o=root 1457342879 1457342879 IN IP4 10.0.0.117 s=call c=IN IP4 10.0.0.117
t=0 0
m=audio 61370 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
2023/09/28 15:41:39.126941 10.0.0.188:5678 -> 10.0.0.177:5060
SIP/2.0 100 Trying
v: SIP/2.0/UDP 10.0.0.177;branch=z9hG4bK87d3.36c49e4bbd41f8c5625007842e35ce3a.0
v: SIP/2.0/UDP 10.0.0.117:40274;received=10.0.0.117;branch=z9hG4bK-30wsteb4v1cl;rport=40274
Record-Route: <sip:10.0.0.177;lr;nat=yes>
f: <sip:10000006@10.0.0.177>;tag=pqxa29jt8v
t: <sip:10000010@10.0.0.177>;tag=c3579c1a01
i: f1d61565275f-tqgd64ng8qmq
CSeq: 2 INVITE
l: 0
2023/09/28 15:41:39.171314 10.0.0.188:5678 -> 10.0.0.177:5060
SIP/2.0 183 Session Progress
v: SIP/2.0/UDP 10.0.0.177;branch=z9hG4bK87d3.36c49e4bbd41f8c5625007842e35ce3a.0
v: SIP/2.0/UDP 10.0.0.117:40274;received=10.0.0.117;branch=z9hG4bK-30wsteb4v1cl;rport=40274
Record-Route: <sip:10.0.0.177;lr;nat=yes>
f: <sip:10000006@10.0.0.177>;tag=pqxa29jt8v
t: <sip:10000010@10.0.0.177>;tag=c3579c1a01
i: f1d61565275f-tqgd64ng8qmq
CSeq: 2 INVITE
m: <sip:10000010@10.0.0.188:5678;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
c: application/sdp
l: 247
v=0
o=- 1982229263 1982229263 IN IP4 10.0.0.188
s=-
c=IN IP4 10.0.0.188
t=0 0
m=audio 59112 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
2023/09/28 15:41:39.172272 10.0.0.177:5060 -> 10.0.0.117:40274
SIP/2.0 183 Session Progress
v: SIP/2.0/UDP 10.0.0.117:40274;received=10.0.0.117;branch=z9hG4bK-30wsteb4v1cl;rport=40274
Record-Route: <sip:10.0.0.177;lr;nat=yes>
f: <sip:10000006@10.0.0.177>;tag=pqxa29jt8v
t: <sip:10000010@10.0.0.177>;tag=c3579c1a01
i: f1d61565275f-tqgd64ng8qmq
CSeq: 2 INVITE
m: <sip:10000010@10.0.0.188:5678;transport=udp;alias=10.0.0.188~5678~1>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
c: application/sdp
l: 247
v=0
o=- 1982229263 1982229263 IN IP4 10.0.0.188
s=-
c=IN IP4 10.0.0.188
t=0 0
m=audio 59112 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
2023/09/28 15:41:59.129391 10.0.0.177:5060 -> 10.0.0.26:7066 INVITE sip:10000010@10.0.0.26:7066 SIP/2.0
Record-Route: <sip:10.0.0.177;lr;nat=yes>
Via: SIP/2.0/UDP 10.0.0.177;branch=z9hG4bK87d3.36c49e4bbd41f8c5625007842e35ce3a.1
v: SIP/2.0/UDP 10.0.0.117:40274;received=10.0.0.117;branch=z9hG4bK-30wsteb4v1cl;rport=40274
f: <sip:10000006@10.0.0.177>;tag=pqxa29jt8v
i: f1d61565275f-tqgd64ng8qmq
CSeq: 2 INVITE
Max-Forwards: 69
User-Agent: snomD785/10.1.159.12
m: <sip:10000006@10.0.0.117:40274;line=y3jwlgtb;alias=10.0.0.117~40274~1>;reg-id=1
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, replaces, from-change
Session-Expires: 3600
Min-SE: 90
c: application/sdp
l: 315
v=0
o=root 1457342879 1457342879 IN IP4 10.0.0.117 s=call c=IN IP4 10.0.0.117
t=0 0
m=audio 61370 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
2023/09/28 15:41:59.129633 10.0.0.177:5060 -> 10.0.0.188:5678 CANCEL sip:10000010@10.0.0.188:5678;transport=udp;line=aab32389;alias=10.0.0.188~5678~1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.177;branch=z9hG4bK87d3.36c49e4bbd41f8c5625007842e35ce3a.0
f: <sip:10000006@10.0.0.177>;tag=pqxa29jt8v
i: f1d61565275f-tqgd64ng8qmq
CSeq: 2 CANCEL
Max-Forwards: 69
l: 0
2023/09/28 15:41:59.130509 10.0.0.188:5678 -> 10.0.0.177:5060
SIP/2.0 200 Ok
v: SIP/2.0/UDP 10.0.0.177;branch=z9hG4bK87d3.36c49e4bbd41f8c5625007842e35ce3a.0
v: SIP/2.0/UDP 10.0.0.117:40274;received=10.0.0.117;branch=z9hG4bK-30wsteb4v1cl;rport=40274
Record-Route: <sip:10.0.0.177;lr;nat=yes>
f: <sip:10000006@10.0.0.177>;tag=pqxa29jt8v
t: <sip:10000010@10.0.0.177>;tag=c3579c1a01
i: f1d61565275f-tqgd64ng8qmq
CSeq: 2 INVITE
m: <sip:10000010@10.0.0.188:5678;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
c: application/sdp
l: 247
v=0
o=- 1982229263 1982229263 IN IP4 10.0.0.188
s=-
c=IN IP4 10.0.0.188
t=0 0
m=audio 59112 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
2023/09/28 15:41:59.130543 10.0.0.188:5678 -> 10.0.0.177:5060
SIP/2.0 200 Ok
v: SIP/2.0/UDP 10.0.0.177;branch=z9hG4bK87d3.36c49e4bbd41f8c5625007842e35ce3a.0
f: <sip:10000006@10.0.0.177>;tag=pqxa29jt8v
t: <sip:10000010@10.0.0.177>;tag=c3579c1a01
i: f1d61565275f-tqgd64ng8qmq
CSeq: 2 CANCEL
m: <sip:10000010@10.0.0.188:5678;transport=udp>
l: 0
------- Original Message -------
On Wednesday, September 27th, 2023 at 9:43 AM, James Lipski <jameslipski@protonmail.com> wrote:
> Hello,
>
> Thanks for the reply. I'm using the bundled/skeleton configurations just to test with a few modifications (adding the actual voicemail server) however even with the skeleton configurations, when the call goes unanswered, and the call forks to voicemail, the INVITE packet is sent first, followed by the CANCEL to endpoint 2. The following is a SIP trace of what I'm seeing:
>
> 10.0.0.117 -- Endpoint 1
> 10.0.0.177 -- Kamailio
> 10.0.0.110 -- Endpoint 2
> 10.0.0.26 -- Voicemail
>
> 2023/09/27 09:27:06.326167 10.0.0.117:42414 -> 10.0.0.177:5060
>
> INVITE sip:10000008@10.0.0.177 SIP/2.0
> v: SIP/2.0/UDP 10.0.0.117:42414;branch=z9hG4bK-dyp7uuyl7zjs;rport
> f: sip:10000006@10.0.0.177;tag=rsxulncekf
>
>
> i: a62d14656998-uhh0lmce49oh
> CSeq: 2 INVITE
> Max-Forwards: 70
> User-Agent: snomD785/10.1.159.12
> m: sip:10000006@10.0.0.117:42414;line=aoec88fl;reg-id=1
>
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Supported: timer, replaces, from-change
> Session-Expires: 3600
> Min-SE: 90
> Proxy-Authorization: Digest
> username="10000006",realm="10.0.0.177",nonce="ZRQu1mUULaq2rH8lPjaCtARP
> y4Wa7KfQ",uri="sip:10000008@10.0.0.177",response="a2cf69a439f25e97b238
> 510377cc4900",algorithm=MD5
> c: application/sdp
> l: 311
>
> v=0
> o=root 44632463 44632463 IN IP4 10.0.0.117 s=call c=IN IP4 10.0.0.117
> t=0 0
> m=audio 65348 RTP/AVP 9 0 8 18 101
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
>
>
> 2023/09/27 09:27:06.329198 10.0.0.177:5060 -> 10.0.0.117:42414
>
> SIP/2.0 100 trying -- your call is important to us
> v: SIP/2.0/UDP
> 10.0.0.117:42414;branch=z9hG4bK-dyp7uuyl7zjs;rport=42414;received=10.0
> .0.117
> f: sip:10000006@10.0.0.177;tag=rsxulncekf
>
>
> i: a62d14656998-uhh0lmce49oh
> CSeq: 2 INVITE
> Content-Length: 0
>
> 2023/09/27 09:27:06.329914 10.0.0.177:5060 -> 10.0.0.110:5063
>
> INVITE sip:10000008@10.0.0.110:5063;alias=10.0.0.110~5063~1 SIP/2.0
> Record-Route: sip:10.0.0.177;lr;nat=yes
>
> Via: SIP/2.0/UDP
> 10.0.0.177;branch=z9hG4bKa57d.d96bf39b755c51e53e995361b9567dc6.0
> v: SIP/2.0/UDP
> 10.0.0.117:42414;received=10.0.0.117;branch=z9hG4bK-dyp7uuyl7zjs;rport
> =42414
> f: sip:10000006@10.0.0.177;tag=rsxulncekf
>
>
> i: a62d14656998-uhh0lmce49oh
> CSeq: 2 INVITE
> Max-Forwards: 69
> User-Agent: snomD785/10.1.159.12
> m:
> sip:10000006@10.0.0.117:42414;line=aoec88fl;alias=10.0.0.117~42414~1;r
> eg-id=1
>
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Supported: timer, replaces, from-change
> Session-Expires: 3600
> Min-SE: 90
> c: application/sdp
> l: 311
>
> v=0
> o=root 44632463 44632463 IN IP4 10.0.0.117 s=call c=IN IP4 10.0.0.117
> t=0 0
> m=audio 65348 RTP/AVP 9 0 8 18 101
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
>
> 2023/09/27 09:27:06.385247 10.0.0.110:5063 -> 10.0.0.177:5060
>
> SIP/2.0 100 Trying
> i: a62d14656998-uhh0lmce49oh
> CSeq: 2 INVITE
> l: 0
> f: sip:10000006@10.0.0.177;tag=rsxulncekf
>
> Record-Route: sip:10.0.0.177;lr;nat=yes
>
> t: sip:10000008@10.0.0.177;tag=SP49be74061f6fb65ae
>
> v: SIP/2.0/UDP
> 10.0.0.177;branch=z9hG4bKa57d.d96bf39b755c51e53e995361b9567dc6.0;recei
> ved=10.0.0.177;rport=5060
> Via: SIP/2.0/UDP
> 10.0.0.117:42414;received=10.0.0.117;branch=z9hG4bK-dyp7uuyl7zjs;rport
> =42414
> Server: OBIHAI/OBi200-3.2.1.5757
>
>
> 2023/09/27 09:27:06.461036 10.0.0.110:5063 -> 10.0.0.177:5060
>
> SIP/2.0 180 Ringing
> i: a62d14656998-uhh0lmce49oh
> CSeq: 2 INVITE
> l: 0
> f: sip:10000006@10.0.0.177;tag=rsxulncekf
>
> Record-Route: sip:10.0.0.177;lr;nat=yes
>
> t: sip:10000008@10.0.0.177;tag=SP49be74061f6fb65ae
>
> v: SIP/2.0/UDP
> 10.0.0.177;branch=z9hG4bKa57d.d96bf39b755c51e53e995361b9567dc6.0;recei
> ved=10.0.0.177;rport=5060
> Via: SIP/2.0/UDP
> 10.0.0.117:42414;received=10.0.0.117;branch=z9hG4bK-dyp7uuyl7zjs;rport
> =42414
> Server: OBIHAI/OBi200-3.2.1.5757
> m: sip:10000008@10.0.0.110:5063
>
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE
>
>
> 2023/09/27 09:27:06.462644 10.0.0.177:5060 -> 10.0.0.117:42414
>
> SIP/2.0 180 Ringing
> i: a62d14656998-uhh0lmce49oh
> CSeq: 2 INVITE
> l: 0
> f: sip:10000006@10.0.0.177;tag=rsxulncekf
>
> Record-Route: sip:10.0.0.177;lr;nat=yes
>
> t: sip:10000008@10.0.0.177;tag=SP49be74061f6fb65ae
>
> Via: SIP/2.0/UDP
> 10.0.0.117:42414;received=10.0.0.117;branch=z9hG4bK-dyp7uuyl7zjs;rport
> =42414
> Server: OBIHAI/OBi200-3.2.1.5757
> m: sip:10000008@10.0.0.110:5063;alias=10.0.0.110~5063~1
>
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE
>
>
>
> 2023/09/27 09:27:51.447756 10.0.0.177:5060 -> 10.0.0.26:7066
>
> INVITE sip:10000008@10.0.0.26:7066 SIP/2.0
> Record-Route: sip:10.0.0.177;lr;nat=yes
>
> Via: SIP/2.0/UDP
> 10.0.0.177;branch=z9hG4bKa57d.d96bf39b755c51e53e995361b9567dc6.1
> v: SIP/2.0/UDP
> 10.0.0.117:42414;received=10.0.0.117;branch=z9hG4bK-dyp7uuyl7zjs;rport
> =42414
> f: sip:10000006@10.0.0.177;tag=rsxulncekf
>
>
> i: a62d14656998-uhh0lmce49oh
> CSeq: 2 INVITE
> Max-Forwards: 69
> User-Agent: snomD785/10.1.159.12
> m:
> sip:10000006@10.0.0.117:42414;line=aoec88fl;alias=10.0.0.117~42414~1;r
> eg-id=1
>
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Supported: timer, replaces, from-change
> Session-Expires: 3600
> Min-SE: 90
> c: application/sdp
> l: 311
>
> v=0
> o=root 44632463 44632463 IN IP4 10.0.0.117 s=call c=IN IP4 10.0.0.117
> t=0 0
> m=audio 65348 RTP/AVP 9 0 8 18 101
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
>
> 2023/09/27 09:27:51.466497 10.0.0.177:5060 -> 10.0.0.110:5063
>
> CANCEL sip:10000008@10.0.0.110:5063;alias=10.0.0.110~5063~1 SIP/2.0
> Via: SIP/2.0/UDP
> 10.0.0.177;branch=z9hG4bKa57d.d96bf39b755c51e53e995361b9567dc6.0
> f: sip:10000006@10.0.0.177;tag=rsxulncekf
>
>
> i: a62d14656998-uhh0lmce49oh
> CSeq: 2 CANCEL
> Max-Forwards: 69
> l: 0
>
>
> 2023/09/27 09:27:51.466503 10.0.0.177:5060 -> 10.0.0.26:7066
>
> ACK sip:10000008@10.0.0.26:7066 SIP/2.0
> Via: SIP/2.0/UDP
> 10.0.0.177;branch=z9hG4bKa57d.d96bf39b755c51e53e995361b9567dc6.1
> f: sip:10000006@10.0.0.177;tag=rsxulncekf
>
> t: sip:10000008@10.0.0.177;tag=2faa70b9-a47d-42d0-87fc-779e397ebc7e
>
> i: a62d14656998-uhh0lmce49oh
> CSeq: 2 ACK
> Max-Forwards: 69
> l: 0
>
>
> ------- Original Message -------
> On Wednesday, September 27th, 2023 at 2:04 AM, Daniel-Constantin Mierla via sr-users sr-users@lists.kamailio.org wrote:
>
>
>
> > Hello,
> >
> > default kamailio.cfg has a skeleton for doing serial forking to a
> > voicemail server. Follow the token WITH_VOICEMAIL to discover the
> > related snippets -- this can be a starting point to implement it in
> > your config.
> >
> > Cheers,
> > Daniel
> >
> > On 27.09.23 04:12, Alex Balashov via sr-users wrote:
> >
> > > Hi James,
> > >
> > > The difference you are describing is between serial and parallel forking. You clearly want the former. There are a variety of ways to achieve that, and the answer will depend on the code path taken to route to your voicemail server.
> > >
> > > -- Alex
> > >
> > > > On Sep 26, 2023, at 5:50 PM, James Lipski via sr-users sr-users@lists.kamailio.org wrote:
> > > >
> > > > Hello,
> > > >
> > > > Is there a way to change the transaction order when a failure fork occurs -- to explain, endpoint 1 calls endpoint 2. Call towards endpoint 2 goes unanswered, and so the call forks to voicemail, I see that an INVITE is sent towards my voicemail server first, followed by a CANCEL towards the endpoint; can I send the CANCEL first to the endpoint and then INVITE towards my voicemail server. I'm essentially using the bundled/sample configurations for testing.
> > > >
> > > > Thank
> > > > you.__________________________________________________________
> > > > Kamailio - Users Mailing List - Non Commercial Discussions To
> > > > unsubscribe send an email to sr-users-leave@lists.kamailio.org
> > > > Important: keep the mailing list in the recipients, do not reply only to the sender!
> > > > Edit mailing list options or unsubscribe:
> > > > --
> > > > Alex Balashov
> > > > Principal Consultant
> > > > Evariste Systems LLC
> > > > Web:
> > > > https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%
> > > > 2Fevaristesys.com%2F&data=05%7C01%7Cbkaufman%40bcmone.com%7C73b1
> > > > 320140764ada4bbf08dbc0600f7a%7Cafc1818e7b6848568913201b9396c4fc%
> > > > 7C1%7C0%7C638315291068700436%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4
> > > > wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7
> > > > C%7C%7C&sdata=opsgjhZGjuZd%2Bam%2Fijw3tgXcofOZOee4u3IkAlyEc90%3D
> > > > &reserved=0
> > > > Tel: +1-706-510-6800
> > >
> > > __________________________________________________________
> > > Kamailio - Users Mailing List - Non Commercial Discussions To
> > > unsubscribe send an email to sr-users-leave@lists.kamailio.org
> > > Important: keep the mailing list in the recipients, do not reply only to the sender!
> > > Edit mailing list options or unsubscribe:
> >
> > --
> > Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda --
> > linkedin.com/in/miconda Kamailio Consultancy - Training Services --
> > asipto.com Kamailio World Conference - kamailioworld.com
> >
> > __________________________________________________________
> > Kamailio - Users Mailing List - Non Commercial Discussions To
> > unsubscribe send an email to sr-users-leave@lists.kamailio.org
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