I took some time to diagram out your call flow since reading a text capture is a bit difficult to visualize.  Included as both the text (which will look bad on anything that doesn’t use monospaced fonts) as well as a screen capture.

 

Given this, I’m still not clear on what change in behavior you’d expect if frames number 7 and 8 were sent in reverse order.  The new INVITE is sent to a completely different server, so the PBX shouldn’t have any awareness of this message, and only get that it’s cancelled.  If you’re looking at it in terms of trying to shave off the sliver of time difference and have the cancel sent 242 microseconds earlier – Note that it took 20 seconds from the last received message to when the cancel was sent.  250 micorseconds isn’t a drop in this bucket.

 

                snom phone         Kamailio           PBX              Voicemail

                10.0.0.117         10.0.0.177         10.0.0.188       10.0.0.26

--------------------+------------------+------------------+----------------+-----

1  15:41:39.122897 | ==== INVITE ===> |                  |                |

         + 0.001541 |                  |                  |                |

2  15:41:39.124438 | <===   100  ==== |                  |                |

         + 0.000900 |                  |                  |                |

3  15:41:39.125338 |                  | ==== INVITE ===> |                |

         + 0.001603 |                  |                  |                |

4  15:41:39.126941 |                  | <===   100  ==== |                |

         + 0.044373 |                  |                  |                |

5  15:41:39.171314 |                  | <===   183  ==== |                |

         + 0.000957 |                  |                  |                |

6  15:41:39.172272 | <===   183  ==== |                  |                |

        + 19.957119 |                  |                  |                |

7  15:41:59.129391 |                  | ============= INVITE ===========> |

         + 0.000242 |                  |                  |                |

8  15:41:59.129633 |                  | ==== CANCEL ===> |                |

         + 0.000876 |                  |                  |                |

9  15:41:59.130509 |                  | <== 200 (INV) == |                |

         + 0.000033 |                  |                  |                |

10  15:41:59.130543 |                  | <== 200 (CAN) == |                |

 

 

A white paper with black text Description automatically generated

 

 

Kaufman
Senior Voice Engineer



E: bkaufman@bcmone.com

 

SIP.US Client Support: 800.566.9810  |  SIPTRUNK Client Support: 800.250.6510  |  Flowroute Client Support: 855.356.9768

img

img

img

 

From: James Lipski via sr-users <sr-users@lists.kamailio.org>
Sent: Thursday, September 28, 2023 9:36 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,

 

To add to my previous reply, the reason for asking whether if it's possible to change the order of transactions (CANCEL first, then INVITE) is I've ran into issues where 2 200OKs are being received roughly at the same time; such as the call example from my previous post, where the receiving UA sends a 200OK for CSeq INVITE and a 200OK for CSeq CANCEL (only tested with 3CX, and Asterisk as the receiving UA); which seems to cause a conflict.

 

When testing this call scenario with another platform (an aged proxy platform that hopefully will be replaced with kamailio), when the call forks, it sends a CANCEL first to the second endpoint and the INVITE to the voicemail server.

 

Thank you.

 

 

------- Original Message -------
On Thursday, September 28th, 2023 at 6:13 PM, Ben Kaufman <bkaufman@bcmone.com> wrote:


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.

A screenshot of a computer screen Description automatically generated

 

-----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

t: <sip:10000010@10.0.0.177>

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

t: <sip:10000010@10.0.0.177>

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

t: <sip:10000010@10.0.0.177>

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

t: <sip:10000010@10.0.0.177>

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

t: <sip:10000010@10.0.0.177>

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

> 

> t: sip:10000008@10.0.0.177

> 

> 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

> 

> t: sip:10000008@10.0.0.177

> 

> 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

> 

> t: sip:10000008@10.0.0.177

> 

> 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

> 

> t: sip:10000008@10.0.0.177

> 

> 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

> 

> t: sip:10000008@10.0.0.177

> 

> 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

> > Important: keep the mailing list in the recipients, do not reply only to the sender!

> > Edit mailing list options or unsubscribe:

__________________________________________________________

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