27 Sep
2023
27 Sep
'23
4:43 p.m.
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="ZRQu1mUULaq2rH8lPjaCtARPy4Wa7KfQ",uri="sip:10000008@10.0.0.177",response="a2cf69a439f25e97b238510377cc4900",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>;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: 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;received=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;received=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>;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: 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(a)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
--
Daniel-Constantin Mierla (@ asipto.com)
twitter.com/miconda -- linkedin.com/in/miconda
Kamailio Consultancy - Training Services -- asipto.com
Kamailio World Conference - kamailioworld.com
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Edit mailing list options or unsubscribe: On Sep 26, 2023, at 5:50 PM, James Lipski via
sr-users sr-users(a)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.__________________________________________________________
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--
Alex Balashov
Principal Consultant
Evariste Systems LLC
Web: https://evaristesys.com
Tel: +1-706-510-6800
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