[Asterisk 20.4.0]No Outgoing Communication

Artemus24 · August 20, 2023, 11:15pm

Hello Asterisk community.

My internet provider is SFR (France) and I was able to recover my telephone identifiers. I was inspired by old versions of Asterisk to configure version 20.4.0. I manage to solve several Warnings except this one which prevents me from having an outgoing telephone call. It’s a random Warning and I manage after several attempts to get the communication. I am using Zoiper 5 SoftPhone. Here is the debug:

Asterisk 20.4.0, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Running as user 'asterisk'
Running under group 'asterisk'
    -- Executing [1023@outgoing:1] Dial("PJSIP/zoiper-00000000", "PJSIP/1023@sfr") in new stack
    -- Called PJSIP/1023@sfr
[Aug 21 01:10:10] WARNING[20607]: res_pjsip_outbound_authenticator_digest.c:507 digest_create_request_with_auth: Endpoint: 'sfr': No auth objects matching realm(s) '' from challenge found.
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1023@outgoing:2] Hangup("PJSIP/zoiper-00000000", "") in new stack
  == Spawn extension (outgoing, 1023, 2) exited non-zero on 'PJSIP/zoiper-00000000'
Debian*CLI>

Artemus24 · August 20, 2023, 11:20pm

This is my file “pjsip.conf”:

[registration]
auth_rejection_permanent=yes

[transport-udp-nat]
bind=0.0.0.0
external_media_address=XXX.XXX.XXX.XXX
external_signaling_address=XXX.XXX.XXX.XXX
local_net=192.168.1.0/24
protocol=udp
type=transport

[transport-udp-ipv6]
type=transport
protocol=udp
bind=::

; --------- ;
; Templates ;
; --------- ;

[my_codecs](!)
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g722

[aor_dynamic](!)
max_contacts=9999
remove_existing=yes
type=aor

[auth_userpass](!)
auth_type=userpass
type=auth

[endpoint_internal](!,my_codecs)
context=outgoing
direct_media=no
force_rport=yes
from_domain=ims.mnc010.mcc208.3gppnetwork.org
ice_support=yes
language=fr
;;rewrite_contact=yes
rtp_symmetric=yes
transport=transport-udp-ipv6
transport=transport-udp-nat
type=endpoint

; --------- ;
; Trunk SFR ;
; --------- ;

[sfr]
contact=sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org
outbound_proxy=sip:residential.p-cscf.sfr.net\;lr
max_contacts=9999
type=aor 

[sfr_auth]
auth_type=userpass
password=XXXXXXXXXXXXXXXX
username=NDIXXXXXXXXXX.XXX.XXX@sfr.fr
type=auth 

[sfr](my_codecs)
100rel=yes
aors=sfr
context=incoming
direct_media=no
force_rport=yes
from_domain=ims.mnc010.mcc208.3gppnetwork.org
from_user=+33XXXXXXXXX
ice_support=yes
outbound_auth=sfr_auth
outbound_proxy=sip:residential.p-cscf.sfr.net\;lr
rewrite_contact=yes
rtp_symmetric=yes
transport=transport-udp-nat
type=endpoint 

[sfr]
endpoint=sfr
match=sip:residential.p-cscf.sfr.net\;lr
type=identify

[sfr]
client_uri=sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org
server_uri=sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org
contact_user=+33XXXXXXXXX
outbound_auth=sfr_auth
outbound_proxy=sip:residential.p-cscf.sfr.net\;lr
transport=transport-udp-nat
type=registration 

; ------------------- ;
; Phone Line 'Zoiper' ;
; ------------------- ;

[zoiper](aor_dynamic)

[zoiper](auth_userpass)
password=zoiper
username=zoiper

[zoiper](endpoint_internal)
auth=zoiper
aors=zoiper
callerid=zoiper

Artemus24 · August 20, 2023, 11:21pm

And this is my file “extensions.conf”:

[general]
autofallthrough=no
clearglobalvars=no
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=DAHDI/G2
TRUNKMSD=1
 
[outgoing]
exten => _X.,1,Dial(PJSIP/${EXTEN}@sfr)
exten => _X.,n,Hangup()

exten => 123,1,VoiceMailMain(${CALLERID(num)})
exten => 123,n,Hangup()

[incoming]
exten => s,1,Dial(PJSIP/zoiper)
exten => s,n,Hangup()

There are certainly too many options.

Sincerely.
Artemus24.
@+

david551 · August 20, 2023, 11:45pm

What is the contents of their 401 response?

ambiorixg12 · August 21, 2023, 1:15am

I’m not a C developer, but after reading the source code and examining the functions related to the warning message, I believe in your case it would be beneficial to post the full SIP request, including the challenge response, as David requested.

digest_create_request_with_auth: Endpoint:

digest_create_request_with_auth function. It receives information about the challenge received from a SIP response and the original SIP request. It then attempts to find appropriate authentication credentials for the given challenge and creates a new SIP request with the necessary authentication headers

github.com

asterisk/asterisk/blob/master/res/res_pjsip_outbound_authenticator_digest.c

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2013, Digium, Inc.
 *
 * Mark Michelson <mmichelson@digium.com>
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 */

/*** MODULEINFO
	<depend>pjproject</depend>

This file has been truncated. show original

tootai · August 21, 2023, 7:49am

Hi

if your box is the last model of Numericable/SFR ones, no luck, SIP is broken or blocked.

Artemus24 · August 21, 2023, 2:58pm

It’s the original but without debug.

I am not at “Numéricable”, nor at “RED BY SFR”, but at SFR in FTTH.

I have the same problem if I use Asterisk behind my Box SFR or behind the ONT SFR. I don’t have this problem if I use Zoiper 5 with my SFR credentials, without going through Asterisk. Except that I can’t use Zoiper 5 because the communication only lasts 32 seconds. The Box SFR does not block outgoing communications because from time to time, I manage to get one.

Asterisk 20.4.0, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Running as user 'asterisk'
Running under group 'asterisk'
Connected to Asterisk 20.4.0 currently running on Debian (pid = 5439)
Debian*CLI> pjsip show registrations

 <Registration/ServerURI..............................>  <Auth....................>  <Status.......>
==========================================================================================

 sfr/sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org  sfr_auth                    Registered        (exp. 3581s)

Objects found: 1

Debian*CLI> pjsip set logger on
PJSIP Logging enabled
<--- Received SIP request (700 bytes) from UDP:192.168.1.11:56849 --->
REGISTER sip:192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:56849;branch=z9hG4bK-524287-1---81d372ea273ba816;rport
Max-Forwards: 70
Contact: <sip:zoiper@192.168.1.11:56849;rinstance=f21141d2b2eb523b;transport=UDP>
To: <sip:zoiper@192.168.1.11;transport=UDP>
From: <sip:zoiper@192.168.1.11;transport=UDP>;tag=424a8c1d
Call-ID: EZWTsTExCJhGdaq7t0t2aQ..
CSeq: 1 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0


<--- Transmitting SIP response (507 bytes) to UDP:192.168.1.11:56849 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.11:56849;rport=56849;received=192.168.1.11;branch=z9hG4bK-524287-1---81d372ea273ba816
Call-ID: EZWTsTExCJhGdaq7t0t2aQ..
From: <sip:zoiper@192.168.1.11>;tag=424a8c1d
To: <sip:zoiper@192.168.1.11>;tag=z9hG4bK-524287-1---81d372ea273ba816
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1692628848/488f4e3a0b966095f8e1514dbfff39df",opaque="769f0a1c5c6b0510",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.4.0
Content-Length:  0


<--- Received SIP request (994 bytes) from UDP:192.168.1.11:56849 --->
REGISTER sip:192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:56849;branch=z9hG4bK-524287-1---15c1242574ceb5fc;rport
Max-Forwards: 70
Contact: <sip:zoiper@192.168.1.11:56849;rinstance=f21141d2b2eb523b;transport=UDP>
To: <sip:zoiper@192.168.1.11;transport=UDP>
From: <sip:zoiper@192.168.1.11;transport=UDP>;tag=424a8c1d
Call-ID: EZWTsTExCJhGdaq7t0t2aQ..
CSeq: 2 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Authorization: Digest username="zoiper",realm="asterisk",nonce="1692628848/488f4e3a0b966095f8e1514dbfff39df",uri="sip:192.168.1.11;transport=UDP",response="d935f2355409dd8337992463190b6fd3",cnonce="e2ea1e264827941546072d381867e4f2",nc=00000001,qop=auth,algorithm=MD5,opaque="769f0a1c5c6b0510"
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0


    -- Added contact 'sip:zoiper@192.168.1.11:56849;transport=UDP;rinstance=f21141d2b2eb523b' to AOR 'zoiper' with expiration of 60 seconds
  == Endpoint zoiper is now Reachable
<--- Transmitting SIP response (495 bytes) to UDP:192.168.1.11:56849 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:56849;rport=56849;received=192.168.1.11;branch=z9hG4bK-524287-1---15c1242574ceb5fc
Call-ID: EZWTsTExCJhGdaq7t0t2aQ..
From: <sip:zoiper@192.168.1.11>;tag=424a8c1d
To: <sip:zoiper@192.168.1.11>;tag=z9hG4bK-524287-1---15c1242574ceb5fc
CSeq: 2 REGISTER
Date: Mon, 21 Aug 2023 14:40:48 GMT
Contact: <sip:zoiper@192.168.1.11:56849;transport=UDP;rinstance=f21141d2b2eb523b>;expires=59
Expires: 60
Server: Asterisk PBX 20.4.0
Content-Length:  0


<--- Received SIP request (863 bytes) from UDP:192.168.1.11:56849 --->
INVITE sip:1023@192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:56849;branch=z9hG4bK-524287-1---b418726bd1974dc9;rport
Max-Forwards: 70
Contact: <sip:zoiper@192.168.1.11:56849;transport=UDP>
To: <sip:1023@192.168.1.11>
From: <sip:zoiper@192.168.1.11;transport=UDP>;tag=9061b154
Call-ID: B3az4jVr-0f74XrD55ZpnA..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 185

v=0
o=Z 0 795311 IN IP4 192.168.1.11
s=Z
c=IN IP4 192.168.1.11
t=0 0
m=audio 53190 RTP/AVP 8 101 0 3 9
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux

<--- Transmitting SIP response (503 bytes) to UDP:192.168.1.11:56849 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.11:56849;rport=56849;received=192.168.1.11;branch=z9hG4bK-524287-1---b418726bd1974dc9
Call-ID: B3az4jVr-0f74XrD55ZpnA..
From: <sip:zoiper@192.168.1.11>;tag=9061b154
To: <sip:1023@192.168.1.11>;tag=z9hG4bK-524287-1---b418726bd1974dc9
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1692628854/038015d4b174ea1edfbb38b9e84aa9ba",opaque="346a5bbb1aed1a33",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.4.0
Content-Length:  0


<--- Received SIP request (351 bytes) from UDP:192.168.1.11:56849 --->
ACK sip:1023@192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:56849;branch=z9hG4bK-524287-1---b418726bd1974dc9;rport
Max-Forwards: 70
To: <sip:1023@192.168.1.11>;tag=z9hG4bK-524287-1---b418726bd1974dc9
From: <sip:zoiper@192.168.1.11;transport=UDP>;tag=9061b154
Call-ID: B3az4jVr-0f74XrD55ZpnA..
CSeq: 1 ACK
Content-Length: 0


<--- Received SIP request (1162 bytes) from UDP:192.168.1.11:56849 --->
INVITE sip:1023@192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:56849;branch=z9hG4bK-524287-1---d4fda8cb62179089;rport
Max-Forwards: 70
Contact: <sip:zoiper@192.168.1.11:56849;transport=UDP>
To: <sip:1023@192.168.1.11>
From: <sip:zoiper@192.168.1.11;transport=UDP>;tag=9061b154
Call-ID: B3az4jVr-0f74XrD55ZpnA..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Authorization: Digest username="zoiper",realm="asterisk",nonce="1692628854/038015d4b174ea1edfbb38b9e84aa9ba",uri="sip:1023@192.168.1.11;transport=UDP",response="91986c7862e4430c6d808ff733bffc9c",cnonce="53d4056f6eb95f1558297204fe5351ed",nc=00000001,qop=auth,algorithm=MD5,opaque="346a5bbb1aed1a33"
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 185

v=0
o=Z 0 795311 IN IP4 192.168.1.11
s=Z
c=IN IP4 192.168.1.11
t=0 0
m=audio 53190 RTP/AVP 8 101 0 3 9
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux

<--- Transmitting SIP response (311 bytes) to UDP:192.168.1.11:56849 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:56849;rport=56849;received=192.168.1.11;branch=z9hG4bK-524287-1---d4fda8cb62179089
Call-ID: B3az4jVr-0f74XrD55ZpnA..
From: <sip:zoiper@192.168.1.11>;tag=9061b154
To: <sip:1023@192.168.1.11>
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Content-Length:  0


    -- Executing [1023@outgoing:1] Dial("PJSIP/zoiper-00000000", "PJSIP/1023@sfr") in new stack
    -- Called PJSIP/1023@sfr
<--- Transmitting SIP request (1302 bytes) to UDP:92.91.179.24:5062 --->
INVITE sip:1023@ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPjb6725869-c224-42d2-91d4-9406dc48e16c
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=e05b2bca-5499-442d-823a-4f9f1ac3bd11
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org>
Contact: <sip:+33XXXXXXXXX@XXX.XXX.XXX.XXX:5060>
Call-ID: 61677eef-412e-45a5-a9ff-6d04e02d5840
CSeq: 21621 INVITE
Route: <sip:residential.p-cscf.sfr.net;lr>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.4.0
Content-Type: application/sdp
Content-Length:   530

v=0
o=- 898213354 898213354 IN IP4 XXX.XXX.XXX.XXX
s=Asterisk
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 10602 RTP/AVP 8 0 3 9 101
a=ice-ufrag:1bafda2f4f558c6a73d26f59422510a9
a=ice-pwd:5f888e59318dd2e256e8316d5a6d2b41
a=candidate:Hc0a8010b 1 UDP 2130706431 192.168.1.11 10602 typ host
a=candidate:Hc0a8010b 2 UDP 2130706430 192.168.1.11 10603 typ host
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (438 bytes) from UDP:92.91.179.24:5062 --->
SIP/2.0 100 Trying
Call-ID: 61677eef-412e-45a5-a9ff-6d04e02d5840
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPjb6725869-c224-42d2-91d4-9406dc48e16c;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=e05b2bca-5499-442d-823a-4f9f1ac3bd11
CSeq: 21621 INVITE
Date: Mon, 21 Aug 2023 14:40:45 GMT
Content-Length: 0


<--- Received SIP response (491 bytes) from UDP:92.91.179.24:5062 --->
SIP/2.0 407 Proxy Authentication Required
Call-ID: 61677eef-412e-45a5-a9ff-6d04e02d5840
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPjb6725869-c224-42d2-91d4-9406dc48e16c;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=64c2d4b7-64e3776d2e155346
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=e05b2bca-5499-442d-823a-4f9f1ac3bd11
CSeq: 21621 INVITE
Date: Mon, 21 Aug 2023 14:40:45 GMT
Content-Length: 0


<--- Transmitting SIP request (503 bytes) to UDP:92.91.179.24:5062 --->
ACK sip:1023@ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPjb6725869-c224-42d2-91d4-9406dc48e16c
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=e05b2bca-5499-442d-823a-4f9f1ac3bd11
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org>;tag=64c2d4b7-64e3776d2e155346
Call-ID: 61677eef-412e-45a5-a9ff-6d04e02d5840
CSeq: 21621 ACK
Route: <sip:92.91.179.24:5062;lr>
Max-Forwards: 70
User-Agent: Asterisk PBX 20.4.0
Content-Length:  0


[Aug 21 16:40:54] WARNING[5471]: res_pjsip_outbound_authenticator_digest.c:507 digest_create_request_with_auth: Endpoint: 'sfr': No auth objects matching realm(s) '' from challenge found.
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1023@outgoing:2] Hangup("PJSIP/zoiper-00000000", "") in new stack
  == Spawn extension (outgoing, 1023, 2) exited non-zero on 'PJSIP/zoiper-00000000'
<--- Transmitting SIP response (379 bytes) to UDP:192.168.1.11:56849 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.11:56849;rport=56849;received=192.168.1.11;branch=z9hG4bK-524287-1---d4fda8cb62179089
Call-ID: B3az4jVr-0f74XrD55ZpnA..
From: <sip:zoiper@192.168.1.11>;tag=9061b154
To: <sip:1023@192.168.1.11>;tag=7a81eacb-355c-43fc-a59d-d7073329ceab
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Reason: Q.850;cause=21
Content-Length:  0


<--- Received SIP request (352 bytes) from UDP:192.168.1.11:56849 --->
ACK sip:1023@192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:56849;branch=z9hG4bK-524287-1---d4fda8cb62179089;rport
Max-Forwards: 70
To: <sip:1023@192.168.1.11>;tag=7a81eacb-355c-43fc-a59d-d7073329ceab
From: <sip:zoiper@192.168.1.11;transport=UDP>;tag=9061b154
Call-ID: B3az4jVr-0f74XrD55ZpnA..
CSeq: 2 ACK
Content-Length: 0


Debian*CLI>

tootai · August 21, 2023, 3:33pm

Did you notice ?

david551 · August 21, 2023, 3:36pm

Artemus24:

<--- Received SIP response (491 bytes) from UDP:92.91.179.24:5062 --->
SIP/2.0 407 Proxy Authentication Required
Call-ID: 61677eef-412e-45a5-a9ff-6d04e02d5840
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPjb6725869-c224-42d2-91d4-9406dc48e16c;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=64c2d4b7-64e3776d2e155346
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=e05b2bca-5499-442d-823a-4f9f1ac3bd11
CSeq: 21621 INVITE
Date: Mon, 21 Aug 2023 14:40:45 GMT
Content-Length: 0

   Header field              where       proxy ACK BYE CAN INV OPT REG
   ___________________________________________________________________
   Priority                    R          ar    -   -   -   o   -   -
   Proxy-Authenticate         407         ar    -   m   -   m   m   m

The peer does not implement SIP. If you should be getting 407, find a SIP provider. If you shouldn’t be getting 407, make sure you are sending to the correct address.

Artemus24 · August 21, 2023, 7:22pm

Here is a successful communication and I haven’t changed anything in the settings.

Asterisk 20.4.0, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Running as user 'asterisk'
Running under group 'asterisk'
Connected to Asterisk 20.4.0 currently running on Debian (pid = 6864)
Debian*CLI> pjsip set logger on
PJSIP Logging enabled
Debian*CLI> pjsip show registrations

 <Registration/ServerURI..............................>  <Auth....................>  <Status.......>
==========================================================================================

 sfr/sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org  sfr_auth                    Registered        (exp. 3460s)

Objects found: 1

<--- Received SIP request (864 bytes) from UDP:192.168.1.11:35801 --->
INVITE sip:1023@192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:35801;branch=z9hG4bK-524287-1---318cf33929ff165f;rport
Max-Forwards: 70
Contact: <sip:zoiper@192.168.1.11:35801;transport=UDP>
To: <sip:1023@192.168.1.11>
From: <sip:zoiper@192.168.1.11;transport=UDP>;tag=a614fb25
Call-ID: _AhI2sArpvwjMMeX0TO5WA..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 186

v=0
o=Z 0 1272258 IN IP4 192.168.1.11
s=Z
c=IN IP4 192.168.1.11
t=0 0
m=audio 50240 RTP/AVP 8 101 0 3 9
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux

<--- Transmitting SIP response (503 bytes) to UDP:192.168.1.11:35801 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.11:35801;rport=35801;received=192.168.1.11;branch=z9hG4bK-524287-1---318cf33929ff165f
Call-ID: _AhI2sArpvwjMMeX0TO5WA..
From: <sip:zoiper@192.168.1.11>;tag=a614fb25
To: <sip:1023@192.168.1.11>;tag=z9hG4bK-524287-1---318cf33929ff165f
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1692645394/a12dce18f09db8a90668c06c1260571e",opaque="12171fa94d7b77b4",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.4.0
Content-Length:  0


<--- Received SIP request (351 bytes) from UDP:192.168.1.11:35801 --->
ACK sip:1023@192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:35801;branch=z9hG4bK-524287-1---318cf33929ff165f;rport
Max-Forwards: 70
To: <sip:1023@192.168.1.11>;tag=z9hG4bK-524287-1---318cf33929ff165f
From: <sip:zoiper@192.168.1.11;transport=UDP>;tag=a614fb25
Call-ID: _AhI2sArpvwjMMeX0TO5WA..
CSeq: 1 ACK
Content-Length: 0


<--- Received SIP request (1163 bytes) from UDP:192.168.1.11:35801 --->
INVITE sip:1023@192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:35801;branch=z9hG4bK-524287-1---fe49247919b5e77e;rport
Max-Forwards: 70
Contact: <sip:zoiper@192.168.1.11:35801;transport=UDP>
To: <sip:1023@192.168.1.11>
From: <sip:zoiper@192.168.1.11;transport=UDP>;tag=a614fb25
Call-ID: _AhI2sArpvwjMMeX0TO5WA..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Authorization: Digest username="zoiper",realm="asterisk",nonce="1692645394/a12dce18f09db8a90668c06c1260571e",uri="sip:1023@192.168.1.11;transport=UDP",response="d12ccf135142024792c95b31d0cab8dd",cnonce="0cea0d454fd3e0bd1e2cefda7fb69396",nc=00000001,qop=auth,algorithm=MD5,opaque="12171fa94d7b77b4"
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 186

v=0
o=Z 0 1272258 IN IP4 192.168.1.11
s=Z
c=IN IP4 192.168.1.11
t=0 0
m=audio 50240 RTP/AVP 8 101 0 3 9
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux

<--- Transmitting SIP response (311 bytes) to UDP:192.168.1.11:35801 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:35801;rport=35801;received=192.168.1.11;branch=z9hG4bK-524287-1---fe49247919b5e77e
Call-ID: _AhI2sArpvwjMMeX0TO5WA..
From: <sip:zoiper@192.168.1.11>;tag=a614fb25
To: <sip:1023@192.168.1.11>
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Content-Length:  0


    -- Executing [1023@outgoing:1] Dial("PJSIP/zoiper-0000001c", "PJSIP/1023@sfr") in new stack
    -- Called PJSIP/1023@sfr
<--- Transmitting SIP request (1302 bytes) to UDP:92.91.179.72:5062 --->
INVITE sip:1023@ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj79e35367-c662-47e6-a939-4b570951337a
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=c4a34d7d-6db4-4614-b793-0ef720b97f8c
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org>
Contact: <sip:+33XXXXXXXXX@XXX.XXX.XXX.XXX:5060>
Call-ID: b5b16872-29a9-4702-a045-fa4946bd1e06
CSeq: 21102 INVITE
Route: <sip:residential.p-cscf.sfr.net;lr>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.4.0
Content-Type: application/sdp
Content-Length:   530

v=0
o=- 934554494 934554494 IN IP4 XXX.XXX.XXX.XXX
s=Asterisk
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 12324 RTP/AVP 8 0 3 9 101
a=ice-ufrag:456ef374129f9dcc7510e12467e0e9e4
a=ice-pwd:7d3949c203c528a867fe7f4914c1fe9e
a=candidate:Hc0a8010b 1 UDP 2130706431 192.168.1.11 12324 typ host
a=candidate:Hc0a8010b 2 UDP 2130706430 192.168.1.11 12325 typ host
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (438 bytes) from UDP:92.91.179.72:5062 --->
SIP/2.0 100 Trying
Call-ID: b5b16872-29a9-4702-a045-fa4946bd1e06
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPj79e35367-c662-47e6-a939-4b570951337a;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=c4a34d7d-6db4-4614-b793-0ef720b97f8c
CSeq: 21102 INVITE
Date: Mon, 21 Aug 2023 19:16:25 GMT
Content-Length: 0


<--- Received SIP response (941 bytes) from UDP:92.91.179.72:5062 --->
SIP/2.0 183 Session Progress
Call-ID: b5b16872-29a9-4702-a045-fa4946bd1e06
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPj79e35367-c662-47e6-a939-4b570951337a;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=649e0815-64e3b80961f7adc-gm-po-lucentPCSF-154063
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=c4a34d7d-6db4-4614-b793-0ef720b97f8c
CSeq: 21102 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-153711@pcgw-0006.imsgroup-001.cor1asbc05.ims.sfr.net:5062;x-afi=108>
Content-Type: application/sdp
RSeq: 1
Content-Length: 235

v=0
o=LucentPCSF 1735577902 1735577902 IN IP4 imsgroup-001.cor1asbc05.ims.sfr.net
s=-
c=IN IP4 92.91.224.133
t=0 0
m=audio 41874 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=ptime:20
a=silenceSupp:off - - - -

<--- Transmitting SIP request (524 bytes) to UDP:92.91.179.72:5062 --->
PRACK sip:lucentNGFS-153711@92.91.179.72:5062;x-afi=108 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj0c77a9d6-4dd9-40ed-b7d6-05bee375bc28
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=c4a34d7d-6db4-4614-b793-0ef720b97f8c
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org>;tag=649e0815-64e3b80961f7adc-gm-po-lucentPCSF-154063
Call-ID: b5b16872-29a9-4702-a045-fa4946bd1e06
CSeq: 21103 PRACK
RAck: 1 21102 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 20.4.0
Content-Length:  0


    -- PJSIP/sfr-0000001d is making progress passing it to PJSIP/zoiper-0000001c
<--- Transmitting SIP response (837 bytes) to UDP:192.168.1.11:35801 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.11:35801;rport=35801;received=192.168.1.11;branch=z9hG4bK-524287-1---fe49247919b5e77e
Call-ID: _AhI2sArpvwjMMeX0TO5WA..
From: <sip:zoiper@192.168.1.11>;tag=a614fb25
To: <sip:1023@192.168.1.11>;tag=cd9a15ab-17aa-461e-947d-1abda9cac4f2
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Contact: <sip:192.168.1.11:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Type: application/sdp
Content-Length:   296

v=0
o=- 0 1272260 IN IP4 192.168.1.11
s=Asterisk
c=IN IP4 192.168.1.11
t=0 0
m=audio 16642 RTP/AVP 8 0 3 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (544 bytes) from UDP:92.91.179.72:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPj0c77a9d6-4dd9-40ed-b7d6-05bee375bc28;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=649e0815-64e3b80961f7adc-gm-po-lucentPCSF-154063
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=c4a34d7d-6db4-4614-b793-0ef720b97f8c
Call-ID: b5b16872-29a9-4702-a045-fa4946bd1e06
CSeq: 21103 PRACK
Content-Length: 0
Contact: <sip:lucentNGFS-153711@pcgw-0006.imsgroup-001.cor1asbc05.ims.sfr.net:5062;x-afi=108>


<--- Received SIP response (689 bytes) from UDP:92.91.179.72:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPj79e35367-c662-47e6-a939-4b570951337a;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=649e0815-64e3b80961f7adc-gm-po-lucentPCSF-154063
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=c4a34d7d-6db4-4614-b793-0ef720b97f8c
Call-ID: b5b16872-29a9-4702-a045-fa4946bd1e06
CSeq: 21102 INVITE
Content-Length: 0
Contact: <sip:lucentNGFS-153711@pcgw-0006.imsgroup-001.cor1asbc05.ims.sfr.net:5062;x-afi=108>
Require: 100rel
RSeq: 2
P-Early-Media: sendrecv
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE


<--- Transmitting SIP request (524 bytes) to UDP:92.91.179.72:5062 --->
PRACK sip:lucentNGFS-153711@92.91.179.72:5062;x-afi=108 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPjf1065fc0-fbe0-4e52-81b5-8fa9943b858b
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=c4a34d7d-6db4-4614-b793-0ef720b97f8c
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org>;tag=649e0815-64e3b80961f7adc-gm-po-lucentPCSF-154063
Call-ID: b5b16872-29a9-4702-a045-fa4946bd1e06
CSeq: 21104 PRACK
RAck: 2 21102 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 20.4.0
Content-Length:  0


    -- PJSIP/sfr-0000001d is ringing
<--- Transmitting SIP response (837 bytes) to UDP:192.168.1.11:35801 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.11:35801;rport=35801;received=192.168.1.11;branch=z9hG4bK-524287-1---fe49247919b5e77e
Call-ID: _AhI2sArpvwjMMeX0TO5WA..
From: <sip:zoiper@192.168.1.11>;tag=a614fb25
To: <sip:1023@192.168.1.11>;tag=cd9a15ab-17aa-461e-947d-1abda9cac4f2
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.1.11:5060>
Content-Type: application/sdp
Content-Length:   296

v=0
o=- 0 1272260 IN IP4 192.168.1.11
s=Asterisk
c=IN IP4 192.168.1.11
t=0 0
m=audio 16642 RTP/AVP 8 0 3 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (544 bytes) from UDP:92.91.179.72:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPjf1065fc0-fbe0-4e52-81b5-8fa9943b858b;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=649e0815-64e3b80961f7adc-gm-po-lucentPCSF-154063
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=c4a34d7d-6db4-4614-b793-0ef720b97f8c
Call-ID: b5b16872-29a9-4702-a045-fa4946bd1e06
CSeq: 21104 PRACK
Content-Length: 0
Contact: <sip:lucentNGFS-153711@pcgw-0006.imsgroup-001.cor1asbc05.ims.sfr.net:5062;x-afi=108>


<--- Received SIP response (976 bytes) from UDP:92.91.179.72:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPj79e35367-c662-47e6-a939-4b570951337a;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=649e0815-64e3b80961f7adc-gm-po-lucentPCSF-154063
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=c4a34d7d-6db4-4614-b793-0ef720b97f8c
Call-ID: b5b16872-29a9-4702-a045-fa4946bd1e06
CSeq: 21102 INVITE
Content-Length: 0
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-153711@pcgw-0006.imsgroup-001.cor1asbc05.ims.sfr.net:5062;x-afi=108>
Supported: timer
Accept-Encoding: identity
User-Agent: Asterisk PBX 20.4.0
Session-Expires: 3600;refresher=uas
X-Genesys-GVP_IN_DATA: 202308212116;017d035166fcad8e;099991023;+33XXXXXXXXX;dRKudjKoHjsoc2J904
X-Genesys-idSI: dRKudjKoHjsoc2J904
X-Genesys-UD_ConnIDinitial: 017d035166fcad8e
X-Genesys-NDI: XXXXXXXXXX
X-Genesys-TAG_PARCOURS: ACU


<--- Transmitting SIP request (498 bytes) to UDP:92.91.179.72:5062 --->
ACK sip:lucentNGFS-153711@92.91.179.72:5062;x-afi=108 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj0200e69c-3a4d-41a8-a79e-4cfc0b618f41
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=c4a34d7d-6db4-4614-b793-0ef720b97f8c
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org>;tag=649e0815-64e3b80961f7adc-gm-po-lucentPCSF-154063
Call-ID: b5b16872-29a9-4702-a045-fa4946bd1e06
CSeq: 21102 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.4.0
Content-Length:  0


    -- PJSIP/sfr-0000001d answered PJSIP/zoiper-0000001c
<--- Transmitting SIP response (871 bytes) to UDP:192.168.1.11:35801 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:35801;rport=35801;received=192.168.1.11;branch=z9hG4bK-524287-1---fe49247919b5e77e
Call-ID: _AhI2sArpvwjMMeX0TO5WA..
From: <sip:zoiper@192.168.1.11>;tag=a614fb25
To: <sip:1023@192.168.1.11>;tag=cd9a15ab-17aa-461e-947d-1abda9cac4f2
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.1.11:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   296

v=0
o=- 0 1272260 IN IP4 192.168.1.11
s=Asterisk
c=IN IP4 192.168.1.11
t=0 0
m=audio 16642 RTP/AVP 8 0 3 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

    -- Channel PJSIP/sfr-0000001d joined 'simple_bridge' basic-bridge <4f2d3e0f-fc5d-48d5-95d3-102e480f0ab1>
    -- Channel PJSIP/zoiper-0000001c joined 'simple_bridge' basic-bridge <4f2d3e0f-fc5d-48d5-95d3-102e480f0ab1>
<--- Received SIP request (412 bytes) from UDP:192.168.1.11:35801 --->
ACK sip:192.168.1.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:35801;branch=z9hG4bK-524287-1---89020cdbe2b976ca;rport
Max-Forwards: 70
Contact: <sip:zoiper@192.168.1.11:35801;transport=UDP>
To: <sip:1023@192.168.1.11>;tag=cd9a15ab-17aa-461e-947d-1abda9cac4f2
From: <sip:zoiper@192.168.1.11>;tag=a614fb25
Call-ID: _AhI2sArpvwjMMeX0TO5WA..
CSeq: 2 ACK
User-Agent: Z 5.6.1 v2.10.19.9
Content-Length: 0


<--- Received SIP request (697 bytes) from UDP:192.168.1.11:35801 --->
BYE sip:192.168.1.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:35801;branch=z9hG4bK-524287-1---7db94cd73baed6ff;rport
Max-Forwards: 70
Contact: <sip:zoiper@192.168.1.11:35801;transport=UDP>
To: <sip:1023@192.168.1.11>;tag=cd9a15ab-17aa-461e-947d-1abda9cac4f2
From: <sip:zoiper@192.168.1.11>;tag=a614fb25
Call-ID: _AhI2sArpvwjMMeX0TO5WA..
CSeq: 3 BYE
User-Agent: Z 5.6.1 v2.10.19.9
Authorization: Digest username="zoiper",realm="asterisk",nonce="1692645394/a12dce18f09db8a90668c06c1260571e",uri="sip:192.168.1.11:5060",response="1c600aa690bf7d30ef17e39d1d10e514",cnonce="34422c34a60b72853de401e18f8221ff",nc=00000002,qop=auth,algorithm=MD5,opaque="12171fa94d7b77b4"
Content-Length: 0


<--- Transmitting SIP response (345 bytes) to UDP:192.168.1.11:35801 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:35801;rport=35801;received=192.168.1.11;branch=z9hG4bK-524287-1---7db94cd73baed6ff
Call-ID: _AhI2sArpvwjMMeX0TO5WA..
From: <sip:zoiper@192.168.1.11>;tag=a614fb25
To: <sip:1023@192.168.1.11>;tag=cd9a15ab-17aa-461e-947d-1abda9cac4f2
CSeq: 3 BYE
Server: Asterisk PBX 20.4.0
Content-Length:  0


    -- Channel PJSIP/zoiper-0000001c left 'native_rtp' basic-bridge <4f2d3e0f-fc5d-48d5-95d3-102e480f0ab1>
    -- Channel PJSIP/sfr-0000001d left 'native_rtp' basic-bridge <4f2d3e0f-fc5d-48d5-95d3-102e480f0ab1>
  == Spawn extension (outgoing, 1023, 1) exited non-zero on 'PJSIP/zoiper-0000001c'
<--- Transmitting SIP request (522 bytes) to UDP:92.91.179.72:5062 --->
BYE sip:lucentNGFS-153711@92.91.179.72:5062;x-afi=108 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj309df199-d5d9-40a1-944c-b12bc9628e80
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=c4a34d7d-6db4-4614-b793-0ef720b97f8c
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org>;tag=649e0815-64e3b80961f7adc-gm-po-lucentPCSF-154063
Call-ID: b5b16872-29a9-4702-a045-fa4946bd1e06
CSeq: 21105 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 20.4.0
Content-Length:  0


<--- Received SIP response (462 bytes) from UDP:92.91.179.72:5062 --->
SIP/2.0 200 OK
Call-ID: b5b16872-29a9-4702-a045-fa4946bd1e06
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPj309df199-d5d9-40a1-944c-b12bc9628e80;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org>;tag=649e0815-64e3b80961f7adc-gm-po-lucentPCSF-154063
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=c4a34d7d-6db4-4614-b793-0ef720b97f8c
CSeq: 21105 BYE
Date: Mon, 21 Aug 2023 19:16:30 GMT
Content-Length: 0


Debian*CLI>

david551 · August 21, 2023, 7:39pm

You sent it to a different address. I guess one of the addresses to which the name resolves knows about you, and the other doesn’t.

Artemus24 · August 22, 2023, 12:22pm

What ? It doesn’t work, but I already know that.

SFR’s SIP identifiers work because I sometimes have outgoing calls.

Here is the structure of my identifiers:

--> Display Name	: +33XXXXXXXXX.
--> UserName		: NDI0XXXXXXXXX.XXX.THD@sfr.fr
--> Password		: PPPPPPPPPPPPPPPP
--> Domain			: ims.mnc010.mcc208.3gppnetwork.org
--> Proxy Server	: mitry.p-cscf.sfr.net:5062
--> 				: corbas.p-cscf.sfr.net:5062
--> 				: trappes.p-cscf.sfr.net:5062

This is what I got from SFR.

The three proxy servers have been replaced by this one:

					: sip:residential.p-cscf.sfr.net;lr`

If it is the ip addresses of the proxy servers, it is not me who makes this change but SFR.
I only indicate “residential.p-cscf.sfr.net”.

I have the correct credentials since I am getting an outgoing communication. Except that I have 5 communications which fail for a successful one.
I think I must be missing an option, or one of them is not configured correctly.

92.91.179.24 and 92.91.179.72 are the IP addresses of the corbas.p-cscf.sfr.net proxy server.
What should I put as the correct address for the proxy server?

~> dig mitry.p-cscf.sfr.net

; <<>> DiG 9.16.42-Debian <<>> mitry.p-cscf.sfr.net
;; global options: +cmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 45402
;; flags: qr rd ra; QUERY: 1, ANSWER: 5, AUTHORITY: 0, ADDITIONAL: 1

;; OPT PSEUDOSECTION:
; EDNS: version: 0, flags:; udp: 4096
;; QUESTION SECTION:
;mitry.p-cscf.sfr.net.		IN	A

;; ANSWER SECTION:
mitry.p-cscf.sfr.net.	2342	IN	A	92.91.129.40
mitry.p-cscf.sfr.net.	2342	IN	A	92.91.129.56
mitry.p-cscf.sfr.net.	2342	IN	A	92.91.129.8
mitry.p-cscf.sfr.net.	2342	IN	A	92.91.129.24
mitry.p-cscf.sfr.net.	2342	IN	A	92.91.129.72

;; Query time: 0 msec
;; SERVER: 192.168.1.1#53(192.168.1.1)
;; WHEN: Tue Aug 22 14:08:33 CEST 2023
;; MSG SIZE  rcvd: 129

> dig corbas.p-cscf.sfr.net

; <<>> DiG 9.16.42-Debian <<>> corbas.p-cscf.sfr.net
;; global options: +cmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 62657
;; flags: qr rd ra; QUERY: 1, ANSWER: 5, AUTHORITY: 0, ADDITIONAL: 1

;; OPT PSEUDOSECTION:
; EDNS: version: 0, flags:; udp: 4096
;; QUESTION SECTION:
;corbas.p-cscf.sfr.net.		IN	A

;; ANSWER SECTION:
corbas.p-cscf.sfr.net.	2266	IN	A	92.91.179.72
corbas.p-cscf.sfr.net.	2266	IN	A	92.91.179.24
corbas.p-cscf.sfr.net.	2266	IN	A	92.91.179.56
corbas.p-cscf.sfr.net.	2266	IN	A	92.91.179.40
corbas.p-cscf.sfr.net.	2266	IN	A	92.91.179.8

;; Query time: 0 msec
;; SERVER: 192.168.1.1#53(192.168.1.1)
;; WHEN: Tue Aug 22 14:09:49 CEST 2023
;; MSG SIZE  rcvd: 130

~> dig trappes.p-cscf.sfr.net

; <<>> DiG 9.16.42-Debian <<>> trappes.p-cscf.sfr.net
;; global options: +cmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 29152
;; flags: qr rd ra; QUERY: 1, ANSWER: 5, AUTHORITY: 0, ADDITIONAL: 1

;; OPT PSEUDOSECTION:
; EDNS: version: 0, flags:; udp: 4096
;; QUESTION SECTION:
;trappes.p-cscf.sfr.net.		IN	A

;; ANSWER SECTION:
trappes.p-cscf.sfr.net.	2230	IN	A	92.91.129.136
trappes.p-cscf.sfr.net.	2230	IN	A	92.91.129.152
trappes.p-cscf.sfr.net.	2230	IN	A	92.91.129.200
trappes.p-cscf.sfr.net.	2230	IN	A	92.91.129.184
trappes.p-cscf.sfr.net.	2230	IN	A	92.91.129.168

;; Query time: 0 msec
;; SERVER: 192.168.1.1#53(192.168.1.1)
;; WHEN: Tue Aug 22 14:10:26 CEST 2023
;; MSG SIZE  rcvd: 131

~> host -t SRV _sip._udp.residential.p-cscf.sfr.net 109.0.66.10
Using domain server:
Name: 109.0.66.10
Address: 109.0.66.10#53
Aliases: 

_sip._udp.residential.p-cscf.sfr.net has SRV record 10 0 5062 mitry.p-cscf.sfr.net.
_sip._udp.residential.p-cscf.sfr.net has SRV record 10 0 5062 trappes.p-cscf.sfr.net.
_sip._udp.residential.p-cscf.sfr.net has SRV record 10 0 5062 corbas.p-cscf.sfr.net.
>

david551 · August 22, 2023, 12:40pm

If you are following their instructions correctly it is SFR’s problem. They are sending a malformed 407 response to some requests, possibly dependent on which proxy DNS selected. The malformed 407 is a problem, for them, in its own right. I’m guessing it is trying to say “I don’t recognize your IP address”, but that is not a reason for omitting a mandatory header.

tootai · August 22, 2023, 1:24pm

[…]

92.91.179.24 and 92.91.179.72 are the IP addresses of the corbas.p-cscf.sfr.net proxy server.
What should I put as the correct address for the proxy server?

Try each one without hostname. Once you get the working one you will now thich hostname you have to use. You could also leave it w/o hostname.

[…]

Artemus24 · August 23, 2023, 10:40am

The allocation of SFR Proxy servers (Mitry, Corbas, Trappes) and IP addresses (five per server) works by rotation. Each new communication changes the previously assigned IP address. In ipv4, I have fifteen IPv4 addresses to manage, the same for IPv6.

In Asterisk, isn’t there an option to take those who are active?

jcolp · August 23, 2023, 10:44am

There is not. Such a thing has been requested, but noone has implemented it.

Pooh · August 23, 2023, 10:53am

How does Asterisk know which ones are active, if they’re being rotated by the
upstream provider?

Antony.

Artemus24 · August 23, 2023, 11:04am

I have a communication that works with the IPv4 address “92.91.129.136”.

For the “outbound_proxy” option, I replace “residential.p-cscf.sfr.net” with this address. If I put “92.91.129.136:5062”, no registration. If I put “sip:92.91.129.136:5062”, I have the registration but no communication.

Artemus24 · August 23, 2023, 11:10am

I’m using Asterisk 20.4.0 with “chan_pjsip”. Several tutorials work according to their author in an old version of Asterisk (15, 16, 18) under “chan_sip”. How to do the same with “chan_pjsip”?

jcolp · August 23, 2023, 11:11am

The chan_sip module didn’t implement DNS support according to the RFC specifications, you can’t make chan_pjsip behave that way currently. It would require additional coding work.