Did the test. It failed. The logs and configs are:
###########################################################
sip.conf
###########################################################
[general]
localnet=192.168.1.0/24
disallow=all ; only used at firt test. Then removed line
register => 9765432:XYZ@sip.inphonex.com/9765432
[provider]
type=peer
context=incoming
host=sip.inphonex.com
defaultuser=9765432
fromuser=9765432
secret=XYZ
canreinvite=no
disallow=all ; only used at firt test. Then removed line
[client]
type=peer
context=outgoing
host=192.168.1.119
canreinvite=no
disallow=all ; only used at firt test. Then removed line
###########################################################
A call log with Asterisk with sip.conf with “disallow=all”
###########################################################
*CLI> == Using SIP RTP CoS mark 5
No compatible codecs, not accepting this offer!
###########################################################
A call log with Asterisk sip.conf with no "disallow=all"
and neither “allow” statements
###########################################################
*CLI> == Using SIP RTP CoS mark 5
Agent policy for SIP/client-00000002 is ‘never’. CC not possible
– Executing [011351211972855@outgoing:1] Dial(“SIP/client-00000002”, “SIP/011351211972855@provider”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/011351211972855@provider
> 0x7f6be8007e70 – Probation passed - setting RTP source address to 93.186.130.50:13116
– SIP/provider-00000003 is ringing
– SIP/provider-00000003 is making progress passing it to SIP/client-00000002
> 0x7f6be8007e70 – Probation passed - setting RTP source address to 93.186.130.50:13116
Unable to find a codec translation path from (gsm) to (ulaw)
> 0x7f6bf000aee0 – Probation passed - setting RTP source address to 192.168.1.119:7078
Unable to find a codec translation path from (gsm) to (ulaw)
Codec mismatch on channel SIP/provider-00000003 setting write format to gsm from ulaw native formats (ulaw)
Unable to find a codec translation path from (ulaw) to (gsm)
Asked to transmit frame type gsm, while native formats is (ulaw) read/write = ulaw/ulaw
Codec mismatch on channel SIP/provider-00000003 setting write format to gsm from ulaw native formats (ulaw)
Unable to find a codec translation path from (ulaw) to (gsm)
Asked to transmit frame type gsm, while native formats is (ulaw) read/write = ulaw/ulaw
Asked to transmit frame type ulaw, while native formats is (gsm) read/write = ulaw/ulaw
Asked to transmit frame type ulaw, while native formats is (gsm) read/write = ulaw/ulaw
Codec mismatch on channel SIP/provider-00000003 setting write format to gsm from ulaw native formats (ulaw)
###########################################################
SIP log of a call
###########################################################
U 2013/11/16 17:39:39.625458 192.168.1.119:5060 -> 192.168.1.160:5060
INVITE sip:011055123456789@192.168.1.119 SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.119:5060;rport;branch=z9hG4bK7323.
From: sip:1001@192.168.1.119;tag=26287.
To: sip:011055123456789@192.168.1.119.
Call-ID: 3032.
CSeq: 20 INVITE.
Contact: sip:1001@192.168.1.119.
Content-Type: application/sdp.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
Max-Forwards: 70.
User-Agent: Linphone/3.6.1 (eXosip2/3.6.0).
Subject: Phone call.
Content-Length: 255.
.
v=0.
o=1001 2462 404 IN IP4 192.168.1.119.
s=Talk.
c=IN IP4 192.168.1.119.
t=0 0.
m=audio 7078 RTP/AVP 0 8 9 3 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:9 G722/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
U 2013/11/16 17:39:39.643092 192.168.1.160:5060 -> 192.168.1.119:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 192.168.1.119:5060;branch=z9hG4bK7323;received=192.168.1.119;rport=5060.
From: sip:1001@192.168.1.119;tag=26287.
To: sip:011055123456789@192.168.1.119.
Call-ID: 3032.
CSeq: 20 INVITE.
Server: Asterisk PBX SVN-branch-11-r402709.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: sip:011055123456789@192.168.1.160:5060.
Content-Length: 0.
.
U 2013/11/16 17:39:39.644543 95.136.84.177:5060 -> 208.239.76.169:5060
INVITE sip:011055123456789@sip.inphonex.com SIP/2.0.
Via: SIP/2.0/UDP 95.136.84.177:5060;branch=z9hG4bK05027122.
Max-Forwards: 70.
From: sip:9765432@95.136.84.177;tag=as0f86965b.
To: sip:011055123456789@sip.inphonex.com.
Contact: sip:9765432@95.136.84.177:5060.
Call-ID: 267e180963e03597310bcee630afc13b@95.136.84.177:5060.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX SVN-branch-11-r402709.
Date: Sat, 16 Nov 2013 17:39:39 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 324.
.
v=0.
o=root 840637508 840637508 IN IP4 95.136.84.177.
s=Asterisk PBX SVN-branch-11-r402709.
c=IN IP4 95.136.84.177.
t=0 0.
m=audio 15970 RTP/AVP 0 3 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
U 2013/11/16 17:39:39.799448 208.239.76.169:5060 -> 95.136.84.177:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 95.136.84.177:5060;branch=z9hG4bK05027122.
From: sip:9765432@95.136.84.177;tag=as0f86965b.
To: sip:011055123456789@sip.inphonex.com.
Call-ID: 267e180963e03597310bcee630afc13b@95.136.84.177:5060.
CSeq: 102 INVITE.
Content-Length: 0.
.
U 2013/11/16 17:39:39.816624 208.239.76.169:5060 -> 95.136.84.177:5060
SIP/2.0 407 Proxy Authentication required.
Via: SIP/2.0/UDP 95.136.84.177:5060;branch=z9hG4bK05027122.
From: sip:9765432@95.136.84.177;tag=as0f86965b.
To: sip:011055123456789@sip.inphonex.com;tag=SD2fr4f99-65007518_8cce6d4f_ba42d4e5-fb47-449f-b4a6-7b6a97ac38fe.
Call-ID: 267e180963e03597310bcee630afc13b@95.136.84.177:5060.
CSeq: 102 INVITE.
Contact: sip:208.239.76.169:5060;transport=udp.
Proxy-Authenticate: Digest realm=“95.136.84.177”,nonce=“0043ef2a3de570c2e55a6ba8d3effdf8”,opaque=“652dcd2fad06ba2116a65d73b47ab9fd”.
Content-Length: 0.
.
U 2013/11/16 17:39:39.816895 95.136.84.177:5060 -> 208.239.76.169:5060
ACK sip:011055123456789@sip.inphonex.com SIP/2.0.
Via: SIP/2.0/UDP 95.136.84.177:5060;branch=z9hG4bK05027122.
Max-Forwards: 70.
From: sip:9765432@95.136.84.177;tag=as0f86965b.
To: sip:011055123456789@sip.inphonex.com;tag=SD2fr4f99-65007518_8cce6d4f_ba42d4e5-fb47-449f-b4a6-7b6a97ac38fe.
Contact: sip:9765432@95.136.84.177:5060.
Call-ID: 267e180963e03597310bcee630afc13b@95.136.84.177:5060.
CSeq: 102 ACK.
User-Agent: Asterisk PBX SVN-branch-11-r402709.
Content-Length: 0.
.
U 2013/11/16 17:39:39.817259 95.136.84.177:5060 -> 208.239.76.169:5060
INVITE sip:011055123456789@sip.inphonex.com SIP/2.0.
Via: SIP/2.0/UDP 95.136.84.177:5060;branch=z9hG4bK6a64880e.
Max-Forwards: 70.
From: sip:9765432@95.136.84.177;tag=as0f86965b.
To: sip:011055123456789@sip.inphonex.com.
Contact: sip:9765432@95.136.84.177:5060.
Call-ID: 267e180963e03597310bcee630afc13b@95.136.84.177:5060.
CSeq: 103 INVITE.
User-Agent: Asterisk PBX SVN-branch-11-r402709.
Proxy-Authorization: Digest username=“9765432”, realm=“95.136.84.177”, algorithm=MD5, uri="sip:011055123456789@sip.inphonex.com", nonce=“0043ef2a3de570c2e55a6ba8d3effdf8”, response=“5dcee625fa291a3299706baf62fd48ea”, opaque=“652dcd2fad06ba2116a65d73b47ab9fd”.
Date: Sat, 16 Nov 2013 17:39:39 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 324.
.
v=0.
o=root 840637508 840637509 IN IP4 95.136.84.177.
s=Asterisk PBX SVN-branch-11-r402709.
c=IN IP4 95.136.84.177.
t=0 0.
m=audio 15970 RTP/AVP 0 3 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
U 2013/11/16 17:39:39.973010 208.239.76.169:5060 -> 95.136.84.177:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 95.136.84.177:5060;branch=z9hG4bK6a64880e.
From: sip:9765432@95.136.84.177;tag=as0f86965b.
To: sip:011055123456789@sip.inphonex.com.
Call-ID: 267e180963e03597310bcee630afc13b@95.136.84.177:5060.
CSeq: 103 INVITE.
Content-Length: 0.
.
U 2013/11/16 17:39:43.127392 208.239.76.169:5060 -> 95.136.84.177:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 95.136.84.177:5060;branch=z9hG4bK6a64880e.
From: sip:9765432@95.136.84.177;tag=as0f86965b.
To: sip:011055123456789@sip.inphonex.com;tag=SD2fr4f99-74378198_8cce6d4f_b2b10335-6bc5-46fe-8c47-a7b299167377.
Call-ID: 267e180963e03597310bcee630afc13b@95.136.84.177:5060.
CSeq: 103 INVITE.
Contact: sip:208.239.76.169:5060;transport=udp.
P-Early-Media: sendrecv.
X-UUID: 5bd320bafa524e69a6387e3d687ff004.
Accept: application/sdp.
Allow: INVITE.
Content-Type: application/sdp.
Content-Length: 207.
.
v=0.
o=- 30625 0 IN IP4 93.186.130.50.
s=IMSS.
c=IN IP4 93.186.130.50.
t=0 0.
m=audio 13116 RTP/AVP 0 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=sqn:0.
a=cdsc: 1 image udptl t38.
U 2013/11/16 17:39:43.128317 192.168.1.160:5060 -> 192.168.1.119:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.1.119:5060;branch=z9hG4bK7323;received=192.168.1.119;rport=5060.
From: sip:1001@192.168.1.119;tag=26287.
To: sip:011055123456789@192.168.1.119;tag=as24b3eff7.
Call-ID: 3032.
CSeq: 20 INVITE.
Server: Asterisk PBX SVN-branch-11-r402709.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: sip:011055123456789@192.168.1.160:5060.
Content-Length: 0.
.
U 2013/11/16 17:39:43.128604 192.168.1.160:5060 -> 192.168.1.119:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 192.168.1.119:5060;branch=z9hG4bK7323;received=192.168.1.119;rport=5060.
From: sip:1001@192.168.1.119;tag=26287.
To: sip:011055123456789@192.168.1.119;tag=as24b3eff7.
Call-ID: 3032.
CSeq: 20 INVITE.
Server: Asterisk PBX SVN-branch-11-r402709.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: sip:011055123456789@192.168.1.160:5060.
Content-Type: application/sdp.
Content-Length: 324.
.
v=0.
o=root 985460714 985460714 IN IP4 192.168.1.160.
s=Asterisk PBX SVN-branch-11-r402709.
c=IN IP4 192.168.1.160.
t=0 0.
m=audio 15688 RTP/AVP 3 0 8 101.
a=rtpmap:3 GSM/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
U 2013/11/16 17:39:45.019159 192.168.1.119:5060 -> 192.168.1.160:5060
CANCEL sip:011055123456789@192.168.1.119 SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.119:5060;rport;branch=z9hG4bK7323.
From: sip:1001@192.168.1.119;tag=26287.
To: sip:011055123456789@192.168.1.119.
Call-ID: 3032.
CSeq: 20 CANCEL.
Max-Forwards: 70.
User-Agent: Linphone/3.6.1 (eXosip2/3.6.0).
Content-Length: 0.
.
U 2013/11/16 17:39:45.019424 192.168.1.160:5060 -> 192.168.1.119:5060
SIP/2.0 487 Request Terminated.
Via: SIP/2.0/UDP 192.168.1.119:5060;branch=z9hG4bK7323;received=192.168.1.119;rport=5060.
From: sip:1001@192.168.1.119;tag=26287.
To: sip:011055123456789@192.168.1.119;tag=as24b3eff7.
Call-ID: 3032.
CSeq: 20 INVITE.
Server: Asterisk PBX SVN-branch-11-r402709.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Length: 0.
.
U 2013/11/16 17:39:45.019581 192.168.1.160:5060 -> 192.168.1.119:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.1.119:5060;branch=z9hG4bK7323;received=192.168.1.119;rport=5060.
From: sip:1001@192.168.1.119;tag=26287.
To: sip:011055123456789@192.168.1.119;tag=as24b3eff7.
Call-ID: 3032.
CSeq: 20 CANCEL.
Server: Asterisk PBX SVN-branch-11-r402709.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Length: 0.
.
U 2013/11/16 17:39:45.020062 95.136.84.177:5060 -> 208.239.76.169:5060
CANCEL sip:011055123456789@sip.inphonex.com SIP/2.0.
Via: SIP/2.0/UDP 95.136.84.177:5060;branch=z9hG4bK6a64880e.
Max-Forwards: 70.
From: sip:9765432@95.136.84.177;tag=as0f86965b.
To: sip:011055123456789@sip.inphonex.com.
Call-ID: 267e180963e03597310bcee630afc13b@95.136.84.177:5060.
CSeq: 103 CANCEL.
User-Agent: Asterisk PBX SVN-branch-11-r402709.
Content-Length: 0.
.
U 2013/11/16 17:39:45.024589 192.168.1.119:5060 -> 192.168.1.160:5060
ACK sip:011055123456789@192.168.1.119 SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.119:5060;rport;branch=z9hG4bK7323.
Route: sip:192.168.1.160;lr.
From: sip:1001@192.168.1.119;tag=26287.
To: sip:011055123456789@192.168.1.119;tag=as24b3eff7.
Call-ID: 3032.
CSeq: 20 ACK.
Content-Length: 0.
.
U 2013/11/16 17:39:45.173483 208.239.76.169:5060 -> 95.136.84.177:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 95.136.84.177:5060;branch=z9hG4bK6a64880e.
From: sip:9765432@95.136.84.177;tag=as0f86965b.
To: sip:011055123456789@sip.inphonex.com;tag=SD2fr4f99-74378198_8cce6d4f_b2b10335-6bc5-46fe-8c47-a7b299167377.
Call-ID: 267e180963e03597310bcee630afc13b@95.136.84.177:5060.
CSeq: 103 CANCEL.
Content-Length: 0.
.
U 2013/11/16 17:39:45.179262 208.239.76.169:5060 -> 95.136.84.177:5060
SIP/2.0 487 Request Terminated.
Via: SIP/2.0/UDP 95.136.84.177:5060;branch=z9hG4bK6a64880e.
From: sip:9765432@95.136.84.177;tag=as0f86965b.
To: sip:011055123456789@sip.inphonex.com;tag=SD2fr4f99-74378198_8cce6d4f_b2b10335-6bc5-46fe-8c47-a7b299167377.
Call-ID: 267e180963e03597310bcee630afc13b@95.136.84.177:5060.
CSeq: 103 INVITE.
Contact: sip:208.239.76.169:5060;transport=udp.
Content-Length: 0.
.
U 2013/11/16 17:39:45.179387 95.136.84.177:5060 -> 208.239.76.169:5060
ACK sip:208.239.76.169:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 95.136.84.177:5060;branch=z9hG4bK6a64880e.
Max-Forwards: 70.
From: sip:9765432@95.136.84.177;tag=as0f86965b.
To: sip:011055123456789@sip.inphonex.com;tag=SD2fr4f99-74378198_8cce6d4f_b2b10335-6bc5-46fe-8c47-a7b299167377.
Contact: sip:9765432@95.136.84.177:5060.
Call-ID: 267e180963e03597310bcee630afc13b@95.136.84.177:5060.
CSeq: 103 ACK.
User-Agent: Asterisk PBX SVN-branch-11-r402709.
Content-Length: 0.