Asterisk SIP incoming calls rejected

Asterisk Asterisk SIP
1

hello,
I have asterisk 11.13.0
I have Outgoing calls perfect but incoming calls rejected

is an asterisk 11.13.0 with trunk below:
disallow=all
allow=alaw&ulaw
dtmfmode=RFC2833
qualify=yes
type=peer
outboundproxy=as1.romtelecom.net
port=5060
username=2260nnnnnnnnnn@as1.romtelecom.net
secret=nnnnnnnnnnnnnn
fromuser=+40233275173
fromdomain=as1.romtelecom.net
host=as1.romtelecom.net
nat=yes
context=from-trunk

and Register String:
+40233275173:ppppppppppppp:uuuuuuuuuuuuuu@as1.romtelecom.net@as1.romtelecom.net/+40233275173

Reliably Transmitting (NAT) to 92.87.198.105:5060:
OPTIONS sip:as1.romtelecom.net SIP/2.0
Via: SIP/2.0/UDP 2.1.62.186:5060;branch=z9hG4bK35c89689;rport
Max-Forwards: 70
From: “Unknown” sip:+40233275173@2.1.62.186;tag=as36a59441
To: sip:as1.romtelecom.net
Contact: sip:+40233275173@2.1.62.186:5060
Call-ID: 4e5688cc4f332c6e0b34abe5171a893a@2.1.62.186:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.13.0)
Date: Sun, 19 Jan 2025 14:46:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:92.87.198.105:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.1.62.186:5060;branch=z9hG4bK35c89689;rport=5060
Call-ID: 4e5688cc4f332c6e0b34abe5171a893a@2.1.62.186:5060
From: "Unknown"sip:+40233275173@2.1.62.186;tag=as36a59441
To: sip:as1.romtelecom.net;tag=q5i5ccb0
CSeq: 102 OPTIONS
Allow: OPTIONS,NOTIFY,SUBSCRIBE,INFO,REGISTER,MESSAGE,REFER,UPDATE,PRACK,BYE,CANCEL,ACK,INVITE
Supported: privacy,precondition,100rel
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘4e5688cc4f332c6e0b34abe5171a893a@2.1.62.186:5060’ Method: OPTIONS

<— SIP read from UDP:92.87.198.105:5060 —>
INVITE sip:+40233275173@as1.romtelecom.net SIP/2.0
Via: SIP/2.0/UDP 92.87.198.105:5060;branch=z9hG4bKc5bdc1crzpzd1cqdaqmibcnrq;Role=3;Hpt=8f42_36
Call-ID: asbcepjgr5jfs6ug6fgsgjz6fpjp3pseiurd@ATS.bucats01.as1.romtelecom.net.148
From: tel:0723828943;phone-context=+40;tag=fgiiqd6s-CC-148
To: sip:+40233275173@as1.romtelecom.net;user=phone
CSeq: 1 INVITE
Alert-Info: http://www.huawei.com/ring/;info=pattern1
Allow: INVITE,ACK,BYE,CANCEL,UPDATE,INFO,PRACK,NOTIFY,REFER,SUBSCRIBE,OPTIONS,MESSAGE
Contact: sip:40723828943@92.87.198.105:5060;Hpt=8f42_16;CxtId=4;TRC=ffffffff-ffffffff
Max-Forwards: 65
Supported: 100rel,timer
Session-Expires: 1800
Min-SE: 600
P-Asserted-Identity: tel:0723828943;phone-context=+40
P-Called-Party-ID: tel:+40233275173
P-Early-Media: gated
Content-Length: 197
Content-Type: application/sdp

v=0
o=- 61630180 61630180 IN IP4 92.87.198.107
s=SBC call
c=IN IP4 92.87.198.107
t=0 0
m=audio 45362 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
— (18 headers 10 lines) —
Sending to 92.87.198.105:5060 (no NAT)
Sending to 92.87.198.105:5060 (no NAT)
Using INVITE request as basis request - asbcepjgr5jfs6ug6fgsgjz6fpjp3pseiurd@ATS.bucats01.as1.romtelecom.net.148
[2025-01-19 16:46:41] NOTICE[3801][C-0000002f]: chan_sip.c:18331 check_user_full: From address missing ‘sip:’, using it anyway
Found peer ‘TRUNK’ for ‘tel’ from 92.87.198.105:5060

<— Reliably Transmitting (NAT) to 92.87.198.105:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 92.87.198.105:5060;branch=z9hG4bKc5bdc1crzpzd1cqdaqmibcnrq;Role=3;Hpt=8f42_36;received=92.87.198.105;rport=5060
From: tel:0723828943;phone-context=+40;tag=fgiiqd6s-CC-148
To: sip:+40233275173@as1.romtelecom.net;user=phone;tag=as56a02c28
Call-ID: asbcepjgr5jfs6ug6fgsgjz6fpjp3pseiurd@ATS.bucats01.as1.romtelecom.net.148
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“1d40e4c1”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘asbcepjgr5jfs6ug6fgsgjz6fpjp3pseiurd@ATS.bucats01.as1.romtelecom.net.148’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:92.87.198.105:5060 —>
ACK sip:+40233275173@as1.romtelecom.net SIP/2.0
Via: SIP/2.0/UDP 92.87.198.105:5060;branch=z9hG4bKc5bdc1crzpzd1cqdaqmibcnrq;Role=3;Hpt=8f42_36
Call-ID: asbcepjgr5jfs6ug6fgsgjz6fpjp3pseiurd@ATS.bucats01.as1.romtelecom.net.148
From: tel:0723828943;phone-context=+40;tag=fgiiqd6s-CC-148
To: sip:+40233275173@as1.romtelecom.net;user=phone;tag=as56a02c28
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘asbcepjgr5jfs6ug6fgsgjz6fpjp3pseiurd@ATS.bucats01.as1.romtelecom.net.148’ Method: ACK

2

On Sunday 19 January 2025 at 20:10:21, spbelmpas via Asterisk Community wrote:

I have asterisk 11.13.0

https://docs.asterisk.org/About-the-Project/Asterisk-Versions/

allow=alaw&ulaw

Invalid syntax - should be:

allow=alaw
allow=ulaw

Finally, chan_sip is deprecated, no longer developed, no longer supported, and
no longer exists in current Asterisk versions.

Antony.

–
“I estimate there’s a world market for about five computers.”

  • Thomas J Watson, Chairman of IBM
3

allow=alaw,ulaw

would also be valid syntax, and it is possible that & is an undocumented alternative for comma.

As implied by the other responses, the OP needs to retry with Asterisk 20, 21, or 22, and chan_pjsip, for good quality support.

The real problem here is that “secret” should be “remotesecret”, assuming Asterisk 11 is recent enough to support that (otherwise, with chan_sip, you have to use the insecure=invite hack),

4

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.