Not able dial outgoing call

dhanunjaya · August 30, 2024, 2:29pm

below i have pasted my configurations and logs sip debug logs i don’t know why the error coming can any one pls help me

Executing [+919036383353@outbound:1] NoOp(“PJSIP/1001-00000012”, “”) in new stack
– Executing [+919036383353@outbound:2] Dial(“PJSIP/1001-00000012”, “PJSIP/+919036383353@mytrunk,30”) in new stack
– Called PJSIP/+919036383353@mytrunk
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [+919036383353@outbound:3] Hangup(“PJSIP/1001-00000012”, “”) in new stack
== Spawn extension (outbound, +919036383353, 3) exited non-zero on ‘PJSIP/1001-00000012’

— Received SIP request (1034 bytes) from UDP:192.168.1.18:47869 —>
INVITE sip:+919036383353@192.168.1.18;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:47869;branch=z9hG4bK-524287-1—cf75282f72049879;rport
Max-Forwards: 70
Contact: sip:1001@192.168.1.18:47869;transport=UDP
To: sip:+919036383353@192.168.1.18
From: sip:1001@192.168.1.18;transport=UDP;tag=0515c236
Call-ID: 6J9ti6LkDrBzQTzk-9QBJg…
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 5.6.4 v2.10.20.4_1
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 340

v=0
o=Z 0 11702030 IN IP4 192.168.1.18
s=Z
c=IN IP4 192.168.1.18
t=0 0
m=audio 44608 RTP/AVP 106 9 98 101 0 8 3
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux

<— Transmitting SIP response (510 bytes) to UDP:192.168.1.18:47869 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.18:47869;rport=47869;received=192.168.1.18;branch=z9hG4bK-524287-1—cf75282f72049879
Call-ID: 6J9ti6LkDrBzQTzk-9QBJg…
From: sip:1001@192.168.1.18;tag=0515c236
To: sip:+919036383353@192.168.1.18;tag=z9hG4bK-524287-1—cf75282f72049879
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1725026849/46bc34455b888712c4a9cff20735168e”,opaque=“3cd5c5c15750c654”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 20.9.2
Content-Length: 0

<— Received SIP request (367 bytes) from UDP:192.168.1.18:47869 —>
ACK sip:+919036383353@192.168.1.18;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:47869;branch=z9hG4bK-524287-1—cf75282f72049879;rport
Max-Forwards: 70
To: sip:+919036383353@192.168.1.18;tag=z9hG4bK-524287-1—cf75282f72049879
From: sip:1001@192.168.1.18;transport=UDP;tag=0515c236
Call-ID: 6J9ti6LkDrBzQTzk-9QBJg…
CSeq: 1 ACK
Content-Length: 0

<— Received SIP request (1340 bytes) from UDP:192.168.1.18:47869 —>
INVITE sip:+919036383353@192.168.1.18;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:47869;branch=z9hG4bK-524287-1—bad52d3543a01917;rport
Max-Forwards: 70
Contact: sip:1001@192.168.1.18:47869;transport=UDP
To: sip:+919036383353@192.168.1.18
From: sip:1001@192.168.1.18;transport=UDP;tag=0515c236
Call-ID: 6J9ti6LkDrBzQTzk-9QBJg…
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 5.6.4 v2.10.20.4_1
Authorization: Digest username=“1001”,realm=“asterisk”,nonce=“1725026849/46bc34455b888712c4a9cff20735168e”,uri="sip:+919036383353@192.168.1.18;transport=UDP",response=“fd32c86a27686e8c24b222eef7b8aedd”,cnonce=“b24cf86407461cf4984cd1d626ec77dd”,nc=00000001,qop=auth,algorithm=MD5,opaque=“3cd5c5c15750c654”
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 340

v=0
o=Z 0 11702030 IN IP4 192.168.1.18
s=Z
c=IN IP4 192.168.1.18
t=0 0
m=audio 44608 RTP/AVP 106 9 98 101 0 8 3
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux

<— Transmitting SIP response (318 bytes) to UDP:192.168.1.18:47869 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.18:47869;rport=47869;received=192.168.1.18;branch=z9hG4bK-524287-1—bad52d3543a01917
Call-ID: 6J9ti6LkDrBzQTzk-9QBJg…
From: sip:1001@192.168.1.18;tag=0515c236
To: sip:+919036383353@192.168.1.18
CSeq: 2 INVITE
Server: Asterisk PBX 20.9.2
Content-Length: 0

-- Executing [+919036383353@outbound:1] NoOp("PJSIP/1001-0000000c", "") in new stack
-- Executing [+919036383353@outbound:2] Dial("PJSIP/1001-0000000c", "PJSIP/+919036383353@mytrunk,30") in new stack
-- Called PJSIP/+919036383353@mytrunk

== Everyone is busy/congested at this time (1:0/1/0)
– Executing [+919036383353@outbound:3] Hangup(“PJSIP/1001-0000000c”, “”) in new stack
== Spawn extension (outbound, +919036383353, 3) exited non-zero on ‘PJSIP/1001-0000000c’
<— Transmitting SIP response (396 bytes) to UDP:192.168.1.18:47869 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.18:47869;rport=47869;received=192.168.1.18;branch=z9hG4bK-524287-1—bad52d3543a01917
Call-ID: 6J9ti6LkDrBzQTzk-9QBJg…
From: sip:1001@192.168.1.18;tag=0515c236
To: sip:+919036383353@192.168.1.18;tag=fac98a84-cf2f-4e8a-a47b-522894b07338
CSeq: 2 INVITE
Server: Asterisk PBX 20.9.2
Reason: Q.850;cause=34
Content-Length: 0

<— Received SIP request (368 bytes) from UDP:192.168.1.18:47869 —>
ACK sip:+919036383353@192.168.1.18;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:47869;branch=z9hG4bK-524287-1—bad52d3543a01917;rport
Max-Forwards: 70
To: sip:+919036383353@192.168.1.18;tag=fac98a84-cf2f-4e8a-a47b-522894b07338
From: sip:1001@192.168.1.18;transport=UDP;tag=0515c236
Call-ID: 6J9ti6LkDrBzQTzk-9QBJg…
CSeq: 2 ACK
Content-Length: 0

[transport-udp]
type=transport
bind=0.0.0.0:5060
protocol=udp
local_net=192.168.1.18

[mytrunk-reg]
type = registration
retry_interval = 20
max_retries = 10
;contact_user = +918042310210@ka.ims.airtel.in
expiration = 3600
;transport = transport-udp
outbound_auth=mytrunk-auth
client_uri = sip:+918042310210@ka.ims.airtel.in
server_uri = sip:+918042310210@10.5.70.3

[mytrunk-auth]
type = auth
password = Wrktop#1
username = +918042310210@ka.ims.airtel.in

[mytrunk]
type = aor
contact = sip:+918042310210@ka.ims.airtel.in
maximum_expiration = 3600

[mytrunk]
type = identify
endpoint = mytrunk
match = ka.ims.airtel.in

[mytrunk]
type = auth
username = mytrunk
password = Wrktop#1

[mytrunk]
type = endpoint
context = inbound
;dtmf_mode = rfc2833
disallow = all
allow = ulaw,alaw
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
direct_media = no
;inband_progress = yes
;from_domain = ka.ims.airtel.in
auth = mytrunk
;outbound_auth = mytrunk
aors = mytrunk

[1001]
type=endpoint
context=outbound
disallow=all
allow=alaw,ulaw
auth=1001
aors=1001
transport=transport-udp
direct_media=no

[1001]
type=aor
remove_existing=yes
max_contacts=1

[1001]
type=auth
username=1001
password=123456

david551 · August 30, 2024, 2:55pm

Please explain your network topology, in particular what 10/8 represents, how your IP routes are configured to reach it, and why it isn’t listed as a local network.

Something looks wrong if you have configured to send INVITEs to a domain name, but REGISTERs to a private range IP address.

dhanunjaya · August 31, 2024, 4:06pm

nothing much david, we have 1 on premise ubuntu system. that is beside one public network and we connected sip provider to onpremises system through lan cable that’s it

david551 · August 31, 2024, 4:50pm

How is your DNS configured to resolve this to their private address?

The domain doesn’t seem to be valid from public name servers, and public name severs shouldn’t point to private address ranges.

$ dig ns ims.airtel.in
;; communications error to 192.168.1.1#53: timed out

; <<>> DiG 9.18.28-1~deb12u2-Debian <<>> ns ims.airtel.in
;; global options: +cmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 16855
;; flags: qr rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 0, ADDITIONAL: 1

;; OPT PSEUDOSECTION:
; EDNS: version: 0, flags:; udp: 1232
;; QUESTION SECTION:
;ims.airtel.in.			IN	NS

;; ANSWER SECTION:
ims.airtel.in.		0	IN	NS	this.name.is.invalid.

;; Query time: 175 msec
;; SERVER: 192.168.1.1#53(192.168.1.1) (UDP)
;; WHEN: Sat Aug 31 17:48:03 BST 2024
;; MSG SIZE  rcvd: 76

dhanunjaya · August 31, 2024, 4:58pm

yes david they given private address only to reach those address they given lan cable connection so that we can reach to private address i can register to my provider successfullly same connection working before but we restarted the system after that it is not working and also strange thing when i try to register with my old system it is showing new error saying like async dns seerver something so we installed asterisk in new system and now this issue coming

pls david if iam wrong can u explain your question i will give you information about that

dhanunjaya · August 31, 2024, 5:03pm

our setup is like we have a system in that we created 2 ethernets 1 for internet and 1 more for sip provider then we added all the ip address to routing table then we check those ip are reachable or not by pinging them all are reachable

david551 · August 31, 2024, 6:10pm

Your register works because you use the IP address, but the INVITE fails because you use the domain name, and don’t appear to have any means that resolve that to the IP address.

It might be possible to replace the domain name with the IP address. You might be able to use the outbound proxy mechanism to get the request to the right place (possibly using both lr and hide options). You might need define the domain in /etc/hosts, or as a locally mastered domain in your DNS server.

dhanunjaya · September 2, 2024, 5:19am

thanks david as u said i have added outbound_proxy in the registration so i’m able dial calls now