Asterisk 13 - Cannot make outbound call through trunk

Asterisk Asterisk Support
1

Hello! I had asterisk 11.6-cert2 and it works fine. Now I’ve made second server with Asterisk 13.4 and trying to configure it like old one. Everything ok, BUT I cannot make outbound call through trunk.

There is debug from working asterisk:
<— SIP read from UDP:10.30.1.143:5060 —>
INVITE sip:7916*******@10.30.1.141;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.143:5060;branch=z9hG4bK2140019459326311564
From: 9995 sip:9995@10.30.1.141;tag=42719609
To: “7916*******” sip:7916*******@10.30.1.141;user=phone
Call-ID: 8383154237863-88711949727093@10.30.1.143
CSeq: 1 INVITE
Contact: sip:9995@10.30.1.143:5060
Max-Forwards: 70
Supported: replaces, join, path
User-Agent: Fanvil C62 2.3.752.385
P-Early-Media: supported
Allow-Events: talk
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 263

Where:

  • 7916******* - number I calling
  • 9995 - phone number from I make call
  • 10.30.1.143 - phone local IP
  • 10.30.1.141 - asterisk local IP

Now I make call with debug from new asterisk:
<— SIP read from UDP:10.30.1.137:5060 —>
INVITE sip:7916*******@9997;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.137:5060;branch=z9hG4bK2836512428195665836
From: 9997 sip:9997@9997;tag=353521267
To: “7916*******” sip:7916*******@9997;user=phone
Call-ID: 292051690823212-230482114013394@10.30.1.137
CSeq: 1 INVITE
Contact: sip:9997@10.30.1.137:5060
Max-Forwards: 70
Supported: replaces, join, path
User-Agent: Fanvil C62 2.3.498.267
P-Early-Media: supported
Allow-Events: talk
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 260

And where:

  • 7916******* - number I calling
  • 9997 - phone number from I make call
  • 10.30.1.137 - phone local IP

And there are some differences:
WORKING: From: 9995 sip:9995@10.30.1.141;tag=42719609
WORKING: To: “7916*******” sip:7916*******@10.30.1.141;user=phone

NOT_WORKING: From: 9997 sip:9997@9997;tag=353521267
NOT_WORKING: To: “7916*******” sip:7916*******@9997;user=phone

I can not understand WHY… Please help me. Also, Asterisk 13 gives next error on outbound calls:

-- Executing [79165157443@msk-test-out:1] NoOp("SIP/9997-00000004", "outgoint call!") in new stack
-- Executing [79165157443@msk-test-out:2] Dial("SIP/9997-00000004", "SIP/7916*******@150511") in new stack

Really destroying SIP dialog ‘328d48d96a15cbe25cd232d97cb9543c@192.168.0.121:5060’ Method: INVITE
[Sep 14 12:38:06] WARNING[10632][C-00000004]: app_dial.c:2381 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/9997-00000004’ status is ‘CHANUNAVAIL’

p.s. 150511 - name of trunk
p.p.s. both asterisks have SAME configurations of sips, trunks, extensions and etc.
p.p.p.s. if you need more debug information and/or sip/trunk conf info - let me know please

2

Hm… I dont know why, but trunk SIP is unreachable. If I change qualify to “no”, then it will be unmonitored and outbound call returns:
[Sep 14 12:51:48] NOTICE[13297][C-00000000]: chan_sip.c:23283 handle_response_invite: Failed to authenticate on INVITE to ‘“My Phone 2 Right” sip:9997@sip.zadarma.com;tag=as1f0f0bc4’

3

We’ll need the configuration (minus passwords) and the sip debug in order to determine what is going on.

1 Like
4

[details=SIP.conf (not whole):] [general]
rtptimeout=60
rtpholdtimeout=600
session-timers=accept
;session-timers=refuse
session-expires=1800

session-minse=90
session-refresher=uas

context=public                  ; Default context for incoming calls. Defaults to 'default'
limitonpeers=yes
allowguest=no                  ; Allow or reject guest calls (default is yes)[/details]

[details=SIP_PHONES.conf] sysadm
deny=0.0.0.0/0.0.0.0
secret=password
dtmfmode=rfc2833
canreinvite=no
context=msk-test-out
nat=no
host=dynamic
type=friend
port=5060
qualify=yes
callgroup=33
pickupgroup=33
disallow=all
allow=g729
allow=alaw
permit=0.0.0.0/0.0.0.0
call-limit=50
directmedia=yes

[9997](sysadm)
secret=4321
dial=SIP/9997
description=Peter
mailbox=9997@device
callerid="Mans Phone 2 Right"<9997>[/details]

[details=SIP_TRUNK.CONF] register=>150511:password@sip.zadarma.com

[150511]
type=friend
context=msk-test-out
dtmfmode=rfc2833
canreinvite=no
insecure=port,invite
fromdomain=sip.zadarma.com
host=sip.zadarma.com
disallow=all
allow=ulaw,alaw
username=*123 (changed)*
trunkname=*123 (changed)*
secret=*password (changed)*
qualify=400
directmedia=yes[/details]

[details=extensions.conf] [msk-test-out]
exten => _7XXXXXXXXXX,1,NoOp(outgoint call!)
same => n,Dial(SIP/${EXTEN}@150511)
[/details]

Debug text from asterisk 13.4 (new) for sip 9997 (when I making outbound call) - NOT WORKING

<— SIP read from UDP:10.30.1.137:5060 —>
INVITE sip:79165157443@9997;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.137:5060;branch=z9hG4bK20756267642246915542
From: 9997 sip:9997@9997;tag=1943821779
To: “79165157443” sip:79165157443@9997;user=phone
Call-ID: 1515125555042-3396856724625@10.30.1.137
CSeq: 1 INVITE
Contact: sip:9997@10.30.1.137:5060
Max-Forwards: 70
Supported: replaces, join, path
User-Agent: Fanvil C62 2.3.498.267
P-Early-Media: supported
Allow-Events: talk
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 261

v=0
o=9997 638732438 199567457 IN IP4 10.30.1.137
s=A conversation
c=IN IP4 10.30.1.137
t=0 0
m=audio 10330 RTP/AVP 8 18 101
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=sendrecv
<------------->
— (15 headers 12 lines) —
Sending to 10.30.1.137:5060 (no NAT)
Sending to 10.30.1.137:5060 (no NAT)
Using INVITE request as basis request - 1515125555042-3396856724625@10.30.1.137
Found peer ‘9997’ for ‘9997’ from 10.30.1.137:5060

<— Reliably Transmitting (no NAT) to 10.30.1.137:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.30.1.137:5060;branch=z9hG4bK20756267642246915542;received=10.30.1.137
From: 9997 sip:9997@9997;tag=1943821779
To: “79165157443” sip:79165157443@9997;user=phone;tag=as7577e0ca
Call-ID: 1515125555042-3396856724625@10.30.1.137
CSeq: 1 INVITE
Server: Asterisk PBX 13.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="5118ddff"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1515125555042-3396856724625@10.30.1.137’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:10.30.1.137:5060 —>
ACK sip:79165157443@9997;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.137:5060;branch=z9hG4bK20756267642246915542
From: 9997 sip:9997@9997;tag=1943821779
To: “79165157443” sip:79165157443@9997;user=phone;tag=as7577e0ca
Call-ID: 1515125555042-3396856724625@10.30.1.137
CSeq: 1 ACK
Contact: sip:9997@10.30.1.137:5060
Max-Forwards: 70
Content-Length: 0

<------------->
— (9 headers 0 lines) —

<— SIP read from UDP:10.30.1.137:5060 —>
INVITE sip:79165157443@9997;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.137:5060;branch=z9hG4bK13120176511639020328
From: 9997 sip:9997@9997;tag=1943821779
To: “79165157443” sip:79165157443@9997;user=phone
Call-ID: 1515125555042-3396856724625@10.30.1.137
CSeq: 2 INVITE
Contact: sip:9997@10.30.1.137:5060
Authorization: Digest username=“9997”, realm=“asterisk”, nonce=“5118ddff”, uri=“sip:79165157443@9997;user=phone”, response=“158d2588aac4493f612b26c00614df28”, algorithm=MD5
Max-Forwards: 70
Supported: replaces, join, path
User-Agent: Fanvil C62 2.3.498.267
P-Early-Media: supported
Allow-Events: talk
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 261

v=0
o=9997 638732438 199567457 IN IP4 10.30.1.137
s=A conversation
c=IN IP4 10.30.1.137
t=0 0
m=audio 10330 RTP/AVP 8 18 101
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=sendrecv
<------------->
— (16 headers 12 lines) —
Sending to 10.30.1.137:5060 (no NAT)
Using INVITE request as basis request - 1515125555042-3396856724625@10.30.1.137
Found peer ‘9997’ for ‘9997’ from 10.30.1.137:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (g729|alaw), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (g729|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.30.1.137:10330
Looking for 79165157443 in msk-test-out (domain 9997)
sip_route_dump: route/path hop: sip:9997@10.30.1.137:5060

<— Transmitting (no NAT) to 10.30.1.137:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.30.1.137:5060;branch=z9hG4bK13120176511639020328;received=10.30.1.137
From: 9997 sip:9997@9997;tag=1943821779
To: “79165157443” sip:79165157443@9997;user=phone
Call-ID: 1515125555042-3396856724625@10.30.1.137
CSeq: 2 INVITE
Server: Asterisk PBX 13.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:79165157443@192.168.0.121:5060
Content-Length: 0

<------------>
– Executing [79165157443@msk-test-out:1] NoOp(“SIP/9997-00000012”, “outgoint call!”) in new stack
– Executing [79165157443@msk-test-out:2] Dial(“SIP/9997-00000012”, “SIP/79165157443@150511”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/79165157443@150511
[Sep 14 13:40:06] NOTICE[14403][C-0000000a]: chan_sip.c:23283 handle_response_invite: Failed to authenticate on INVITE to ‘“Mans Phone 2 Right” sip:9997@sip.zadarma.com;tag=as3bde6014’
– SIP/150511-00000013 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Auto fallthrough, channel ‘SIP/9997-00000012’ status is ‘CONGESTION’

<— Reliably Transmitting (no NAT) to 10.30.1.137:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.30.1.137:5060;branch=z9hG4bK13120176511639020328;received=10.30.1.137
From: 9997 sip:9997@9997;tag=1943821779
To: “79165157443” sip:79165157443@9997;user=phone;tag=as25ea52cb
Call-ID: 1515125555042-3396856724625@10.30.1.137
CSeq: 2 INVITE
Server: Asterisk PBX 13.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0

<------------>

<— SIP read from UDP:10.30.1.137:5060 —>
ACK sip:79165157443@9997;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.137:5060;branch=z9hG4bK13120176511639020328
From: 9997 sip:9997@9997;tag=1943821779
To: “79165157443” sip:79165157443@9997;user=phone;tag=as25ea52cb
Call-ID: 1515125555042-3396856724625@10.30.1.137
CSeq: 2 ACK
Contact: sip:9997@10.30.1.137:5060
Max-Forwards: 70
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘1515125555042-3396856724625@10.30.1.137’ Method: ACK[/details]

[details=Debug text from asterisk 11.6 (old) - EVERYTING OK]<— SIP read from UDP:10.30.1.143:5060 —>
INVITE sip:79165157443@10.30.1.141;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.143:5060;branch=z9hG4bK1863526357288857172
From: 9995 sip:9995@10.30.1.141;tag=2042624222
To: “79165157443” sip:79165157443@10.30.1.141;user=phone
Call-ID: 21314320024181-192819468981@10.30.1.143
CSeq: 1 INVITE
Contact: sip:9995@10.30.1.143:5060
Max-Forwards: 70
Supported: replaces, join, path
User-Agent: Fanvil C62 2.3.752.385
P-Early-Media: supported
Allow-Events: talk
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 262

v=0
o=9995 1818128517 279479643 IN IP4 10.30.1.143
s=A conversation
c=IN IP4 10.30.1.143
t=0 0
m=audio 10048 RTP/AVP 8 18 101
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=sendrecv
<------------->
— (15 headers 12 lines) —
Sending to 10.30.1.143:5060 (no NAT)
Sending to 10.30.1.143:5060 (no NAT)
Using INVITE request as basis request - 21314320024181-192819468981@10.30.1.143
Found peer ‘9995’ for ‘9995’ from 10.30.1.143:5060

<— Reliably Transmitting (no NAT) to 10.30.1.143:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.30.1.143:5060;branch=z9hG4bK1863526357288857172;received=10.30.1.143
From: 9995 sip:9995@10.30.1.141;tag=2042624222
To: “79165157443” sip:79165157443@10.30.1.141;user=phone;tag=as2995b89a
Call-ID: 21314320024181-192819468981@10.30.1.143
CSeq: 1 INVITE
Server: Asterisk PBX 11.6-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="32f5cd44"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘21314320024181-192819468981@10.30.1.143’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:10.30.1.143:5060 —>
ACK sip:79165157443@10.30.1.141;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.143:5060;branch=z9hG4bK1863526357288857172
From: 9995 sip:9995@10.30.1.141;tag=2042624222
To: “79165157443” sip:79165157443@10.30.1.141;user=phone;tag=as2995b89a
Call-ID: 21314320024181-192819468981@10.30.1.143
CSeq: 1 ACK
Contact: sip:9995@10.30.1.143:5060
Max-Forwards: 70
Content-Length: 0

<------------->
— (9 headers 0 lines) —

<— SIP read from UDP:10.30.1.143:5060 —>
INVITE sip:79165157443@10.30.1.141;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.143:5060;branch=z9hG4bK2866916640229912295
From: 9995 sip:9995@10.30.1.141;tag=2042624222
To: “79165157443” sip:79165157443@10.30.1.141;user=phone
Call-ID: 21314320024181-192819468981@10.30.1.143
CSeq: 2 INVITE
Contact: sip:9995@10.30.1.143:5060
Authorization: Digest username=“9995”, realm=“asterisk”, nonce=“32f5cd44”, uri="sip:79165157443@10.30.1.141;user=phone", response=“84250c92826070f7ce9e359fd4eba7f9”, algorithm=MD5
Max-Forwards: 70
Supported: replaces, join, path
User-Agent: Fanvil C62 2.3.752.385
P-Early-Media: supported
Allow-Events: talk
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 262

v=0
o=9995 1818128517 279479643 IN IP4 10.30.1.143
s=A conversation
c=IN IP4 10.30.1.143
t=0 0
m=audio 10048 RTP/AVP 8 18 101
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=sendrecv
<------------->
— (16 headers 12 lines) —
Sending to 10.30.1.143:5060 (no NAT)
Using INVITE request as basis request - 21314320024181-192819468981@10.30.1.143
Found peer ‘9995’ for ‘9995’ from 10.30.1.143:5060
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|g729), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.30.1.143:10048
Looking for 79165157443 in msk-test-out (domain 10.30.1.141)
list_route: hop: sip:9995@10.30.1.143:5060

<— Transmitting (no NAT) to 10.30.1.143:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.30.1.143:5060;branch=z9hG4bK2866916640229912295;received=10.30.1.143
From: 9995 sip:9995@10.30.1.141;tag=2042624222
To: “79165157443” sip:79165157443@10.30.1.141;user=phone
Call-ID: 21314320024181-192819468981@10.30.1.143
CSeq: 2 INVITE
Server: Asterisk PBX 11.6-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:79165157443@10.30.1.141:5060
Content-Length: 0

<------------>

<— SIP read from UDP:10.30.1.143:5060 —>
CANCEL sip:79165157443@10.30.1.141;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.143:5060;branch=z9hG4bK2866916640229912295
From: 9995 sip:9995@10.30.1.141;tag=2042624222
To: “79165157443” sip:79165157443@10.30.1.141;user=phone
Call-ID: 21314320024181-192819468981@10.30.1.143
CSeq: 2 CANCEL
Max-Forwards: 70
User-Agent: Fanvil C62 2.3.752.385
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 10.30.1.143:5060 (no NAT)

<— Reliably Transmitting (no NAT) to 10.30.1.143:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.30.1.143:5060;branch=z9hG4bK2866916640229912295;received=10.30.1.143
From: 9995 sip:9995@10.30.1.141;tag=2042624222
To: “79165157443” sip:79165157443@10.30.1.141;user=phone;tag=as0265afe8
Call-ID: 21314320024181-192819468981@10.30.1.143
CSeq: 2 INVITE
Server: Asterisk PBX 11.6-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>

<— Transmitting (no NAT) to 10.30.1.143:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.30.1.143:5060;branch=z9hG4bK2866916640229912295;received=10.30.1.143
From: 9995 sip:9995@10.30.1.141;tag=2042624222
To: “79165157443” sip:79165157443@10.30.1.141;user=phone;tag=as0265afe8
Call-ID: 21314320024181-192819468981@10.30.1.143
CSeq: 2 CANCEL
Server: Asterisk PBX 11.6-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>

<— SIP read from UDP:10.30.1.143:5060 —>
ACK sip:79165157443@10.30.1.141;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.143:5060;branch=z9hG4bK2866916640229912295
From: 9995 sip:9995@10.30.1.141;tag=2042624222
To: “79165157443” sip:79165157443@10.30.1.141;user=phone;tag=as0265afe8
Call-ID: 21314320024181-192819468981@10.30.1.143
CSeq: 2 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘21314320024181-192819468981@10.30.1.143’ Method: ACK

Both asterisks have same configs.

5

Are you limiting the debug… because I don’t see any sip messages for your outgoing call attempt, just incoming.

1 Like
6

I limited by “sip set debug peer PEER_NAME”. Shall I make whole debug without “peer PEER_NAME”?

7

Yes, that would be more useful.

1 Like
8

I think asterisk tries make call to mobile phone via SIP of real phone instead of TRUNK…

[details=Asterisk 13.4 debug]test_pbx*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:10.30.1.138:56032 --->

<------------->

<--- SIP read from UDP:10.30.1.137:5060 --->
INVITE sip:79165157443@9997;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.137:5060;branch=z9hG4bK1126019946365411618
From: 9997 <sip:9997@9997>;tag=2667919259
To: "79165157443" <sip:79165157443@9997;user=phone>
Call-ID: 156071691620432-17914465126364@10.30.1.137
CSeq: 1 INVITE
Contact: <sip:9997@10.30.1.137:5060>
Max-Forwards: 70
Supported: replaces, join, path
User-Agent: Fanvil C62 2.3.498.267
P-Early-Media: supported
Allow-Events: talk
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 262

v=0
o=9997 1083525097 225817361 IN IP4 10.30.1.137
s=A conversation
c=IN IP4 10.30.1.137
t=0 0
m=audio 10346 RTP/AVP 8 18 101
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=sendrecv
<------------->
--- (15 headers 12 lines) ---
Sending to 10.30.1.137:5060 (no NAT)
Sending to 10.30.1.137:5060 (no NAT)
Using INVITE request as basis request - 156071691620432-17914465126364@10.30.1.137
Found peer '9997' for '9997' from 10.30.1.137:5060

<--- Reliably Transmitting (no NAT) to 10.30.1.137:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.30.1.137:5060;branch=z9hG4bK1126019946365411618;received=10.30.1.137
From: 9997 <sip:9997@9997>;tag=2667919259
To: "79165157443" <sip:79165157443@9997;user=phone>;tag=as17cee168
Call-ID: 156071691620432-17914465126364@10.30.1.137
CSeq: 1 INVITE
Server: Asterisk PBX 13.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23355037"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '156071691620432-17914465126364@10.30.1.137' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:10.30.1.137:5060 --->
ACK sip:79165157443@9997;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.137:5060;branch=z9hG4bK1126019946365411618
From: 9997 <sip:9997@9997>;tag=2667919259
To: "79165157443" <sip:79165157443@9997;user=phone>;tag=as17cee168
Call-ID: 156071691620432-17914465126364@10.30.1.137
CSeq: 1 ACK
Contact: <sip:9997@10.30.1.137:5060>
Max-Forwards: 70
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:10.30.1.137:5060 --->
INVITE sip:79165157443@9997;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.137:5060;branch=z9hG4bK112819761878226383
From: 9997 <sip:9997@9997>;tag=2667919259
To: "79165157443" <sip:79165157443@9997;user=phone>
Call-ID: 156071691620432-17914465126364@10.30.1.137
CSeq: 2 INVITE
Contact: <sip:9997@10.30.1.137:5060>
Authorization: Digest username="9997", realm="asterisk", nonce="23355037", uri="sip:79165157443@9997;user=phone", response="387a23aa0ff963a01d5ade00ab3f2b93", algorithm=MD5
Max-Forwards: 70
Supported: replaces, join, path
User-Agent: Fanvil C62 2.3.498.267
P-Early-Media: supported
Allow-Events: talk
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 262

v=0
o=9997 1083525097 225817361 IN IP4 10.30.1.137
s=A conversation
c=IN IP4 10.30.1.137
t=0 0
m=audio 10346 RTP/AVP 8 18 101
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=sendrecv
<------------->
--- (16 headers 12 lines) ---
Sending to 10.30.1.137:5060 (no NAT)
Using INVITE request as basis request - 156071691620432-17914465126364@10.30.1.137
Found peer '9997' for '9997' from 10.30.1.137:5060
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (g729|alaw), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (g729|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.30.1.137:10346
Looking for 79165157443 in msk-test-out (domain 9997)
sip_route_dump: route/path hop: <sip:9997@10.30.1.137:5060>

<--- Transmitting (no NAT) to 10.30.1.137:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.30.1.137:5060;branch=z9hG4bK112819761878226383;received=10.30.1.137
From: 9997 <sip:9997@9997>;tag=2667919259
To: "79165157443" <sip:79165157443@9997;user=phone>
Call-ID: 156071691620432-17914465126364@10.30.1.137
CSeq: 2 INVITE
Server: Asterisk PBX 13.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:79165157443@192.168.0.121:5060>
Content-Length: 0


<------------>
    -- Executing [79165157443@msk-test-out:1] NoOp("SIP/9997-00000020", "outgoint call!") in new stack
    -- Executing [79165157443@msk-test-out:2] Dial("SIP/9997-00000020", "SIP/79165157443@150511") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 10336
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 185.45.152.161:5060:
INVITE sip:79165157443@sip.zadarma.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK12d0fdad
Max-Forwards: 70
From: "Mans Phone 2 Right" <sip:9997@sip.zadarma.com>;tag=as2e260001
To: <sip:79165157443@sip.zadarma.com>
Contact: <sip:9997@192.168.0.121:5060>
Call-ID: 5ccb883647807f1f3607700846a4ac54@sip.zadarma.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.4.0
Date: Wed, 14 Sep 2016 11:15:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 651708801 651708801 IN IP4 192.168.0.121
s=Asterisk PBX 13.4.0
c=IN IP4 192.168.0.121
t=0 0
m=audio 10336 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
    -- Called SIP/79165157443@150511

<--- SIP read from UDP:185.45.152.161:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK12d0fdad;rport=53718;received=62.140.236.228
From: "Mans Phone 2 Right" <sip:9997@sip.zadarma.com>;tag=as2e260001
To: <sip:79165157443@sip.zadarma.com>;tag=bcc9de9cd72655a99664e3be44d308f3.27f8
Call-ID: 5ccb883647807f1f3607700846a4ac54@sip.zadarma.com
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="V9kyZVfZMTkbqHcQSIkdkXXQrfaXGvcr", qop="auth"
Server: Zadarma server
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 185.45.152.161:5060:
ACK sip:79165157443@sip.zadarma.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK12d0fdad
Max-Forwards: 70
From: "Mans Phone 2 Right" <sip:9997@sip.zadarma.com>;tag=as2e260001
To: <sip:79165157443@sip.zadarma.com>;tag=bcc9de9cd72655a99664e3be44d308f3.27f8
Contact: <sip:9997@192.168.0.121:5060>
Call-ID: 5ccb883647807f1f3607700846a4ac54@sip.zadarma.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.4.0
Content-Length: 0


---
Audio is at 10336
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 185.45.152.161:5060:
INVITE sip:79165157443@sip.zadarma.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK5a0b7d5c
Max-Forwards: 70
From: "Mans Phone 2 Right" <sip:9997@sip.zadarma.com>;tag=as2e260001
To: <sip:79165157443@sip.zadarma.com>
Contact: <sip:9997@192.168.0.121:5060>
Call-ID: 5ccb883647807f1f3607700846a4ac54@sip.zadarma.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX 13.4.0
Proxy-Authorization: Digest username="150511", realm="sip.zadarma.com", algorithm=MD5, uri="sip:79165157443@sip.zadarma.com", nonce="V9kyZVfZMTkbqHcQSIkdkXXQrfaXGvcr", response="2e3d5d9244cdc2a7b38403deaeafece5", qop=auth, cnonce="13536acd", nc=00000001
Date: Wed, 14 Sep 2016 11:15:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 651708801 651708802 IN IP4 192.168.0.121
s=Asterisk PBX 13.4.0
c=IN IP4 192.168.0.121
t=0 0
m=audio 10336 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:185.45.152.161:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK5a0b7d5c;rport=53718;received=62.140.236.228
From: "Mans Phone 2 Right" <sip:9997@sip.zadarma.com>;tag=as2e260001
To: <sip:79165157443@sip.zadarma.com>;tag=bcc9de9cd72655a99664e3be44d308f3.fa97
Call-ID: 5ccb883647807f1f3607700846a4ac54@sip.zadarma.com
CSeq: 103 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="V9kyZVfZMTkbqHcQSIkdkXXQrfaXGvcr", qop="auth"
Server: Zadarma server
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 185.45.152.161:5060:
ACK sip:79165157443@sip.zadarma.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK5a0b7d5c
Max-Forwards: 70
From: "Mans Phone 2 Right" <sip:9997@sip.zadarma.com>;tag=as2e260001
To: <sip:79165157443@sip.zadarma.com>;tag=bcc9de9cd72655a99664e3be44d308f3.fa97
Contact: <sip:9997@192.168.0.121:5060>
Call-ID: 5ccb883647807f1f3607700846a4ac54@sip.zadarma.com
CSeq: 103 ACK
User-Agent: Asterisk PBX 13.4.0
Content-Length: 0


---
Audio is at 10336
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 185.45.152.161:5060:
INVITE sip:79165157443@sip.zadarma.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK6ee84757
Max-Forwards: 70
From: "Mans Phone 2 Right" <sip:9997@sip.zadarma.com>;tag=as2e260001
To: <sip:79165157443@sip.zadarma.com>
Contact: <sip:9997@192.168.0.121:5060>
Call-ID: 5ccb883647807f1f3607700846a4ac54@sip.zadarma.com
CSeq: 104 INVITE
User-Agent: Asterisk PBX 13.4.0
Proxy-Authorization: Digest username="150511", realm="sip.zadarma.com", algorithm=MD5, uri="sip:79165157443@sip.zadarma.com", nonce="V9kyZVfZMTkbqHcQSIkdkXXQrfaXGvcr", response="33a5850ca1ffd5d63dea390e9e6401b9", qop=auth, cnonce="3069f451", nc=00000002
Date: Wed, 14 Sep 2016 11:15:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 651708801 651708803 IN IP4 192.168.0.121
s=Asterisk PBX 13.4.0
c=IN IP4 192.168.0.121
t=0 0
m=audio 10336 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:185.45.152.161:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK6ee84757;rport=53718;received=62.140.236.228
From: "Mans Phone 2 Right" <sip:9997@sip.zadarma.com>;tag=as2e260001
To: <sip:79165157443@sip.zadarma.com>;tag=bcc9de9cd72655a99664e3be44d308f3.ecca
Call-ID: 5ccb883647807f1f3607700846a4ac54@sip.zadarma.com
CSeq: 104 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="V9kyZVfZMTkbqHcQSIkdkXXQrfaXGvcr", qop="auth"
Server: Zadarma server
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 185.45.152.161:5060:
ACK sip:79165157443@sip.zadarma.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK6ee84757
Max-Forwards: 70
From: "Mans Phone 2 Right" <sip:9997@sip.zadarma.com>;tag=as2e260001
To: <sip:79165157443@sip.zadarma.com>;tag=bcc9de9cd72655a99664e3be44d308f3.ecca
Contact: <sip:9997@192.168.0.121:5060>
Call-ID: 5ccb883647807f1f3607700846a4ac54@sip.zadarma.com
CSeq: 104 ACK
User-Agent: Asterisk PBX 13.4.0
Content-Length: 0


---
Audio is at 10336
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 185.45.152.161:5060:
INVITE sip:79165157443@sip.zadarma.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK5ce42971
Max-Forwards: 70
From: "Mans Phone 2 Right" <sip:9997@sip.zadarma.com>;tag=as2e260001
To: <sip:79165157443@sip.zadarma.com>
Contact: <sip:9997@192.168.0.121:5060>
Call-ID: 5ccb883647807f1f3607700846a4ac54@sip.zadarma.com
CSeq: 105 INVITE
User-Agent: Asterisk PBX 13.4.0
Proxy-Authorization: Digest username="150511", realm="sip.zadarma.com", algorithm=MD5, uri="sip:79165157443@sip.zadarma.com", nonce="V9kyZVfZMTkbqHcQSIkdkXXQrfaXGvcr", response="c62ea835dbe643bf5f6cc0996e517e01", qop=auth, cnonce="384d7279", nc=00000003
Date: Wed, 14 Sep 2016 11:15:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 651708801 651708804 IN IP4 192.168.0.121
s=Asterisk PBX 13.4.0
c=IN IP4 192.168.0.121
t=0 0
m=audio 10336 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:185.45.152.161:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK5ce42971;rport=53718;received=62.140.236.228
From: "Mans Phone 2 Right" <sip:9997@sip.zadarma.com>;tag=as2e260001
To: <sip:79165157443@sip.zadarma.com>;tag=bcc9de9cd72655a99664e3be44d308f3.4e40
Call-ID: 5ccb883647807f1f3607700846a4ac54@sip.zadarma.com
CSeq: 105 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="V9kyZVfZMTkbqHcQSIkdkXXQrfaXGvcr", qop="auth"
Server: Zadarma server
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 185.45.152.161:5060:
ACK sip:79165157443@sip.zadarma.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK5ce42971
Max-Forwards: 70
From: "Mans Phone 2 Right" <sip:9997@sip.zadarma.com>;tag=as2e260001
To: <sip:79165157443@sip.zadarma.com>;tag=bcc9de9cd72655a99664e3be44d308f3.4e40
Contact: <sip:9997@192.168.0.121:5060>
Call-ID: 5ccb883647807f1f3607700846a4ac54@sip.zadarma.com
CSeq: 105 ACK
User-Agent: Asterisk PBX 13.4.0
Content-Length: 0


---
[Sep 14 14:15:05] NOTICE[14403][C-00000012]: chan_sip.c:23283 handle_response_invite: Failed to authenticate on INVITE to '"Mans Phone 2 Right" <sip:9997@sip.zadarma.com>;tag=as2e260001'
    -- SIP/150511-00000021 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Auto fallthrough, channel 'SIP/9997-00000020' status is 'CONGESTION'

<--- Reliably Transmitting (no NAT) to 10.30.1.137:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.30.1.137:5060;branch=z9hG4bK112819761878226383;received=10.30.1.137
From: 9997 <sip:9997@9997>;tag=2667919259
To: "79165157443" <sip:79165157443@9997;user=phone>;tag=as5e9a1cdb
Call-ID: 156071691620432-17914465126364@10.30.1.137
CSeq: 2 INVITE
Server: Asterisk PBX 13.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0


<------------>
Really destroying SIP dialog '5ccb883647807f1f3607700846a4ac54@sip.zadarma.com' Method: INVITE

<--- SIP read from UDP:10.30.1.137:5060 --->
ACK sip:79165157443@9997;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.137:5060;branch=z9hG4bK112819761878226383
From: 9997 <sip:9997@9997>;tag=2667919259
To: "79165157443" <sip:79165157443@9997;user=phone>;tag=as5e9a1cdb
Call-ID: 156071691620432-17914465126364@10.30.1.137
CSeq: 2 ACK
Contact: <sip:9997@10.30.1.137:5060>
Max-Forwards: 70
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '156071691620432-17914465126364@10.30.1.137' Method: ACK
test_pbx*CLI> sip set debug off
SIP Debugging Disabled
test_pbx*CLI>[/details]
9

What happens if you set the fromuser option in the sip.conf entry for your trunk to your username for the trunk?

1 Like
10

Oh my god… It works now! With “fromuser” on trunk sip! Thank you SO MUCH! I spent 2 days per 8-9 hours to configure fresh asterisk with old one settings! And missed this string when I modified trunk sip (because I was thinking it could has changes in new asterisk version). Thank you! Very very much!