Unable to dial local extensions

Hi,

I’ve shifted my server on cloud but unable to dial local extensions.

Getting following error

== Using SIP RTP CoS mark 5
– Called SIP/1340
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [1340@outbound:3] Macro(“SIP/2000-00000037”, “vmtech,1340”) in new stack
– Executing [s@macro-vmtech:1] VoiceMail(“SIP/2000-00000037”, “1340@vmtech”) in new stack
> 0x7fd2456f2d20 – Probation passed - setting RTP source address to 182.176.xxx.xx:23924

Currently running Asterisk 11.25.1

Attaching debug file

SIPdebug.txt (4.5 KB)

Reliably Transmitting (NAT) to 182.176.xxx.xxx:1188:
INVITE sip:1000@182.176.xxx.xxx:1191;transport=udp SIP/2.0



<--- SIP read from UDP:182.176.xxx.xxx:1188 --->
SIP/2.0 404 Not Found

<------------>
    -- Executing [1000@outbound:1] Macro("SIP/2000-00000047", "callrin,1000") in new stack
    -- Executing [s@macro-callrin:1] Set("SIP/2000-00000047", "CALLERID(name)=1000--2000") in new stack
    -- Executing [s@macro-callrin:2] Set("SIP/2000-00000047", "CDR(accountcode)=INBOUND") in new stack
    -- Executing [s@macro-callrin:3] Set("SIP/2000-00000047", "date=2017-08-07") in new stack
    -- Executing [s@macro-callrin:4] Set("SIP/2000-00000047", "time=12-06-31") in new stack
    -- Executing [s@macro-callrin:5] Set("SIP/2000-00000047", "MONITOR_FILENAME=/var/spool/asterisk/monitor/2017-08-07/INBOUND/1000--2000---12-06-31") in new stack
    -- Executing [s@macro-callrin:6] Monitor("SIP/2000-00000047", "wav,/var/spool/asterisk/monitor/2017-08-07/INBOUND/1000--2000---12-06-31,m") in new stack
    -- Executing [1000@outbound:2] Dial("SIP/2000-00000047", "SIP/1000,20,Tt") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 14766
Adding codec 1000008 (g729) to SDP
Adding codec 1000001 (g723) to SDP
Adding codec 1000003 (ulaw) to SDP
Adding codec 1000002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 182.176.xxx.xxx:1188:
INVITE sip:1000@182.176.xxx.xxx:1191;transport=udp SIP/2.0
Via: SIP/2.0/UDP 194.88.xxx.xxx:5060;branch=z9hG4bK012fc7e7;rport
Max-Forwards: 70
From: "1000--2000" <sip:2000@194.88.xxx.xxx>;tag=as51ab6806
To: <sip:1000@182.176.xxx.xxx:1191;transport=udp>
Contact: <sip:2000@194.88.xxx.xxx:5060>
Call-ID: 79eed95a31434bd6233283ae2a9cab5b@194.88.xxx.xxx:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.25.1
Date: Mon, 07 Aug 2017 16:06:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 348

v=0
o=root 175501252 175501252 IN IP4 194.88.xxx.xxx
s=Asterisk PBX 11.25.1
c=IN IP4 194.88.xxx.xxx
t=0 0
m=audio 14766 RTP/AVP 18 4 0 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/1000

<--- SIP read from UDP:182.176.xxx.xxx:1188 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 194.88.xxx.xxx:5060;branch=z9hG4bK012fc7e7;rport
From: "1000--2000" <sip:2000@194.88.xxx.xxx>;tag=as51ab6806
To: <sip:1000@182.176.xxx.xxx:1191;transport=udp>
Call-ID: 79eed95a31434bd6233283ae2a9cab5b@194.88.xxx.xxx:5060
Date: Mon, 07 Aug 2017 14:06:55 GMT
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 182.176.xxx.xxx:1188:
ACK sip:1000@182.176.xxx.xxx:1191;transport=udp SIP/2.0
Via: SIP/2.0/UDP 194.88.xxx.xxx:5060;branch=z9hG4bK012fc7e7;rport
Max-Forwards: 70
From: "1000--2000" <sip:2000@194.88.xxx.xxx>;tag=as51ab6806
To: <sip:1000@182.176.xxx.xxx:1191;transport=udp>
Contact: <sip:2000@194.88.xxx.xxx:5060>
Call-ID: 79eed95a31434bd6233283ae2a9cab5b@194.88.xxx.xxx:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.25.1
Content-Length: 0


---
Scheduling destruction of SIP dialog '79eed95a31434bd6233283ae2a9cab5b@194.88.xxx.xxx:5060' in 16384 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1000@outbound:3] Macro("SIP/2000-00000047", "vmtech,1000") in new stack
    -- Executing [s@macro-vmtech:1] VoiceMail("SIP/2000-00000047", "1000@vmtech") in new stack
Audio is at 19276
Adding codec 1000008 (g729) to SDP
Adding codec 1000003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 182.176.xxx.xxx:29373 --->
SIP/2.0 2000 OK
Via: SIP/2.0/UDP 192.168.10.40:1496;branch=z9hG4bK78c80266;received=182.176.xxx.xxx;rport=29373
From: "2000" <sip:2000@194.88.xxx.xxx>;tag=0015f9dd75dab7591b0b97ed-28d8698b
To: <sip:1000@194.88.xxx.xxx>;tag=as79f50cfd
Call-ID: 0015f9dd-75da008d-36217f27-0059de1b@192.168.11.106
CSeq: 102 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1000@194.88.xxx.xxx:5060>
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 2114304497 2114304497 IN IP4 194.88.xxx.xxx
s=Asterisk PBX 11.25.1
c=IN IP4 194.88.xxx.xxx
t=0 0
m=audio 19276 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Yes, i found that earlier. I can call to extension 2000 from 1000 but not vice versa. settings is exactly same

I’ll recheck, may be I’m missing something

Have checked, configurations are same.

both extensions are registered…

The devices are obviously registered; they can’t return 404 if Asterisk doesn’t know their IP address. The problem is that the device at that address does not like the URI that it is being sent.

On both end I’m using cisco 7960

How can I resolve that

Issue resolved

Still cant figure why it was happening. The DID provider changed the bind port and it start working…