Broadvoice Got SIP response 503 Service Unavailable [SOLVED]

Hi,

I’m running Asterisk 1.8.11.1 @office.

The Broadvoice service work fine with all 1.6 version and early 1.8 behind a NAT but about 2 months ago stop working.

No made changes in the firewall NAT rules. Right now I’m @home via my Xlite softphone working fine without problems

Any suggestions or thoughts?

Alex Celi

This is the info

central*CLI> sip show peers
Name/username              Host                                    Dyn Forcerport ACL Port     Status     
488/488                    181.64.96.122                            D                 11037    OK (182 ms) 
sip.broadvoice.com/305422  206.15.148.221                                            5060     OK (131 ms) 

sip.conf

externip=190.12.68.20
localnet=192.168.20.0/255.255.255.0
localnet=192.168.10.0/255.255.255.0
nat=comedia

pedantic=no                     
register => 3054221494@sip.broadvoice.com:XXXXXXXXXX:3054221494@sip.broadvoice.com

[sip.broadvoice.com]
type=friend
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=3054221494             
defaultuser=3054221494
authname=3054221494
secret=XXXXXXXXX
context=entrantes
dtmfmode=inband                 
dtmf=inband
nat=comedia
directmedia=no
qualify=yes
callgroup=1
pickupgroup=1
disallow=all
allow=ulaw
allow=alaw

I turned on sip debug. This is what I received

181.64.96.122: Is my home IP
190.12.68.20 or central.cipher.pe: is office IP
206.15.148.221: Broadvoice Server

<--- SIP read from UDP:181.64.96.122:11037 --->
INVITE sip:90018006273999@central.cipher.pe SIP/2.0
Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:488@181.64.96.122:11037>
To: "90018006273999"<sip:90018006273999@central.cipher.pe>
From: "488"<sip:488@central.cipher.pe>;tag=93cce179
Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1014k stamp 56015
Content-Length: 235

v=0
o=- 8 2 IN IP4 192.168.7.33
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.7.33
t=0 0
m=audio 2424 RTP/AVP 0 8 3 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=alt:1 1 : hC2wRjti 7Lt7EhaI 192.168.7.33 2424
a=sendrecv
<------------->
--- (12 headers 10 lines) ---
Sending to 181.64.96.122:11037 (NAT)
Using INVITE request as basis request - ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
Found peer '488' for '488' from 181.64.96.122:11037

<--- Reliably Transmitting (no NAT) to 181.64.96.122:11037 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;received=181.64.96.122;rport=11037
From: "488"<sip:488@central.cipher.pe>;tag=93cce179
To: "90018006273999"<sip:90018006273999@central.cipher.pe>;tag=as77d2f824
Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.11.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a1fded4"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.' in 11648 ms (Method: INVITE)

<--- SIP read from UDP:181.64.96.122:11037 --->
ACK sip:90018006273999@central.cipher.pe SIP/2.0
Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;rport
To: "90018006273999"<sip:90018006273999@central.cipher.pe>;tag=as77d2f824
From: "488"<sip:488@central.cipher.pe>;tag=93cce179
Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:181.64.96.122:11037 --->
INVITE sip:90018006273999@central.cipher.pe SIP/2.0
Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-a8ee0d381f58006a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:488@181.64.96.122:11037>
To: "90018006273999"<sip:90018006273999@central.cipher.pe>
From: "488"<sip:488@central.cipher.pe>;tag=93cce179
Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1014k stamp 56015
Authorization: Digest username="488",realm="asterisk",nonce="0a1fded4",uri="sip:90018006273999@central.cipher.pe",response="597c1f9bfb78f897ec94139eba9bf061",algorithm=MD5
Content-Length: 235

v=0
o=- 8 2 IN IP4 192.168.7.33
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.7.33
t=0 0
m=audio 2424 RTP/AVP 0 8 3 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=alt:1 1 : hC2wRjti 7Lt7EhaI 192.168.7.33 2424
a=sendrecv
<------------->
--- (13 headers 10 lines) ---
Sending to 181.64.96.122:11037 (no NAT)
Using INVITE request as basis request - ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
Found peer '488' for '488' from 181.64.96.122:11037
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.7.33:2424
Looking for 90018006273999 in gerencia (domain central.cipher.pe)
list_route: hop: <sip:488@181.64.96.122:11037>

<--- Transmitting (no NAT) to 181.64.96.122:11037 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-a8ee0d381f58006a-1---d8754z-;received=181.64.96.122;rport=11037
From: "488"<sip:488@central.cipher.pe>;tag=93cce179
To: "90018006273999"<sip:90018006273999@central.cipher.pe>
Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.11.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:90018006273999@192.168.10.180:5060>
Content-Length: 0


<------------>
    -- Executing [90018006273999@gerencia:1] Dial("SIP/488-00000000", "SIP/18006273999@sip.broadvoice.com,,Tt") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 11220
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 206.15.148.221:5060:
INVITE sip:18006273999@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK47c45d00
Max-Forwards: 70
From: "Celi M Carbajal" <sip:3054221494@sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999@sip.broadvoice.com>
Contact: <sip:3054221494@192.168.10.180:5060>
Call-ID: 71e46a1e52ecd53c591f47f12589a04c@sip.broadvoice.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11.1
Date: Fri, 04 May 2012 06:54:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 209

v=0
o=root 1056464358 1056464358 IN IP4 192.168.10.180
s=Asterisk PBX 1.8.11.1
c=IN IP4 192.168.10.180
t=0 0
m=audio 11220 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
    -- Called SIP/18006273999@sip.broadvoice.com
Retransmitting #1 (no NAT) to 206.15.148.221:5060:
INVITE sip:18006273999@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK47c45d00
Max-Forwards: 70
From: "Celi M Carbajal" <sip:3054221494@sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999@sip.broadvoice.com>
Contact: <sip:3054221494@192.168.10.180:5060>
Call-ID: 71e46a1e52ecd53c591f47f12589a04c@sip.broadvoice.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11.1
Date: Fri, 04 May 2012 06:54:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 209

v=0
o=root 1056464358 1056464358 IN IP4 192.168.10.180
s=Asterisk PBX 1.8.11.1
c=IN IP4 192.168.10.180
t=0 0
m=audio 11220 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:206.15.148.221:5060 --->
SIP/2.0 100 Trying
Call-ID: 71e46a1e52ecd53c591f47f12589a04c@sip.broadvoice.com
CSeq: 102 INVITE
From: "Celi M Carbajal" <sip:3054221494@sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999@sip.broadvoice.com>
Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK47c45d00;received=190.12.68.20;rport=5060
Content-Length: 0

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

<--- SIP read from UDP:206.15.148.221:5060 --->
SIP/2.0 503 Service Unavailable
Call-ID: 71e46a1e52ecd53c591f47f12589a04c@sip.broadvoice.com
CSeq: 102 INVITE
From: "Celi M Carbajal" <sip:3054221494@sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999@sip.broadvoice.com>;tag=qrst
Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK47c45d00;received=190.12.68.20;rport=5060
User-Agent: Asterisk PBX 1.8.11.1
Content-Length: 171
Content-Type: application/sdp

v=0
o=3232238260 1056464358 1056464358 IN IP4 192.168.10.180
s=-
c=IN IP4 192.168.10.180
t=0 0
m=audio 11220 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
<------------->
--- (9 headers 8 lines) ---
    -- Got SIP response 503 "Service Unavailable" back from 206.15.148.221:5060
Transmitting (no NAT) to 206.15.148.221:5060:
ACK sip:18006273999@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK47c45d00
Max-Forwards: 70
From: "Celi M Carbajal" <sip:3054221494@sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999@sip.broadvoice.com>;tag=qrst
Contact: <sip:3054221494@192.168.10.180:5060>
Call-ID: 71e46a1e52ecd53c591f47f12589a04c@sip.broadvoice.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.11.1
Content-Length: 0


---
    -- SIP/sip.broadvoice.com-00000001 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [90018006273999@gerencia:2] Congestion("SIP/488-00000000", "") in new stack

<--- Reliably Transmitting (no NAT) to 181.64.96.122:11037 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-a8ee0d381f58006a-1---d8754z-;received=181.64.96.122;rport=11037
From: "488"<sip:488@central.cipher.pe>;tag=93cce179
To: "90018006273999"<sip:90018006273999@central.cipher.pe>;tag=as17386e93
Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.11.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Circuit/channel congestion
X-Asterisk-HangupCauseCode: 34
Content-Length: 0


<------------>
Really destroying SIP dialog '71e46a1e52ecd53c591f47f12589a04c@sip.broadvoice.com' Method: INVITE
  == Spawn extension (gerencia, 90018006273999, 2) exited non-zero on 'SIP/488-00000000'

<--- SIP read from UDP:181.64.96.122:11037 --->
ACK sip:90018006273999@central.cipher.pe SIP/2.0
Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-a8ee0d381f58006a-1---d8754z-;rport
To: "90018006273999"<sip:90018006273999@central.cipher.pe>;tag=as17386e93
From: "488"<sip:488@central.cipher.pe>;tag=93cce179
Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog 'ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.' Method: ACK

<--- SIP read from UDP:206.15.148.221:5060 --->
SIP/2.0 503 Service Unavailable
Call-ID: 71e46a1e52ecd53c591f47f12589a04c@sip.broadvoice.com
CSeq: 102 INVITE
From: "Celi M Carbajal" <sip:3054221494@sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999@sip.broadvoice.com>;tag=qrst
Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK47c45d00;received=190.12.68.20;rport=5060
User-Agent: Asterisk PBX 1.8.11.1
Content-Length: 171
Content-Type: application/sdp

v=0
o=3232238260 1056464358 1056464358 IN IP4 192.168.10.180
s=-
c=IN IP4 192.168.10.180
t=0 0
m=audio 11220 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
<------------->
--- (9 headers 8 lines) ---

One more thing. When I configure the Broadvoice in a Xlite on the same LAN, works fine.

Alex Celi

I try connecting the Asterisk 1.8.12 (just upgrade it) server direct to the Internet with IP and no luck. Not working with Broadvoice.

But with another Asterisk server with Asterisk 1.6.1.0 in the same internet connection and same BV account, Broadvoice works fine. Then, I try with my laptop with Xlite, and Broadvoice works fine too.

I call to broadvoice support service, and they think that perhaps is an Asterisk bug or not configured option with 1.8.* version

Any idea?

Best regards,

Alex Celi

There is something wrong with your NAT settings because asterisk sends the private IP for media - 192.168.10.180

The solution:

I stop the Asterisk, made a backup of configuration files, delete files and modules previously installed.

Then Reinstall with default config files with “make samples” and I only insert the configuration of Broadvoice and immediately worked. Then I restore the configuration files /etc/asterisk (thinking it was a problem of modules) and stopped working again, It mean that the problem was in the configuration files, or too much or too less lines.

So what I made is transfer line x line what I needed from old to new configuration file.

As soon detected where it was what was missing or left over, I will publish the forum

Alex Celi