Ing. CIP Alejandro Celi Mariategui
2012-May-04 07:11 UTC
[asterisk-users] Broadvoice Got SIP response 503 Service Unavailable
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 at sip.broadvoice.com:XXXXXXXXXX:3054221494 at 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 at 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 at 181.64.96.122:11037>
To: "90018006273999"<sip:90018006273999 at
central.cipher.pe>
From: "488"<sip:488 at 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 at central.cipher.pe>;tag=93cce179
To: "90018006273999"<sip:90018006273999 at
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 at 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 at
central.cipher.pe>;tag=as77d2f824
From: "488"<sip:488 at 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 at 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 at 181.64.96.122:11037>
To: "90018006273999"<sip:90018006273999 at
central.cipher.pe>
From: "488"<sip:488 at 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
at
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 at 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 at central.cipher.pe>;tag=93cce179
To: "90018006273999"<sip:90018006273999 at
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 at 192.168.10.180:5060>
Content-Length: 0
<------------>
-- Executing [90018006273999 at gerencia:1]
Dial("SIP/488-00000000", "SIP/18006273999 at
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 at 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 at
sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999 at sip.broadvoice.com>
Contact: <sip:3054221494 at 192.168.10.180:5060>
Call-ID: 71e46a1e52ecd53c591f47f12589a04c at 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 at sip.broadvoice.com
Retransmitting #1 (no NAT) to 206.15.148.221:5060:
INVITE sip:18006273999 at 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 at
sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999 at sip.broadvoice.com>
Contact: <sip:3054221494 at 192.168.10.180:5060>
Call-ID: 71e46a1e52ecd53c591f47f12589a04c at 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 at sip.broadvoice.com
CSeq: 102 INVITE
From: "Celi M Carbajal" <sip:3054221494 at
sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999 at 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 at sip.broadvoice.com
CSeq: 102 INVITE
From: "Celi M Carbajal" <sip:3054221494 at
sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999 at 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 at 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 at
sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999 at sip.broadvoice.com>;tag=qrst
Contact: <sip:3054221494 at 192.168.10.180:5060>
Call-ID: 71e46a1e52ecd53c591f47f12589a04c at 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 at 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 at central.cipher.pe>;tag=93cce179
To: "90018006273999"<sip:90018006273999 at
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 at 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 at 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 at
central.cipher.pe>;tag=as17386e93
From: "488"<sip:488 at 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 at sip.broadvoice.com
CSeq: 102 INVITE
From: "Celi M Carbajal" <sip:3054221494 at
sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999 at 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) ---
----------------------------------------------------------------
This message was sent using IMP, the Internet Messaging Program.
isrlgb at gmail.com
2012-May-04 09:23 UTC
[asterisk-users] Broadvoice Got SIP response 503 Service Unavailable
Broadvoice has a lot of problems for the last 2 months
-----Original Message-----
From: "Ing. CIP Alejandro Celi Mariategui" <alex at
linux.org.pe>
Sender: asterisk-users-bounces at lists.digium.com
Date: Fri, 04 May 2012 02:11:11
To: <asterisk-users at lists.digium.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Subject: [asterisk-users] Broadvoice Got SIP response 503 Service Unavailable
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 at sip.broadvoice.com:XXXXXXXXXX:3054221494 at 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 at 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 at 181.64.96.122:11037>
To: "90018006273999"<sip:90018006273999 at
central.cipher.pe>
From: "488"<sip:488 at 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 at central.cipher.pe>;tag=93cce179
To: "90018006273999"<sip:90018006273999 at
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 at 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 at
central.cipher.pe>;tag=as77d2f824
From: "488"<sip:488 at 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 at 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 at 181.64.96.122:11037>
To: "90018006273999"<sip:90018006273999 at
central.cipher.pe>
From: "488"<sip:488 at 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
at
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 at 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 at central.cipher.pe>;tag=93cce179
To: "90018006273999"<sip:90018006273999 at
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 at 192.168.10.180:5060>
Content-Length: 0
<------------>
-- Executing [90018006273999 at gerencia:1]
Dial("SIP/488-00000000", "SIP/18006273999 at
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 at 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 at
sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999 at sip.broadvoice.com>
Contact: <sip:3054221494 at 192.168.10.180:5060>
Call-ID: 71e46a1e52ecd53c591f47f12589a04c at 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 at sip.broadvoice.com
Retransmitting #1 (no NAT) to 206.15.148.221:5060:
INVITE sip:18006273999 at 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 at
sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999 at sip.broadvoice.com>
Contact: <sip:3054221494 at 192.168.10.180:5060>
Call-ID: 71e46a1e52ecd53c591f47f12589a04c at 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 at sip.broadvoice.com
CSeq: 102 INVITE
From: "Celi M Carbajal" <sip:3054221494 at
sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999 at 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 at sip.broadvoice.com
CSeq: 102 INVITE
From: "Celi M Carbajal" <sip:3054221494 at
sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999 at 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 at 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 at
sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999 at sip.broadvoice.com>;tag=qrst
Contact: <sip:3054221494 at 192.168.10.180:5060>
Call-ID: 71e46a1e52ecd53c591f47f12589a04c at 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 at 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 at central.cipher.pe>;tag=93cce179
To: "90018006273999"<sip:90018006273999 at
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 at 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 at 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 at
central.cipher.pe>;tag=as17386e93
From: "488"<sip:488 at 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 at sip.broadvoice.com
CSeq: 102 INVITE
From: "Celi M Carbajal" <sip:3054221494 at
sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006273999 at 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) ---
----------------------------------------------------------------
This message was sent using IMP, the Internet Messaging Program.
--
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