Re: Broadvoice Got SIP response 503 Service Unavailable

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Broadvoice has a lot of problems for the last 2 months 

-----Original Message-----
From: "Ing. CIP Alejandro Celi Mariategui" <alex@xxxxxxxxxxxx>
Sender: asterisk-users-bounces@xxxxxxxxxxxxxxxx
Date: Fri, 04 May 2012 02:11:11 
To: <asterisk-users@xxxxxxxxxxxxxxxx>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users@xxxxxxxxxxxxxxxx>
Subject:  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@xxxxxxxxxxxxxxxxxx:XXXXXXXXXX:3054221494@xxxxxxxxxxxxxxxxxx

     [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@xxxxxxxxxxxxxxxxx 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@xxxxxxxxxxxxxxxxx>
     From: "488"<sip:488@xxxxxxxxxxxxxxxxx>;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@xxxxxxxxxxxxxxxxx>;tag=93cce179
     To: "90018006273999"<sip:90018006273999@xxxxxxxxxxxxxxxxx>;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@xxxxxxxxxxxxxxxxx SIP/2.0
     Via: SIP/2.0/UDP  
192.168.7.33:19116;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;rport
     To: "90018006273999"<sip:90018006273999@xxxxxxxxxxxxxxxxx>;tag=as77d2f824
     From: "488"<sip:488@xxxxxxxxxxxxxxxxx>;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@xxxxxxxxxxxxxxxxx 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@xxxxxxxxxxxxxxxxx>
     From: "488"<sip:488@xxxxxxxxxxxxxxxxx>;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@xxxxxxxxxxxxxxxxx",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@xxxxxxxxxxxxxxxxx>;tag=93cce179
     To: "90018006273999"<sip:90018006273999@xxxxxxxxxxxxxxxxx>
     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@xxxxxxxxxxxxxxxxxx,,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@xxxxxxxxxxxxxxxxxx SIP/2.0
     Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK47c45d00
     Max-Forwards: 70
     From: "Celi M Carbajal" <sip:3054221494@xxxxxxxxxxxxxxxxxx>;tag=as18a86be7
     To: <sip:18006273999@xxxxxxxxxxxxxxxxxx>
     Contact: <sip:3054221494@192.168.10.180:5060>
     Call-ID: 71e46a1e52ecd53c591f47f12589a04c@xxxxxxxxxxxxxxxxxx
     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@xxxxxxxxxxxxxxxxxx
     Retransmitting #1 (no NAT) to 206.15.148.221:5060:
     INVITE sip:18006273999@xxxxxxxxxxxxxxxxxx SIP/2.0
     Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK47c45d00
     Max-Forwards: 70
     From: "Celi M Carbajal" <sip:3054221494@xxxxxxxxxxxxxxxxxx>;tag=as18a86be7
     To: <sip:18006273999@xxxxxxxxxxxxxxxxxx>
     Contact: <sip:3054221494@192.168.10.180:5060>
     Call-ID: 71e46a1e52ecd53c591f47f12589a04c@xxxxxxxxxxxxxxxxxx
     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@xxxxxxxxxxxxxxxxxx
     CSeq: 102 INVITE
     From: "Celi M Carbajal" <sip:3054221494@xxxxxxxxxxxxxxxxxx>;tag=as18a86be7
     To: <sip:18006273999@xxxxxxxxxxxxxxxxxx>
     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@xxxxxxxxxxxxxxxxxx
     CSeq: 102 INVITE
     From: "Celi M Carbajal" <sip:3054221494@xxxxxxxxxxxxxxxxxx>;tag=as18a86be7
     To: <sip:18006273999@xxxxxxxxxxxxxxxxxx>;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@xxxxxxxxxxxxxxxxxx SIP/2.0
     Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK47c45d00
     Max-Forwards: 70
     From: "Celi M Carbajal" <sip:3054221494@xxxxxxxxxxxxxxxxxx>;tag=as18a86be7
     To: <sip:18006273999@xxxxxxxxxxxxxxxxxx>;tag=qrst
     Contact: <sip:3054221494@192.168.10.180:5060>
     Call-ID: 71e46a1e52ecd53c591f47f12589a04c@xxxxxxxxxxxxxxxxxx
     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@xxxxxxxxxxxxxxxxx>;tag=93cce179
     To: "90018006273999"<sip:90018006273999@xxxxxxxxxxxxxxxxx>;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@xxxxxxxxxxxxxxxxxx' 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@xxxxxxxxxxxxxxxxx SIP/2.0
     Via: SIP/2.0/UDP  
192.168.7.33:19116;branch=z9hG4bK-d8754z-a8ee0d381f58006a-1---d8754z-;rport
     To: "90018006273999"<sip:90018006273999@xxxxxxxxxxxxxxxxx>;tag=as17386e93
     From: "488"<sip:488@xxxxxxxxxxxxxxxxx>;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@xxxxxxxxxxxxxxxxxx
     CSeq: 102 INVITE
     From: "Celi M Carbajal" <sip:3054221494@xxxxxxxxxxxxxxxxxx>;tag=as18a86be7
     To: <sip:18006273999@xxxxxxxxxxxxxxxxxx>;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) ---




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