Broadvoice responde Got SIP response 503 Service Unavailable
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Ing CIP. Alejandro Celi
May 7, 2012, 4:53:36 PM5/7/12
to aster...@googlegroups.com
Hola,
Estoy con un Asterisk 1.8.12 en mi oficina
El servicio de Broadvoice funciono bien con todos los Asterisk 1.6 y algunos de la 1.8, estando detras de un firewall/NAT, pero hace unos meses dejo de funcionar
No se hicieron cambios en las reglas del firewall ni NAT. Las pruebas las hice desde mi casa por medio de un Softphone Xlite, el mismo que trabaja sin problemas
En medio de la desesperacion, le puse un IP publico al servidor para obviar el firewall y no funciona. Efectue las pruebas con la misma cuenta y la misma coneccion de internet, pero esta vez con un Asterisk 1.6.1 y funciono sin problemas. Asi mismo hice las pruebas desde mi laptop configurando la cuenta en un Xlite y tambien funcion sin problemas.
Llame al soporte de Broadvoice y lo que me dijeron es que podria ser un bug de Asterisk, o hay algo que no he configurado con las version 1.8.*.
Alguna idea o ayuda?
Aqui esta la informacion
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 =>
30542...@sip.broadvoice.com:XXXXXXXXXX:30542...@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
Encendi el sip debug.y estos son los resultados
181.64.96.122: El IP desde mi casa
190.12.68.20 or central.cipher.pe: El IP del servidor en la oficina
206.15.148.221: Broadvoice Server
<--- SIP read from UDP:181.64.96.122:11037 --->
INVITE sip:9001800...@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:4...@181.64.96.122:11037>
To: "90018006273999"<sip:9001800...@central.cipher.pe>
From: "488"<sip:4...@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:4...@central.cipher.pe>;tag=93cce179
To: "90018006273999"<sip:9001800...@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:9001800...@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:9001800...@central.cipher.pe>;tag=as77d2f824
From: "488"<sip:4...@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:9001800...@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:4...@181.64.96.122:11037>
To: "90018006273999"<sip:9001800...@central.cipher.pe>
From: "488"<sip:4...@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:9001800...@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:4...@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:4...@central.cipher.pe>;tag=93cce179
To: "90018006273999"<sip:9001800...@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:9001800...@192.168.10.180:5060>
Content-Length: 0
<------------>
-- Executing [90018006273999@gerencia:1]
Dial("SIP/488-00000000", "SIP/18006...@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:18006...@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:30542...@sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006...@sip.broadvoice.com>
Contact: <sip:30542...@192.168.10.180:5060>
Call-ID: 71e46a1e52ecd53c...@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/18006...@sip.broadvoice.com
Retransmitting #1 (no NAT) to 206.15.148.221:5060:
INVITE sip:18006...@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:30542...@sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006...@sip.broadvoice.com>
Contact: <sip:30542...@192.168.10.180:5060>
Call-ID: 71e46a1e52ecd53c...@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: 71e46a1e52ecd53c...@sip.broadvoice.com
CSeq: 102 INVITE
From: "Celi M Carbajal" <sip:30542...@sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006...@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: 71e46a1e52ecd53c...@sip.broadvoice.com
CSeq: 102 INVITE
From: "Celi M Carbajal" <sip:30542...@sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006...@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:18006...@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:30542...@sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006...@sip.broadvoice.com>;tag=qrst
Contact: <sip:30542...@192.168.10.180:5060>
Call-ID: 71e46a1e52ecd53c...@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:4...@central.cipher.pe>;tag=93cce179
To: "90018006273999"<sip:9001800...@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
'71e46a1e52ecd53c...@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:9001800...@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:9001800...@central.cipher.pe>;tag=as17386e93
From: "488"<sip:4...@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: 71e46a1e52ecd53c...@sip.broadvoice.com
CSeq: 102 INVITE
From: "Celi M Carbajal" <sip:30542...@sip.broadvoice.com>;tag=as18a86be7
To: <sip:18006...@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) ---
Saludos,
Alex Celi
|
Fernando Villares
May 7, 2012, 5:45:43 PM5/7/12
to aster...@googlegroups.com
probaste con peer??? recorda que un friend matchea por username mientras que un peer por la IP!!!!
2012/5/7 Ing CIP. Alejandro Celi <al...@linux.org.pe>
--
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ricardo vargas
May 8, 2012, 11:03:53 AM5/8/12
to asterisk-es
Yo he instalado a varios de mis clientes troncales de broadvoice y a
la fecha he tenido muchos problemas con ellos, te recomiendo no seguir
con este carrier, en cuanto el problema que posees intenta colocar en
el /etc/hosts la ip del servidor sip de atlanta, yo lo tengo así y
pues me ha funcionado.
127.0.0.1 localhost
127.0.1.1 debian
206.15.156.221 sip.broadvoice.com
On 7 mayo, 16:45, Fernando Villares <fvilla...@gmail.com> wrote:
> probaste con peer??? recorda que un friend matchea por username mientras
> que un peer por la IP!!!!
>
> 2012/5/7 Ing CIP. Alejandro Celi <a...@linux.org.pe>
>
>
>
>
>
>
>
> > **
> > 3054221...@sip.broadvoice.com:XXXXXXXXXX:3054221...@sip.broadvoice.com
> > To: "90018006273999"<sip:90018006273...@central.cipher.pe
> > To: "90018006273999"<sip:90018006273...@central.cipher.pe>
> > >;tag=as18a86be7
> > To: <sip:18006273...@sip.broadvoice.com>
> > Contact: <sip:3054221...@192.168.10.180:5060>
> > Call-ID: 71e46a1e52ecd53c591f47f12589a...@sip.broadvoice.com
> > >;tag=as18a86be7
> > To: <sip:18006273...@sip.broadvoice.com>
> > Contact: <sip:3054221...@192.168.10.180:5060>
> > Call-ID: 71e46a1e52ecd53c591f47f12589a...@sip.broadvoice.com
>
> ...
>
> leer más »
la fecha he tenido muchos problemas con ellos, te recomiendo no seguir
con este carrier, en cuanto el problema que posees intenta colocar en
el /etc/hosts la ip del servidor sip de atlanta, yo lo tengo así y
pues me ha funcionado.
127.0.0.1 localhost
127.0.1.1 debian
206.15.156.221 sip.broadvoice.com
On 7 mayo, 16:45, Fernando Villares <fvilla...@gmail.com> wrote:
> probaste con peer??? recorda que un friend matchea por username mientras
> que un peer por la IP!!!!
>
>
>
>
>
>
>
>
> > **
>
> > [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
>
> > Encendi el sip debug.y estos son los resultados
>
> > 181.64.96.122: El IP desde mi casa
> > 190.12.68.20 or central.cipher.pe: El IP del servidor en la oficina
> > 206.15.148.221: Broadvoice Server
>
> > <--- SIP read from UDP:181.64.96.122:11037 --->
> > INVITE sip:90018006273...@central.cipher.pe SIP/2.0
> > [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
>
> > Encendi el sip debug.y estos son los resultados
>
> > 181.64.96.122: El IP desde mi casa
> > 190.12.68.20 or central.cipher.pe: El IP del servidor en la oficina
> > 206.15.148.221: Broadvoice Server
>
> > <--- SIP read from UDP:181.64.96.122:11037 --->
> > Via: SIP/2.0/UDP
> > 192.168.7.33:19116
> > ;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;rport
> > Max-Forwards: 70
> > Contact: <sip:4...@181.64.96.122:11037>
> > To: "90018006273999"<sip:90018006273...@central.cipher.pe>
> > 192.168.7.33:19116
> > ;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;rport
> > Max-Forwards: 70
> > Contact: <sip:4...@181.64.96.122:11037>
> > >;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:90018006273...@central.cipher.pe SIP/2.0
> > 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 --->
> > Via: SIP/2.0/UDP
> > 192.168.7.33:19116
> > ;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;rport
> > To: "90018006273999"<sip:90018006273...@central.cipher.pe
> > 192.168.7.33:19116
> > ;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;rport
> > >;tag=as77d2f824
> > From: "488"<sip:4...@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:90018006273...@central.cipher.pe SIP/2.0
> > From: "488"<sip:4...@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 --->
> > Via: SIP/2.0/UDP
> > 192.168.7.33:19116
> > ;branch=z9hG4bK-d8754z-a8ee0d381f58006a-1---d8754z-;rport
> > Max-Forwards: 70
> > Contact: <sip:4...@181.64.96.122:11037>
> > To: "90018006273999"<sip:90018006273...@central.cipher.pe>
> > 192.168.7.33:19116
> > ;branch=z9hG4bK-d8754z-a8ee0d381f58006a-1---d8754z-;rport
> > Max-Forwards: 70
> > Contact: <sip:4...@181.64.96.122:11037>
> > From: "488"<sip:4...@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:90018006273...@central.cipher.pe
> > 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="
> > 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:90018006273...@192.168.10.180:5060>
> > CSeq: 2 INVITE
> > Server: Asterisk PBX 1.8.11.1
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> > NOTIFY, INFO, PUBLISH
> > Supported: replaces, timer
> > Content-Length: 0
>
> > <------------>
> > -- Executing [90018006273999@gerencia:1]
> > Dial("SIP/488-00000000", "SIP/18006273...@sip.broadvoice.com,,Tt") in
>
> > <------------>
> > -- Executing [90018006273999@gerencia:1]
> > 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:18006273...@sip.broadvoice.com SIP/2.0
> > == 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:
> > Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK47c45d00
> > Max-Forwards: 70
> > From: "Celi M Carbajal" <sip:3054221...@sip.broadvoice.com
> > Max-Forwards: 70
> > >;tag=as18a86be7
> > To: <sip:18006273...@sip.broadvoice.com>
> > Contact: <sip:3054221...@192.168.10.180:5060>
> > Call-ID: 71e46a1e52ecd53c591f47f12589a...@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/18006273...@sip.broadvoice.com
> > 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
>
> > ---
> > Retransmitting #1 (no NAT) to 206.15.148.221:5060:
> > INVITE sip:18006273...@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:3054221...@sip.broadvoice.com
> > Max-Forwards: 70
> > >;tag=as18a86be7
> > To: <sip:18006273...@sip.broadvoice.com>
> > Contact: <sip:3054221...@192.168.10.180:5060>
> > Call-ID: 71e46a1e52ecd53c591f47f12589a...@sip.broadvoice.com
>
> ...
>
> leer más »
Ing CIP. Alejandro Celi
May 9, 2012, 6:10:09 PM5/9/12
to aster...@googlegroups.com
La solucion:
Detuve el Asterisk, saque un backup de los archivos de configuracion, borre los modulos y archivos instalados previamente.
Reinstale, agregandole con el "make samples" y solo le puse la configuracion del Broadvoice y de inmediato funciono. De ahi restaure los archivos de configuracion de /et/asterisk (pensando que era un problema de modulos) y de nuevo dejo de funcionar, osea que el problema estaba en los archivos de configuracion, o le falta o le sobra algo.
Entonces lo que hice fue ir pasando solo las lineas que necesitaba, archivo por archivo de configuracion.
Ni bien detecte donde estaba lo que faltaba o sobraba, lo publico en la lista
Saludos,
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