[Asterisk-Users] tryting call problem client sip (ser) to client
sip (asterisk) error
Walter Willis
walterwn at gmail.com
Mon Aug 22 18:06:00 MST 2005
I am tryting call client sip (SER) to client sip (Asterisk) and produce error:
Aug 21 18:48:13 WARNING[13370]: chan_sip.c:695 retrans_pkt: Maximum
retries exceeded on call
9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185 for seqno 18308
(Non-critical Response)
how to fix the problem????
i am configure ser:
if (method=="INVITE") {
if (uri=~"sip:1[1-9][0-9]+ at .*") {
rewritehostport("192.168.0.183:5080");
};
};
an asterisk:
sip.conf
; config Xlite
[1234]
;context=sip
context=from-ser
type=friend
auth=md5
username=1234
secret=chooseapassword
;fromdomain=sorcier.com.pe ; para prueba de ser -asterisk
callerid="First Extension" <1234>
host=dynamic
canreinvite=no
;disallow=all
;allow=gsm
;allow=ulaw
;allow=alaw
;and conexion the ser to asterisk
;
[ser-sip]
type=friend ; permitimos llamadas entrantes y
salientes. Usar peer si solo es MWI
context=ser-asterisk ; este es el contexto que usan las
llamadas entrantes
;host=sorcier.com.pe ; Este es tu hostname o IP del servidor SER
host=192.168.0.183
fromdomain=sorcier.com.pe ; este es tu SER_DOMAIN (nombre de dominio del SER)
;insecure=very ; Permite que las llamadas que viene del
SER pasen a Asterisk
insecure=yes
;mailbox=user at context ; esto es para listar las cuentas de voicemail
;i am copy the voip-info
and the file the extensions.conf
; Configuracion al servidor ser, para llamada de ida
[from-ser]
exten => _X.,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,tr)
[ser-asterisk]
; Ignora el dígito 0
;ignorepat => 0
; conexion a un telefono sip
exten => _1X.,1,Dial(SIP/${EXTEN})
i am probe diferents combinations, but no work
debug with asterisk and view itis:
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2005.08.22 19:35:48 =~=~=~=~=~=~=~=~=~=~=~=
Sip read:
0 headers, 0 lines
Sip read:
INVITE sip:1234 at 192.168.0.183:5080 SIP/2.0
Record-Route: <sip:192.168.0.183;ftag=1168742407;lr=on>
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bKb80c.4bf51016.0
Via: SIP/2.0/UDP
192.168.0.185:5060;rport=5060;branch=z9hG4bK9A2466494C1D4A3A9250CA2F7C44CB67
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=1168742407
To: <sip:1234 at sorcier.com.pe>
Contact: <sip:rbolivar at 192.168.0.185:5060>
Call-ID: 9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185
CSeq: 18308 INVITE
Max-Forwards: 16
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 299
v=0
o=rbolivar 29402538 29402809 IN IP4 192.168.0.185
s=X-Lite
c=IN IP4 192.168.0.185
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
13 headers, 13 lines
Using latest request as basis request
Sending to 192.168.0.183 : 5060 (non-NAT)
Found peer 'ser-sip'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.185:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e
(gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe
(gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Looking for 1234 in ser-asterisk
list_route: hop: <sip:192.168.0.183;ftag=1168742407;lr=on>
list_route: hop: <sip:rbolivar at 192.168.0.185:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bKb80c.4bf51016.0
Via: SIP/2.0/UDP
192.168.0.185:5060;branch=z9hG4bK9A2466494C1D4A3A9250CA2F7C44CB67
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=1168742407
To: <sip:1234 at sorcier.com.pe>
Call-ID: 9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185
CSeq: 18308 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234 at 192.168.0.183:5080>
Content-Length: 0
to 192.168.0.183:5060
Sip read:
INVITE sip:1234 at 192.168.0.183:5080 SIP/2.0
Record-Route: <sip:192.168.0.183;ftag=1168742407;lr=on>
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bKb80c.4bf51016.1
Via: SIP/2.0/UDP
192.168.0.185:5060;rport=5060;branch=z9hG4bK9A2466494C1D4A3A9250CA2F7C44CB67
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=1168742407
To: <sip:1234 at sorcier.com.pe>
Contact: <sip:rbolivar at 192.168.0.185:5060>
Call-ID: 9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185
CSeq: 18308 INVITE
Max-Forwards: 16
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 299
v=0
o=rbolivar 29402538 29402809 IN IP4 192.168.0.185
s=X-Lite
c=IN IP4 192.168.0.185
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
13 headers, 13 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bKb80c.4bf51016.1
Via: SIP/2.0/UDP
192.168.0.185:5060;branch=z9hG4bK9A2466494C1D4A3A9250CA2F7C44CB67
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=1168742407
To: <sip:1234 at sorcier.com.pe>
Call-ID: 9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185
CSeq: 18308 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234 at 192.168.0.183:5080>
Content-Length: 0
to 192.168.0.183:5060
-- Executing Dial("SIP/sorcier.com.pe-081520b8", "SIP/1234") in new stack
We're at 192.168.0.183 port 11872
Answering/Requesting with root capability 0x4 (ulaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:1234 at 192.168.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.183:5080;branch=z9hG4bK76f306dc
From: "rbolivar" <sip:asterisk at 192.168.0.183:5080>;tag=as6a942965
To: <sip:1234 at 192.168.0.182:5060>
Contact: <sip:asterisk at 192.168.0.183:5080>
Call-ID: 619d0bce278dccce3b7280f409c3550f at 192.168.0.183
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sun, 21 Aug 2005 23:48:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 13674 13674 IN IP4 192.168.0.183
s=session
c=IN IP4 192.168.0.183
t=0 0
m=audio 11872 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 192.168.0.182:5060
-- Called 1234
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.183:5080;branch=z9hG4bK76f306dc
From: "rbolivar" <sip:asterisk at 192.168.0.183:5080>;tag=as6a942965
To: <sip:1234 at 192.168.0.182:5060>;tag=4009343049
Contact: <sip:1234 at 192.168.0.182:5060>
Call-ID: 619d0bce278dccce3b7280f409c3550f at 192.168.0.183
CSeq: 102 INVITE
Server: X-Lite release 1103m
Content-Length: 0
9 headers, 0 lines
Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.183:5080;branch=z9hG4bK76f306dc
From: "rbolivar" <sip:asterisk at 192.168.0.183:5080>;tag=as6a942965
To: <sip:1234 at 192.168.0.182:5060>;tag=4009343049
Contact: <sip:1234 at 192.168.0.182:5060>
Call-ID: 619d0bce278dccce3b7280f409c3550f at 192.168.0.183
CSeq: 102 INVITE
Server: X-Lite release 1103m
Content-Length: 0
9 headers, 0 lines
-- SIP/1234-ff3b is ringing
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bKb80c.4bf51016.0
Via: SIP/2.0/UDP
192.168.0.185:5060;branch=z9hG4bK9A2466494C1D4A3A9250CA2F7C44CB67
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=1168742407
To: <sip:1234 at sorcier.com.pe>;tag=as072fce3f
Call-ID: 9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185
CSeq: 18308 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234 at 192.168.0.183:5080>
Content-Length: 0
to 192.168.0.183:5060
Sip read:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.183:5080;branch=z9hG4bK76f306dc
From: "rbolivar" <sip:asterisk at 192.168.0.183:5080>;tag=as6a942965
To: <sip:1234 at 192.168.0.182:5060>;tag=4009343049
Contact: <sip:1234 at 192.168.0.182:5060>
Call-ID: 619d0bce278dccce3b7280f409c3550f at 192.168.0.183
CSeq: 102 INVITE
Content-Type: application/sdp
Server: X-Lite release 1103m
Content-Length: 293
v=0
o=1234 1721485 1728435 IN IP4 192.168.0.182
s=X-Lite
c=IN IP4 192.168.0.182
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
10 headers, 13 lines
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.182:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e
(gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe
(gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
list_route: hop: <sip:1234 at 192.168.0.182:5060>
set_destination: Parsing <sip:1234 at 192.168.0.182:5060> for
address/port to send to
set_destination: set destination to 192.168.0.182, port 5060
Transmitting:
ACK sip:1234 at 192.168.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.183:5080;branch=z9hG4bK7eea7657
From: "rbolivar" <sip:asterisk at 192.168.0.183:5080>;tag=as6a942965
To: <sip:1234 at 192.168.0.182:5060>;tag=4009343049
Contact: <sip:asterisk at 192.168.0.183:5080>
Call-ID: 619d0bce278dccce3b7280f409c3550f at 192.168.0.183
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.168.0.182:5060
-- SIP/1234-ff3b answered SIP/sorcier.com.pe-081520b8
We're at 192.168.0.183 port 17240
Answering with capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bKb80c.4bf51016.0
Via: SIP/2.0/UDP
192.168.0.185:5060;branch=z9hG4bK9A2466494C1D4A3A9250CA2F7C44CB67
Record-Route: <sip:192.168.0.183;ftag=1168742407;lr=on>
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=1168742407
To: <sip:1234 at sorcier.com.pe>;tag=as072fce3f
Call-ID: 9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185
CSeq: 18308 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234 at 192.168.0.183:5080>
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 13674 13674 IN IP4 192.168.0.183
s=session
c=IN IP4 192.168.0.183
t=0 0
m=audio 17240 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 192.168.0.183:5060
-- Attempting native bridge of SIP/sorcier.com.pe-081520b8 and SIP/1234-ff3b
Sip read:
CANCEL sip:1234 at 192.168.0.183:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bKb80c.4bf51016.1
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=1168742407
Call-ID: 9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185
To: <sip:1234 at sorcier.com.pe>
CSeq: 18308 CANCEL
User-Agent: Sip EXpress router(0.10.99-dev14-tcp (i386/linux))
Content-Length: 0
8 headers, 0 lines
Sending to 192.168.0.183 : 5060 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bKb80c.4bf51016.0
Via: SIP/2.0/UDP
192.168.0.185:5060;branch=z9hG4bK9A2466494C1D4A3A9250CA2F7C44CB67
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=1168742407
To: <sip:1234 at sorcier.com.pe>;tag=as072fce3f
Call-ID: 9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185
CSeq: 18308 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234 at 192.168.0.183:5080>
Content-Length: 0
to 192.168.0.183:5060
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bKb80c.4bf51016.1
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=1168742407
To: <sip:1234 at sorcier.com.pe>;tag=as072fce3f
Call-ID: 9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185
CSeq: 18308 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234 at 192.168.0.183:5080>
Content-Length: 0
to 192.168.0.183:5060
set_destination: Parsing <sip:1234 at 192.168.0.182:5060> for
address/port to send to
set_destination: set destination to 192.168.0.182, port 5060
Reliably Transmitting:
BYE sip:1234 at 192.168.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.183:5080;branch=z9hG4bK6d8b8a37
From: "rbolivar" <sip:asterisk at 192.168.0.183:5080>;tag=as6a942965
To: <sip:1234 at 192.168.0.182:5060>;tag=4009343049
Contact: <sip:asterisk at 192.168.0.183:5080>
Call-ID: 619d0bce278dccce3b7280f409c3550f at 192.168.0.183
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.168.0.182:5060
== Spawn extension (ser-asterisk, 1234, 1) exited non-zero on
'SIP/sorcier.com.pe-081520b8'
Sip read:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.183:5080;branch=z9hG4bK6d8b8a37
From: "rbolivar" <sip:asterisk at 192.168.0.183:5080>;tag=as6a942965
To: <sip:1234 at 192.168.0.182:5060>;tag=4009343049
Contact: <sip:1234 at 192.168.0.182:5060>
Call-ID: 619d0bce278dccce3b7280f409c3550f at 192.168.0.183
CSeq: 103 BYE
Server: X-Lite release 1103m
Content-Length: 0
9 headers, 0 lines
Destroying call '619d0bce278dccce3b7280f409c3550f at 192.168.0.183'
Sip read:
ACK sip:1234 at 192.168.0.183:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bKb80c.4bf51016.0
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=1168742407
Call-ID: 9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185
To: <sip:1234 at sorcier.com.pe>;tag=as072fce3f
CSeq: 18308 ACK
User-Agent: Sip EXpress router(0.10.99-dev14-tcp (i386/linux))
Content-Length: 0
8 headers, 0 lines
Sip read:
0 headers, 0 lines
Retransmitting #1 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bKb80c.4bf51016.0
Via: SIP/2.0/UDP
192.168.0.185:5060;branch=z9hG4bK9A2466494C1D4A3A9250CA2F7C44CB67
Record-Route: <sip:192.168.0.183;ftag=1168742407;lr=on>
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=1168742407
To: <sip:1234 at sorcier.com.pe>;tag=as072fce3f
Call-ID: 9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185
CSeq: 18308 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234 at 192.168.0.183:5080>
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 13674 13674 IN IP4 192.168.0.183
s=session
c=IN IP4 192.168.0.183
t=0 0
m=audio 17240 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 192.168.0.183:5060
Retransmitting #2 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bKb80c.4bf51016.0
Via: SIP/2.0/UDP
192.168.0.185:5060;branch=z9hG4bK9A2466494C1D4A3A9250CA2F7C44CB67
Record-Route: <sip:192.168.0.183;ftag=1168742407;lr=on>
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=1168742407
To: <sip:1234 at sorcier.com.pe>;tag=as072fce3f
Call-ID: 9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185
CSeq: 18308 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234 at 192.168.0.183:5080>
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 13674 13674 IN IP4 192.168.0.183
s=session
c=IN IP4 192.168.0.183
t=0 0
m=audio 17240 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 192.168.0.183:5060
Retransmitting #3 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bKb80c.4bf51016.0
Via: SIP/2.0/UDP
192.168.0.185:5060;branch=z9hG4bK9A2466494C1D4A3A9250CA2F7C44CB67
Record-Route: <sip:192.168.0.183;ftag=1168742407;lr=on>
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=1168742407
To: <sip:1234 at sorcier.com.pe>;tag=as072fce3f
Call-ID: 9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185
CSeq: 18308 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234 at 192.168.0.183:5080>
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 13674 13674 IN IP4 192.168.0.183
s=session
c=IN IP4 192.168.0.183
t=0 0
m=audio 17240 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 192.168.0.183:5060
Retransmitting #4 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bKb80c.4bf51016.0
Via: SIP/2.0/UDP
192.168.0.185:5060;branch=z9hG4bK9A2466494C1D4A3A9250CA2F7C44CB67
Record-Route: <sip:192.168.0.183;ftag=1168742407;lr=on>
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=1168742407
To: <sip:1234 at sorcier.com.pe>;tag=as072fce3f
Call-ID: 9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185
CSeq: 18308 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234 at 192.168.0.183:5080>
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 13674 13674 IN IP4 192.168.0.183
s=session
c=IN IP4 192.168.0.183
t=0 0
m=audio 17240 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 192.168.0.183:5060
Retransmitting #5 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bKb80c.4bf51016.0
Via: SIP/2.0/UDP
192.168.0.185:5060;branch=z9hG4bK9A2466494C1D4A3A9250CA2F7C44CB67
Record-Route: <sip:192.168.0.183;ftag=1168742407;lr=on>
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=1168742407
To: <sip:1234 at sorcier.com.pe>;tag=as072fce3f
Call-ID: 9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185
CSeq: 18308 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234 at 192.168.0.183:5080>
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 13674 13674 IN IP4 192.168.0.183
s=session
c=IN IP4 192.168.0.183
t=0 0
m=audio 17240 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 192.168.0.183:5060
Aug 21 18:48:13 WARNING[13370]: chan_sip.c:695 retrans_pkt: Maximum
retries exceeded on call
9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185 for seqno 18308
(Non-critical Response)
Destroying call '9B82BB18-0F51-4665-A1FE-F0D88D3D0938 at 192.168.0.185'
the client sip (SER) call to client sip (asterisk) and return error 404
WHAT IS THE PROBLEM??? OR HOW TO FIX THE ERROR??
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