Arslan Saeed
2003-Nov-15 13:11 UTC
[Asterisk-Users] Internal server error - cannot align media streams - help needed
Hi,
I configured asterisk on redhat linux 9 box. I installed two different
ip softphones (SJPHONE and X-PRO) and got them registered with asterisk.
The call from one phone to another does get routed via asterisk, but
there is one problem coming up. As soon as call is accepted by the end
user , it is automatically disconnected with the error "cannot align
media streams". If I enable SIP debugging on asterisk, then I find the
following output
"-- Got SIP response 500 "Internal server error (cannot align media
streams)" back from 197.7.75.129"
followed by the following debug message
(no NAT) to 197.7.75.129:5060
-- SIP/2001-a513 is circuit-busy
== Everyone is busy at this time
We're at 197.7.75.85 port 16816
Answering with preferred capability 2147483647
Answering with non-codec capability 1
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Below is the configuration of asterisk
SIP.CONF
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about here
[2000]
type=soft1 ; This device takes and makes calls
username=2000 ; Username on device
secret=friend ; Password for device
host=dynamic ; This host is not on the same IP addr every time
context=from-sip ; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
; voicemailbox has messages in it
[2001] ; Duplicate of 2000, except with different auth
data
type=soft2
username=2001
secret=friend
host=dynamic
context=from-sip
mailbox=101
EXTENSIONS.CONF
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
[bogon-calls]
[from-sip]
exten => 2000,1,Dial(SIP/2000,20)
exten => 2000,2,Voicemail(u2000)
exten => 2000,102,Voicemail(b2000)
exten => 2000,103,Hangup
exten => 2001,1,Dial(SIP/2001,20)
exten => 2001,2,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangup
exten => 2999,1,VoicemailMain(${CALLERIDNUM})
Thanls
Arslan,
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TeleSIP
2003-Nov-15 14:04 UTC
[Asterisk-Users] Internal server error - cannot align media streams - help needed
Try this:
In your [general] section:
disallow=all
allow=ulaw
allow=alaw
this forces * to only accept ulaw and alaw codecs.
----- Original Message -----
From: Arslan Saeed
To: asterisk-users@lists.digium.com
Sent: Saturday, November 15, 2003 3:11 PM
Subject: [Asterisk-Users] Internal server error - cannot align media
streams - help needed
Hi,
I configured asterisk on redhat linux 9 box. I installed two different ip
softphones (SJPHONE and X-PRO) and got them registered with asterisk. The
call from one phone to another does get routed via asterisk, but there is
one problem coming up. As soon as call is accepted by the end user , it is
automatically disconnected with the error "cannot align media
streams". If I
enable SIP debugging on asterisk, then I find the following output
"-- Got SIP response 500 "Internal server error (cannot align media
streams)" back from 197.7.75.129"
followed by the following debug message
(no NAT) to 197.7.75.129:5060
-- SIP/2001-a513 is circuit-busy
== Everyone is busy at this time
We're at 197.7.75.85 port 16816
Answering with preferred capability 2147483647
Answering with non-codec capability 1
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Below is the configuration of asterisk
SIP.CONF
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about here
[2000]
type=soft1 ; This device takes and makes calls
username=2000 ; Username on device
secret=friend ; Password for device
host=dynamic ; This host is not on the same IP addr every time
context=from-sip ; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
; voicemailbox has messages in it
[2001] ; Duplicate of 2000, except with different auth data
type=soft2
username=2001
secret=friend
host=dynamic
context=from-sip
mailbox=101
EXTENSIONS.CONF
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
[bogon-calls]
[from-sip]
exten => 2000,1,Dial(SIP/2000,20)
exten => 2000,2,Voicemail(u2000)
exten => 2000,102,Voicemail(b2000)
exten => 2000,103,Hangup
exten => 2001,1,Dial(SIP/2001,20)
exten => 2001,2,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangup
exten => 2999,1,VoicemailMain(${CALLERIDNUM})
Thanls
Arslan,
Arslan Saeed
2003-Nov-15 16:32 UTC
[Asterisk-Users] Internal server error - cannot align media streams - help needed
Thanks,
Problem solved when explicitly configured to allow only g711 codec.
Arslan.
-----Original Message-----
From: TeleSIP [mailto:ricvil@telesip.net]
Sent: Sunday, November 16, 2003 2:05 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Internal server error - cannot align media
streams - help needed
Try this:
In your [general] section:
disallow=all
allow=ulaw
allow=alaw
this forces * to only accept ulaw and alaw codecs.
----- Original Message -----
From: Arslan Saeed
To: asterisk-users@lists.digium.com
Sent: Saturday, November 15, 2003 3:11 PM
Subject: [Asterisk-Users] Internal server error - cannot align media
streams - help needed
Hi,
I configured asterisk on redhat linux 9 box. I installed two different
ip
softphones (SJPHONE and X-PRO) and got them registered with asterisk.
The
call from one phone to another does get routed via asterisk, but there
is
one problem coming up. As soon as call is accepted by the end user , it
is
automatically disconnected with the error "cannot align media
streams".
If I
enable SIP debugging on asterisk, then I find the following output
"-- Got SIP response 500 "Internal server error (cannot align media
streams)" back from 197.7.75.129"
followed by the following debug message
(no NAT) to 197.7.75.129:5060
-- SIP/2001-a513 is circuit-busy
== Everyone is busy at this time
We're at 197.7.75.85 port 16816
Answering with preferred capability 2147483647
Answering with non-codec capability 1
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Below is the configuration of asterisk
SIP.CONF
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about here
[2000]
type=soft1 ; This device takes and makes calls
username=2000 ; Username on device
secret=friend ; Password for device
host=dynamic ; This host is not on the same IP addr every time
context=from-sip ; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
; voicemailbox has messages in it
[2001] ; Duplicate of 2000, except with different auth
data
type=soft2
username=2001
secret=friend
host=dynamic
context=from-sip
mailbox=101
EXTENSIONS.CONF
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
[bogon-calls]
[from-sip]
exten => 2000,1,Dial(SIP/2000,20)
exten => 2000,2,Voicemail(u2000)
exten => 2000,102,Voicemail(b2000)
exten => 2000,103,Hangup
exten => 2001,1,Dial(SIP/2001,20)
exten => 2001,2,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangup
exten => 2999,1,VoicemailMain(${CALLERIDNUM})
Thanls
Arslan,
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