Re: [asterisk-users] Error 488 Not Acceptable Here
Hi, Am Donnerstag, den 23.05.2013, 20:48 +0200 schrieb Maximilian Grobecker: Am 22.05.2013 16:39, schrieb Andrew Colin: Hi guys, Any idea why I am getting this error when someone tries to send me a T38 Fax? Hi, Maybe you have not allowed T.38 as acceptable codec ;-) You can try with allow=all in your sip.conf. No, T.38 is not a codec and so allow=all will not help. To use T.38 You have to enable T.38 with t38pt_udptl = yes in sip.conf. The reason, why You get a 488 Not Acceptable Here488 Not Acceptable Here, is only detectable with a SIP Trace. There are many reasons e.g. - Your asterisk version does not support T.38 - T.38 is not enabled (see above) - T.38 is enabled, but not at the relevant peers (in most versions of asterisk there is only support for T.38 passthrough, so both peers have to support T.38) - There are problems in the transmission and some peers wants to switch back to audio level and the other or asterisk is not willing to support this. The last reason may occur, if You have NAT and do not correctly forward the data ports of T.38 (UDPTL Ports). Best way is to get a SIP Trace to analyse. If You provide one, You should also tell, which version of asterisk. HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error 488 Not Acceptable Here
Hi, Maybe you have not allowed T.38 as acceptable codec ;-) You can try with allow=all in your sip.conf. Am 22.05.2013 16:39, schrieb Andrew Colin: Hi guys, Any idea why I am getting this error when someone tries to send me a T38 Fax? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- - Portunity GmbH - Werner-Seelenbinder-Str. 23 -- 42477 Radevormwald - Germany - - Portal: http://www.portunity.de - - General: Phone: +49 (0)202 - 69555 - 0 - eMail/SIP: i...@portunity.de - Fax: +49 (0)202 - 69555 - 190 - - Support: Phone: +49 (0)202 - 69555 - 300 - eMail/SIP: supp...@portunity.de - - Amtsgericht Koeln HRB 38162 - USt-Identnummer DE206277867 - Geschaeftsfuehrung: Bjoern Ruecker, Bernd Schnell -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error 488 Not Acceptable Here
488 not acceptable is due to codec error. Make sure you have right codec in place between the end points. On Fri, May 24, 2013 at 12:18 AM, Maximilian Grobecker m.grobec...@portunity.de wrote: Hi, Maybe you have not allowed T.38 as acceptable codec ;-) You can try with allow=all in your sip.conf. Am 22.05.2013 16:39, schrieb Andrew Colin: Hi guys, Any idea why I am getting this error when someone tries to send me a T38 Fax? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- - Portunity GmbH - Werner-Seelenbinder-Str. 23 -- 42477 Radevormwald - Germany - - Portal: http://www.portunity.de - - General: Phone: +49 (0)202 - 69555 - 0 - eMail/SIP: i...@portunity.de - Fax: +49 (0)202 - 69555 - 190 - - Support: Phone: +49 (0)202 - 69555 - 300 - eMail/SIP: supp...@portunity.de - - Amtsgericht Koeln HRB 38162 - USt-Identnummer DE206277867 - Geschaeftsfuehrung: Bjoern Ruecker, Bernd Schnell -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error 488 Not Acceptable Here
Hi guys, Any idea why I am getting this error when someone tries to send me a T38 Fax? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error 488 - Not Acceptable Here
I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone
are registered by the below information
*CLI sip show peers
Name/usernameHost Mask Port Status
2002/2002192.168.22.199 (D) 255.255.255.255 5060 Unmonitored
2001/2001192.168.22.200 (D) 255.255.255.255 5060 Unmonitored
2000/2000192.168.22.198 (D) 255.255.255.255 5060 Unmonitored
*CLI sip show users
Username Secret Authen Def.Context A/C
2002 ciscomd5,plaintextdemo No
2001 ciscomd5,plaintextdemo No
2000 ciscomd5,plaintextdemo No
All 3 phones and the asterisk box are on the 192.168.22.0/24 subnet.
I've attached my sip.conf and extensions.conf file for review...
When I start the server and a phone dials another phone I get the below
answer.
*CLI -- Executing Dial(SIP/2001-0bb5, SIP/2002|30|tr) in new stack
-- Called 2002
-- Got SIP response 488 Not Acceptable Here back from 192.168.22.199
== No one is available to answer at this time
-- Timeout on SIP/2001-0bb5
I *believe* the sip response might be from the phone itself - and not a
asterisk misconfig. I'm just wanting a second pair of eyes.
I put in
canreinvite=no
for each phone profile as people have said this is needed for buggy
Cisco phones.
;
; SIP Configuration for Asterisk
;
; Syntax for specifying a SIP device in extensions.conf is
; SIP/devicename where devicename is defined in a section below.
;
; You may also use
; SIP/[EMAIL PROTECTED] to call any SIP user on the Internet
; (Don't forget to enable DNS SRV records if you want to use this)
;
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED]
; where the proxyhostname is defined in a section below
;
; Useful CLI commands to check peers/users:
; sip show peers Show all SIP peers (including friends)
; sip show users Show all SIP users (including friends)
; sip show registry Show status of hosts we register with
;
; sip debug Show all SIP messages
;
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
;bindaddr = 192.168.22.254; 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
tos = lowdelay; can be lowdelay, throughput, reliability, mincost
[2000]
type=friend ; This device takes and makes calls
username=2000 ; Username on device
secret=cisco; Password for device
;host=192.168.22.1 ; This host is not on the same IP addr every time
host=dynamic
context=demo; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
canreinvite=no
; voicemailbox has messages in it
[2001]; Duplicate of 2000, except with different auth data
type=friend
username=2001
secret=cisco
host=dynamic
;host=192.168.22.2
context=demo
mailbox=101
canreinvite=no
[2002]; Duplicate of 2000, except with different auth data
type=friend
username=2002
secret=cisco
;host=192.168.22.3
host=dynamic
context=demo
mailbox=102
canreinvite=no
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
[incoming]
exten = s,1,Echo ;for testing the connection
;exten = s,1,Playback,demo-thanks ;for playing a file
;
; The General category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
; You can include other config files, use the #include command (without the ';')
; Note that this is different from the include command that includes contexts within
; other contexts. The #include command works in all asterisk configuration files.
;#include filename.conf
; The Globals category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
Re: [Asterisk-Users] error 488 - Not Acceptable Here
Take out the allow=all in your sip.conf and put in allow= for the codec
you want to use and disallow=all.
On Wed, 2004-04-07 at 15:18, Roger wrote:
I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone
are registered by the below information
*CLI sip show peers
Name/usernameHost Mask Port Status
2002/2002192.168.22.199 (D) 255.255.255.255 5060 Unmonitored
2001/2001192.168.22.200 (D) 255.255.255.255 5060 Unmonitored
2000/2000192.168.22.198 (D) 255.255.255.255 5060 Unmonitored
*CLI sip show users
Username Secret Authen Def.Context A/C
2002 ciscomd5,plaintextdemo No
2001 ciscomd5,plaintextdemo No
2000 ciscomd5,plaintextdemo No
All 3 phones and the asterisk box are on the 192.168.22.0/24 subnet.
I've attached my sip.conf and extensions.conf file for review...
When I start the server and a phone dials another phone I get the below
answer.
*CLI -- Executing Dial(SIP/2001-0bb5, SIP/2002|30|tr) in new stack
-- Called 2002
-- Got SIP response 488 Not Acceptable Here back from 192.168.22.199
== No one is available to answer at this time
-- Timeout on SIP/2001-0bb5
I *believe* the sip response might be from the phone itself - and not a
asterisk misconfig. I'm just wanting a second pair of eyes.
I put in
canreinvite=no
for each phone profile as people have said this is needed for buggy
Cisco phones.
__
;
; SIP Configuration for Asterisk
;
; Syntax for specifying a SIP device in extensions.conf is
; SIP/devicename where devicename is defined in a section below.
;
; You may also use
; SIP/[EMAIL PROTECTED] to call any SIP user on the Internet
; (Don't forget to enable DNS SRV records if you want to use this)
;
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED]
; where the proxyhostname is defined in a section below
;
; Useful CLI commands to check peers/users:
; sip show peers Show all SIP peers (including friends)
; sip show users Show all SIP users (including friends)
; sip show registry Show status of hosts we register with
;
; sip debug Show all SIP messages
;
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
;bindaddr = 192.168.22.254; 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
tos = lowdelay ; can be lowdelay, throughput, reliability, mincost
[2000]
type=friend ; This device takes and makes calls
username=2000 ; Username on device
secret=cisco ; Password for device
;host=192.168.22.1 ; This host is not on the same IP addr every time
host=dynamic
context=demo ; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
canreinvite=no
; voicemailbox has messages in it
[2001]; Duplicate of 2000, except with different auth data
type=friend
username=2001
secret=cisco
host=dynamic
;host=192.168.22.2
context=demo
mailbox=101
canreinvite=no
[2002]; Duplicate of 2000, except with different auth data
type=friend
username=2002
secret=cisco
;host=192.168.22.3
host=dynamic
context=demo
mailbox=102
canreinvite=no
__
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
[incoming]
exten = s,1,Echo ;for testing the connection
;exten = s,1,Playback,demo-thanks ;for playing a file
;
; The General category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
; You can include other config files, use the #include command (without the ';')
; Note that this is different from the include command that includes contexts
within
; other contexts. The #include command works in all asterisk configuration files.
;#include filename.conf
; The Globals category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
Re: [Asterisk-Users] error 488 - Not Acceptable Here
Eric Wieling wrote: Take out the allow=all in your sip.conf and put in allow= for the codec you want to use and disallow=all. Holy crap it worked! sip.conf disallow=all ; disallow all codecs allow=ulaw ; Allow all codecs allow=alaw ; Allow all codecs ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
