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f.namuri at credires.it
Guest
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 Posted: Wed Jan 15, 2014 3:59 am Post subject: [asterisk-users] No compatible codecs, not accepting this of |
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|
Hello,
I'm having this issue on my pbx, it appears that asterisk is refusing
the codecs that my providers is proposing.
My trunk configuration is:
---
username=5x5x7x9x0x3
type=friend
secret=CRcxn7sqwm
qualify=yes
port=5060
insecure=port,invite
host=sip.txtxlxoxp.it
fromuser=5x5x7x9x0x3
fromdomain=sip.txtxlxoxp.it
disallow=all
context=from-trunk
allow=alaw
---
A typical invite from my provider is:
<--- SIP read from UDP:xx.yy.xx.yy:5060 ---> INVITE sip:5x5x7x9x0x3@192.168.1.168:5060 SIP/2.0
Via: SIP/2.0/UDP xx.yy.xx.yy:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7
From: <sip:3x8x6x3x3x@10.39.1.19;user=phone>;tag=SDdgce901-90915
To: "SIPLineUser SIPLineUser"<sip:5x5x7x9x0x3@sip.txtxlxoxp.it>
Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1
CSeq: 59458 INVITE
Content-Type: application/sdp
Contact: <sip:3x8x6x3x3x@xx.yy.xx.yy:5060;user=phone;transport=udp>
User-Agent: Nortel SESM 14.1.0.12
Max-Forwards: 19
Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin,com.nortelnetworks.im.encryption,replaces,100rel
Remote-Party-ID: <sip:3x8x6x3x3x@sip.txtxlxoxp.it;user=phone>;screen=yes;screen-ind=0;party=calling;counter=0;npi=NPI_E164;ton=TON_NATIONAL
P-Asserted-Identity: <sip:3x8x6x3x3x@10.39.1.19;user=phone>
Allow: UPDATE,REFER
Content-Length: 293
v=0
o=- 0 138163748 IN IP4 xx.yy.xx.yy
s=IMSS
e=unknown@invalid.net
c=IN IP4 xx.yy.xx.yy
t=0 0
m=audio 43718 RTP/AVP 8 18 3 101
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sqn: 0
a=cdsc: 1 image udptl t38
<------------->
<--- SIP read from UDP:xx.yy.xx.yy:5060 ---> INVITE sip:5x5x7x9x0x3@192.168.1.168:5060 SIP/2.0
Via: SIP/2.0/UDP xx.yy.xx.yy:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7
From: <sip:3x8x6x3x3x@10.39.1.19;user=phone>;tag=SDdgce901-90915
To: "SIPLineUser SIPLineUser"<sip:5x5x7x9x0x3@sip.txtxlxoxp.it>
Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1
CSeq: 59458 INVITE
Content-Type: application/sdp
Contact: <sip:3x8x6x3x3x@xx.yy.xx.yy:5060;user=phone;transport=udp>
User-Agent: Nortel SESM 14.1.0.12
Max-Forwards: 19
Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin,com.nortelnetworks.im.encryption,replaces,100rel
Remote-Party-ID: <sip:3x8x6x3x3x@sip.txtxlxoxp.it;user=phone>;screen=yes;screen-ind=0;party=calling;counter=0;npi=NPI_E164;ton=TON_NATIONAL
P-Asserted-Identity: <sip:3x8x6x3x3x@10.39.1.19;user=phone>
Allow: UPDATE,REFER
Content-Length: 293
v=0
o=- 0 138163748 IN IP4 xx.yy.xx.yy
s=IMSS
e=unknown@invalid.net
c=IN IP4 xx.yy.xx.yy
t=0 0
m=audio 43718 RTP/AVP 8 18 3 101
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sqn: 0
a=cdsc: 1 image udptl t38
<------------->
I noted that in the invite I get the rtpmap attribute only for codec 18,
3 but not for 8, it could be a problem?
The refuse is:
<--- Reliably Transmitting (NAT) to xx.yy.xx.yy:5060 --->
SIP/2.0 488 Not acceptable here^M
Via: SIP/2.0/UDP 77.239.128.7:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7;received=77.239.128.7;rport=5060^M
From: <sip:3x8x6x3x3x@10.39.1.19;user=phone>;tag=SDdgce901-90915^M
To: "SIPLineUser SIPLineUser"<sip:5x5x7x9x0x3@sip.totalvoip.it>;tag=as08516b97^M
Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1^M
CSeq: 59458 INVITE^M
Server: FPBX-2.11.0(10.12.3)^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Reason: Q.850;cause=58^M
Content-Length: 0^M
^M
<------------>
Have you any advice on how to troubleshoot it?
Thanks in advance
All the best,
Francesco Namuri
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james at fivecats.org
Guest
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| Posted: Wed Jan 15, 2014 4:09 am Post subject: [asterisk-users] No compatible codecs, not accepting this of |
|
|
On 1/15/2014 3:59 AM, Francesco Namuri wrote:
| Quote: | Hello,
I'm having this issue on my pbx, it appears that asterisk is refusing
the codecs that my providers is proposing.
My trunk configuration is:
|
Pretty simple -
| Quote: | ---
username=5x5x7x9x0x3
type=friend
secret=CRcxn7sqwm
qualify=yes
port=5060
insecure=port,invite
host=sip.txtxlxoxp.it
fromuser=5x5x7x9x0x3
fromdomain=sip.txtxlxoxp.it
disallow=all
context=from-trunk
allow=alaw
|
Here you're disallowing all codecs except alaw.
| Quote: | ---
A typical invite from my provider is:
<--- SIP read from UDP:xx.yy.xx.yy:5060 ---> INVITE sip:5x5x7x9x0x3@192.168.1.168:5060 SIP/2.0
Via: SIP/2.0/UDP xx.yy.xx.yy:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7
From: <sip:3x8x6x3x3x@10.39.1.19;user=phone>;tag=SDdgce901-90915
To: "SIPLineUser SIPLineUser"<sip:5x5x7x9x0x3@sip.txtxlxoxp.it>
Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1
CSeq: 59458 INVITE
Content-Type: application/sdp
Contact: <sip:3x8x6x3x3x@xx.yy.xx.yy:5060;user=phone;transport=udp>
User-Agent: Nortel SESM 14.1.0.12
Max-Forwards: 19
Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin,com.nortelnetworks.im.encryption,replaces,100rel
Remote-Party-ID: <sip:3x8x6x3x3x@sip.txtxlxoxp.it;user=phone>;screen=yes;screen-ind=0;party=calling;counter=0;npi=NPI_E164;ton=TON_NATIONAL
P-Asserted-Identity: <sip:3x8x6x3x3x@10.39.1.19;user=phone>
Allow: UPDATE,REFER
Content-Length: 293
v=0
o=- 0 138163748 IN IP4 xx.yy.xx.yy
s=IMSS
e=unknown@invalid.net
c=IN IP4 xx.yy.xx.yy
t=0 0
m=audio 43718 RTP/AVP 8 18 3 101
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000/1
|
But....provider will only send GSM or G729.
So either you need to talk your provider into sending alaw or you need
change your allow line to "allow=alaw,gsm".
--
_____________________________________________________________________
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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f.namuri at credires.it
Guest
|
| Posted: Wed Jan 15, 2014 4:39 am Post subject: [asterisk-users] No compatible codecs, not accepting this of |
|
|
Il 15/01/2014 10.09, James Sharp ha scritto:
| Quote: | On 1/15/2014 3:59 AM, Francesco Namuri wrote:
| Quote: | Hello,
I'm having this issue on my pbx, it appears that asterisk is refusing
the codecs that my providers is proposing.
My trunk configuration is:
|
Pretty simple -
| Quote: | ---
username=5x5x7x9x0x3
type=friend
secret=CRcxn7sqwm
qualify=yes
port=5060
insecure=port,invite
host=sip.txtxlxoxp.it
fromuser=5x5x7x9x0x3
fromdomain=sip.txtxlxoxp.it
disallow=all
context=from-trunk
allow=alaw
|
Here you're disallowing all codecs except alaw.
| Quote: | ---
A typical invite from my provider is:
<--- SIP read from UDP:xx.yy.xx.yy:5060 ---> INVITE
sip:5x5x7x9x0x3@192.168.1.168:5060 SIP/2.0
Via: SIP/2.0/UDP
xx.yy.xx.yy:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7
From: <sip:3x8x6x3x3x@10.39.1.19;user=phone>;tag=SDdgce901-90915
To: "SIPLineUser SIPLineUser"<sip:5x5x7x9x0x3@sip.txtxlxoxp.it>
Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1
CSeq: 59458 INVITE
Content-Type: application/sdp
Contact: <sip:3x8x6x3x3x@xx.yy.xx.yy:5060;user=phone;transport=udp>
User-Agent: Nortel SESM 14.1.0.12
Max-Forwards: 19
Supported:
com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin,com.nortelnetworks.im.encryption,replaces,100rel
Remote-Party-ID:
<sip:3x8x6x3x3x@sip.txtxlxoxp.it;user=phone>;screen=yes;screen-ind=0;party=calling;counter=0;npi=NPI_E164;ton=TON_NATIONAL
P-Asserted-Identity: <sip:3x8x6x3x3x@10.39.1.19;user=phone>
Allow: UPDATE,REFER
Content-Length: 293
v=0
o=- 0 138163748 IN IP4 xx.yy.xx.yy
s=IMSS
e=unknown@invalid.net
c=IN IP4 xx.yy.xx.yy
t=0 0
m=audio 43718 RTP/AVP 8 18 3 101
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000/1
|
But....provider will only send GSM or G729.
So either you need to talk your provider into sending alaw or you need
change your allow line to "allow=alaw,gsm"
|
Hello James,
thanks for your answer, I supposed this too, but my provider answered me
that as
m=audio 43718 RTP/AVP 8 18 3 101
^ ^ ^---- GSM proposal
^ ^------- G729 proposal
^---------- aLaw proposal
And that
a=rtpmap:18 G729/8000 proposed as media conversion
a=rtpmap:3 GSM/8000/1 because the call is made by a mobile
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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mailinglist+asterisk a...
Guest
|
| Posted: Wed Jan 15, 2014 5:50 am Post subject: [asterisk-users] No compatible codecs, not accepting this of |
|
|
On 15/01/14 09:39, Francesco Namuri wrote:
| Quote: | Hello James,
thanks for your answer, I supposed this too, but my provider answered me
that as
m=audio 43718 RTP/AVP 8 18 3 101
^ ^ ^---- GSM proposal
^ ^------- G729 proposal
^---------- aLaw proposal
And that
a=rtpmap:18 G729/8000 proposed as media conversion
a=rtpmap:3 GSM/8000/1 because the call is made by a mobile
|
I would agree with what your service provider has said. If you look at
the RFC http://tools.ietf.org/html/rfc4566#section-5.14 the '8 18 3 101'
parameters are a list of media formats. The first is the one which
should be used but (preferred choice) but the other may be used. Numbers
in the range 96-127 are dynamic payload types and these must have a
corresponding 'a=' line specifying the payload type and the codec options.
Lower numbers have static payload assignments and according to that RFC
dont have to have corresponding 'a=' lines. A list of types can be found
at http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml
However in all SIP traces I have seen there has always been a 'a=' line
for every payload type offered. The static payload type numbers are used
but there is still the 'a=' line.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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james at fivecats.org
Guest
|
| Posted: Wed Jan 15, 2014 5:54 am Post subject: [asterisk-users] No compatible codecs, not accepting this of |
|
|
On 1/15/2014 5:50 AM, Gareth Blades wrote:
| Quote: | On 15/01/14 09:39, Francesco Namuri wrote:
| Quote: | Hello James,
thanks for your answer, I supposed this too, but my provider answered me
that as
m=audio 43718 RTP/AVP 8 18 3 101
^ ^ ^---- GSM proposal
^ ^------- G729 proposal
^---------- aLaw proposal
And that
a=rtpmap:18 G729/8000 proposed as media conversion
a=rtpmap:3 GSM/8000/1 because the call is made by a mobile
|
I would agree with what your service provider has said. If you look at
the RFC http://tools.ietf.org/html/rfc4566#section-5.14 the '8 18 3 101'
parameters are a list of media formats. The first is the one which
should be used but (preferred choice) but the other may be used. Numbers
in the range 96-127 are dynamic payload types and these must have a
corresponding 'a=' line specifying the payload type and the codec options.
Lower numbers have static payload assignments and according to that RFC
dont have to have corresponding 'a=' lines. A list of types can be found
at http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml
However in all SIP traces I have seen there has always been a 'a=' line
for every payload type offered. The static payload type numbers are used
but there is still the 'a=' line.
|
I missed the RTP/AVP line. I'll go back to lurking for a bit 
--
_____________________________________________________________________
-- 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
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f.namuri at credires.it
Guest
|
| Posted: Wed Jan 15, 2014 9:35 am Post subject: [asterisk-users] No compatible codecs, not accepting this of |
|
|
Il 15/01/2014 09.59, Francesco Namuri ha scritto:
| Quote: | Hello,
I'm having this issue on my pbx, it appears that asterisk is refusing
the codecs that my providers is proposing.
My trunk configuration is:
---
username=5x5x7x9x0x3
type=friend
secret=CRcxn7sqwm
qualify=yes
port=5060
insecure=port,invite
host=sip.txtxlxoxp.it
fromuser=5x5x7x9x0x3
fromdomain=sip.txtxlxoxp.it
disallow=all
context=from-trunk
allow=alaw
---
A typical invite from my provider is:
<--- SIP read from UDP:xx.yy.xx.yy:5060 ---> INVITE sip:5x5x7x9x0x3@192.168.1.168:5060 SIP/2.0
Via: SIP/2.0/UDP xx.yy.xx.yy:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7
From: <sip:3x8x6x3x3x@10.39.1.19;user=phone>;tag=SDdgce901-90915
To: "SIPLineUser SIPLineUser"<sip:5x5x7x9x0x3@sip.txtxlxoxp.it>
Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1
CSeq: 59458 INVITE
Content-Type: application/sdp
Contact: <sip:3x8x6x3x3x@xx.yy.xx.yy:5060;user=phone;transport=udp>
User-Agent: Nortel SESM 14.1.0.12
Max-Forwards: 19
Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin,com.nortelnetworks.im.encryption,replaces,100rel
Remote-Party-ID: <sip:3x8x6x3x3x@sip.txtxlxoxp.it;user=phone>;screen=yes;screen-ind=0;party=calling;counter=0;npi=NPI_E164;ton=TON_NATIONAL
P-Asserted-Identity: <sip:3x8x6x3x3x@10.39.1.19;user=phone>
Allow: UPDATE,REFER
Content-Length: 293
v=0
o=- 0 138163748 IN IP4 xx.yy.xx.yy
s=IMSS
e=unknown@invalid.net
c=IN IP4 xx.yy.xx.yy
t=0 0
m=audio 43718 RTP/AVP 8 18 3 101
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sqn: 0
a=cdsc: 1 image udptl t38
<------------->
<--- SIP read from UDP:xx.yy.xx.yy:5060 ---> INVITE sip:5x5x7x9x0x3@192.168.1.168:5060 SIP/2.0
Via: SIP/2.0/UDP xx.yy.xx.yy:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7
From: <sip:3x8x6x3x3x@10.39.1.19;user=phone>;tag=SDdgce901-90915
To: "SIPLineUser SIPLineUser"<sip:5x5x7x9x0x3@sip.txtxlxoxp.it>
Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1
CSeq: 59458 INVITE
Content-Type: application/sdp
Contact: <sip:3x8x6x3x3x@xx.yy.xx.yy:5060;user=phone;transport=udp>
User-Agent: Nortel SESM 14.1.0.12
Max-Forwards: 19
Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin,com.nortelnetworks.im.encryption,replaces,100rel
Remote-Party-ID: <sip:3x8x6x3x3x@sip.txtxlxoxp.it;user=phone>;screen=yes;screen-ind=0;party=calling;counter=0;npi=NPI_E164;ton=TON_NATIONAL
P-Asserted-Identity: <sip:3x8x6x3x3x@10.39.1.19;user=phone>
Allow: UPDATE,REFER
Content-Length: 293
v=0
o=- 0 138163748 IN IP4 xx.yy.xx.yy
s=IMSS
e=unknown@invalid.net
c=IN IP4 xx.yy.xx.yy
t=0 0
m=audio 43718 RTP/AVP 8 18 3 101
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sqn: 0
a=cdsc: 1 image udptl t38
<------------->
I noted that in the invite I get the rtpmap attribute only for codec 18,
3 but not for 8, it could be a problem?
The refuse is:
<--- Reliably Transmitting (NAT) to xx.yy.xx.yy:5060 --->
SIP/2.0 488 Not acceptable here^M
Via: SIP/2.0/UDP 77.239.128.7:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7;received=77.239.128.7;rport=5060^M
From: <sip:3x8x6x3x3x@10.39.1.19;user=phone>;tag=SDdgce901-90915^M
To: "SIPLineUser SIPLineUser"<sip:5x5x7x9x0x3@sip.totalvoip.it>;tag=as08516b97^M
Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1^M
CSeq: 59458 INVITE^M
Server: FPBX-2.11.0(10.12.3)^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Reason: Q.850;cause=58^M
Content-Length: 0^M
^M
<------------>
|
Found the problem, but I'm wondering how it's possible...
I've a wrong configuration in a trunk:
---------
[dev0x8x4x7x1x]
disallow=all
username=0x8x4x7x1x
type=friend
secret=secret
qualify=yes
port=5060
insecure=port,invite
host=siprouter.devtel.it
fromuser=0x8x4x7x1x
fromdomain=sxpxoxtxr.xextxl.xt
context=from-trunk-dxvxtxallow=alaw
---------
but this is not the incriminated trunk, it's only one of the trunks of
this provider.
This wrong configuration makes unusable all the trunks of this provider
(only incoming calls), also if other trunk (as in my case) are
configured correctly.
Another strange behavior is that my other providers works good also with
the misconfigured trunk.
Doing a dump of a INVITE from others server I get an a= attribute ofr
any codec allowed...
Maybe is this the problem?
Thanks again for all answers...
All the best
Francesco
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