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Stefan at WPF

2012-Jun-18 20:04 UTC

[asterisk-users] Error SIP/2.0 488 Not acceptable here

Hello,

a person trying to call me by my phone number is getting the error 488 Not
acceptable here. I googled that error, seems like this error is normally
caused by a failed codec negotation, though I have no clue how I could have
read this out of the logs. Anyway, my setup is as follows:
Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider
The user calling me is also using Sipgate and is calling my landline phone
number from Sipgate (not [my sip id]@sipgate.de).

My sip.conf including the codec restrictions looks like this (I left out my
local sip account)

[general]> port=5060
> bindaddr=0.0.0.0
> context=other
> language=de
> allowguest=no
>
> qualify=no
> disallow=all
> allow=alaw
> allow=ulaw
> allow=g729
> allow=gsm
> allow=slinear
> srvlookup=yes
>
> register =>
<MY_SIP_ID>:<password>@sipgate.de/<MY_SIP_ID>
>
>
>
> [sipgate]
> type=friend
> insecure=invite
> nat=yes
> username=<MY_SIP_ID>
> fromuser=<MY_SIP_ID>
> fromdomain=sipgate.de
> secret=<password>
> host=sipgate.de
> qualify=yes
> canreinvite=no
> dtmfmode=rfc2833
> context = from_external_voip_provider
>

The relevant part from my full asterisk log /var/log/asterisk/full
including the 488 Not acceptable here error message:


[Jun 18 20:15:26] VERBOSE[1164] chan_sip.c:> <--- SIP read from UDP:217.10.79.9:5060 --->
> INVITE sip:<MY_SIP_ID>@192.168.5.11:5060 SIP/2.0
> Record-Route: <sip:217.10.79.9;lr;ftag=8cgn1bajqb>
> Record-Route: <sip:172.20.40.3;lr=on>
> Record-Route: <sip:217.10.79.9;lr;ftag=8cgn1bajqb>
> Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK8f5c.48627b3.0
> Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK8f5c.48627b3.0
> Via: SIP/2.0/UDP 217.10.79.9:5060
> ;received=217.10.68.222;branch=z9hG4bK-un6p0cm50qse
> Via: SIP/2.0/UDP 192.168.0.8:2048
>
;received=<CALLING_PARTY_IP_ADDRESS>;branch=z9hG4bK-un6p0cm50qse;rport=2048
> From: "sipgate.de"
<sip:<CALLING_PARTY_SIP_ID>@sipgate.de>;tag=8cgn1bajqb
> To: <sip:0049<MY_PHONE_NUMBER>@sipgate.de;user=phone>
> Call-ID: 4fdf703d880d-ywqwnfbbj1h7
> CSeq: 2 INVITE
> Max-Forwards: 67
> Contact:
>
<sip:<CALLING_PARTY_SIP_ID>@<CALLING_PARTY_IP_ADDRESS>:2048;line=swnt2d3t>;reg-id=1
> X-Serialnumber: 000413251D76
> User-Agent: snom300/8.7.3.7
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
> MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Supported: timer, 100rel, replaces, from-change
> Session-Expires: 3600;refresher=uas
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 522
> P-Asserted-Identity:
<sip:<CALLING_PARTY_PHONE_NUMBER>@sipgate.de>
>
> v=0
> o=root 269390684 269390684 IN IP4 192.168.0.8
> s=call
> c=IN IP4 217.10.77.20
> t=0 0
> m=audio 62652 RTP/AVP 9 0 8 3 99 108 18 101
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:Ed8iHaP3BXNVeXHj98PRa6sJyImNer3ImjUvDZps
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:99 G726-32/8000
> a=rtpmap:108 AAL2-G726-32/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
> a=direction:active
> a=nortpproxy:yes
> <------------->
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: --- (25 headers 21 lines) ---
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Sending to
217.10.79.9:5060(NAT)
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Using INVITE request as basis
> request - 4fdf703d880d-ywqwnfbbj1h7
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found peer 'sipgate'
for
> '<CALLING_PARTY_SIP_ID>' from 217.10.79.9:5060
> [Jun 18 20:15:26] VERBOSE[1164] netsock2.c:   == Using SIP RTP CoS mark 5
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 9
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 0
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 8
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 3
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 99
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 108
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 18
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 101
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format
> G722 for ID 9
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format
> PCMU for ID 0
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format
> PCMA for ID 8
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format
> GSM for ID 3
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format
> G726-32 for ID 99
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format
> AAL2-G726-32 for ID 108
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format
> G729 for ID 18
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format
> telephone-event for ID 101
> [Jun 18 20:15:26] WARNING[1164] chan_sip.c: We are requesting SRTP, but
> they responded without it!
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c:
> <--- Reliably Transmitting (NAT) to 217.10.79.9:5060 --->
> SIP/2.0 488 Not acceptable here
> Via: SIP/2.0/UDP 217.10.79.9:5060
> ;branch=z9hG4bK8f5c.48627b3.0;received=217.10.79.9;rport=5060
> Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK8f5c.48627b3.0
> Via: SIP/2.0/UDP 217.10.79.9:5060
> ;received=217.10.68.222;branch=z9hG4bK-un6p0cm50qse
> Via: SIP/2.0/UDP 192.168.0.8:2048
>
;received=<CALLING_PARTY_IP_ADDRESS>;branch=z9hG4bK-un6p0cm50qse;rport=2048
> From: "sipgate.de"
<sip:<CALLING_PARTY_SIP_ID>@sipgate.de>;tag=8cgn1bajqb
> To:
<sip:0049<MY_PHONE_NUMBER>@sipgate.de;user=phone>;tag=as6364b798
> Call-ID: 4fdf703d880d-ywqwnfbbj1h7
> CSeq: 2 INVITE
> Server: Asterisk PBX 1.8.13.0~dfsg-1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces, timer
> Content-Length: 0
>
I am having problems to see to what "488 Not acceptable here" relates
to?
What is not acceptable? Is it maybe about
> [Jun 18 20:15:26] WARNING[1164] chan_sip.c: We are requesting SRTP, but
> they responded without it!
and not a codec problem?


I am not sure if this is relevant and if it really shows the working
codecs, bot for completeness the outputs of "core show codecs" and
"core
show translation" follow:
> core show codecs
> Disclaimer: this command is for informational purposes only.
>         It does not indicate anything about your configuration.
>                 INT    BINARY                  HEX   TYPE       NAME
> DESCRIPTION
>
>
-----------------------------------------------------------------------------------
>                   1 (1 <<  0)                (0x1)  audio       g723
> (G.723.1)
>                   2 (1 <<  1)                (0x2)  audio        gsm
> (GSM)
>                   4 (1 <<  2)                (0x4)  audio       ulaw
> (G.711 u-law)
>                   8 (1 <<  3)                (0x8)  audio       alaw
> (G.711 A-law)
>                  16 (1 <<  4)               (0x10)  audio   g726aal2
> (G.726 AAL2)
>                  32 (1 <<  5)               (0x20)  audio      adpcm
> (ADPCM)
>                  64 (1 <<  6)               (0x40)  audio       slin 
(16
> bit Signed Linear PCM)
>                 128 (1 <<  7)               (0x80)  audio      lpc10
> (LPC10)
>                 256 (1 <<  8)              (0x100)  audio       g729
> (G.729A)
>                 512 (1 <<  9)              (0x200)  audio      speex
> (SpeeX)
>                1024 (1 << 10)              (0x400)  audio       ilbc
> (iLBC)
>                2048 (1 << 11)              (0x800)  audio       g726
> (G.726 RFC3551)
>                4096 (1 << 12)             (0x1000)  audio       g722
> (G722)
>                8192 (1 << 13)             (0x2000)  audio     siren7
> (ITU G.722.1 (Siren7, licensed from Polycom))
>               16384 (1 << 14)             (0x4000)  audio    siren14
> (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))
>               32768 (1 << 15)             (0x8000)  audio     slin16 
(16
> bit Signed Linear PCM (16kHz))
>               65536 (1 << 16)            (0x10000)  image       jpeg
> (JPEG image)
>              131072 (1 << 17)            (0x20000)  image        png
> (PNG image)
>              262144 (1 << 18)            (0x40000)  video       h261
> (H.261 Video)
>              524288 (1 << 19)            (0x80000)  video       h263
> (H.263 Video)
>             1048576 (1 << 20)           (0x100000)  video      h263p
> (H.263+ Video)
>             2097152 (1 << 21)           (0x200000)  video       h264
> (H.264 Video)
>             4194304 (1 << 22)           (0x400000)  video      mpeg4
> (MPEG4 Video)
>             8388608 (1 << 23)           (0x800000)  video    unknown
> (unknown)
>            16777216 (1 << 24)          (0x1000000)  video    unknown
> (unknown)
>            33554432 (1 << 25)          (0x2000000)   text    unknown
> (unknown)
>            67108864 (1 << 26)          (0x4000000)   text        red
> (T.140 Realtime Text with redundancy)
>           134217728 (1 << 27)          (0x8000000)   text       t140
> (Passthrough T.140 Realtime Text)
>           268435456 (1 << 28)         (0x10000000)   text    unknown
> (unknown)
>           536870912 (1 << 29)         (0x20000000)   text    unknown
> (unknown)
>          1073741824 (1 << 30)         (0x40000000)  (unk)    unknown
> (unknown)
>          2147483648 (1 << 31)         (0x80000000)  (unk)    unknown
> (unknown)
>          4294967296 (1 << 32)        (0x100000000)  audio       g719
> (ITU G.719)
>          8589934592 (1 << 33)        (0x200000000)  audio    speex16
> (SpeeX 16khz)
>         17179869184 (1 << 34)        (0x400000000)  audio    unknown
> (unknown)
>         34359738368 (1 << 35)        (0x800000000)  audio    unknown
> (unknown)
>         68719476736 (1 << 36)       (0x1000000000)  audio    unknown
> (unknown)
>        137438953472 (1 << 37)       (0x2000000000)  audio    unknown
> (unknown)
>        274877906944 (1 << 38)       (0x4000000000)  audio    unknown
> (unknown)
>        549755813888 (1 << 39)       (0x8000000000)  audio    unknown
> (unknown)
>       1099511627776 (1 << 40)      (0x10000000000)  audio    unknown
> (unknown)
>       2199023255552 (1 << 41)      (0x20000000000)  audio    unknown
> (unknown)
>       4398046511104 (1 << 42)      (0x40000000000)  audio    unknown
> (unknown)
>       8796093022208 (1 << 43)      (0x80000000000)  audio    unknown
> (unknown)
>      17592186044416 (1 << 44)     (0x100000000000)  audio    unknown
> (unknown)
>      35184372088832 (1 << 45)     (0x200000000000)  audio    unknown
> (unknown)
>      70368744177664 (1 << 46)     (0x400000000000)  audio    unknown
> (unknown)
>     140737488355328 (1 << 47)     (0x800000000000)  audio    testlaw
> (G.711 test-law)
>     281474976710656 (1 << 48)    (0x1000000000000)  video    unknown
> (unknown)
>     562949953421312 (1 << 49)    (0x2000000000000)  video    unknown
> (unknown)
>    1125899906842624 (1 << 50)    (0x4000000000000)  video    unknown
> (unknown)
>    2251799813685248 (1 << 51)    (0x8000000000000)  video    unknown
> (unknown)
>    4503599627370496 (1 << 52)   (0x10000000000000)  video    unknown
> (unknown)
>    9007199254740992 (1 << 53)   (0x20000000000000)  video    unknown
> (unknown)
>   18014398509481984 (1 << 54)   (0x40000000000000)  video    unknown
> (unknown)
>   36028797018963968 (1 << 55)   (0x80000000000000)  video    unknown
> (unknown)
>   72057594037927936 (1 << 56)  (0x100000000000000)  video    unknown
> (unknown)
>  144115188075855872 (1 << 57)  (0x200000000000000)  video    unknown
> (unknown)
>  288230376151711744 (1 << 58)  (0x400000000000000)  video    unknown
> (unknown)
>  576460752303423488 (1 << 59)  (0x800000000000000)  video    unknown
> (unknown)
> 1152921504606846976 (1 << 60) (0x1000000000000000)  video    unknown
> (unknown)
> 2305843009213693952 (1 << 61) (0x2000000000000000)  video    unknown
> (unknown)
> 4611686018427387904 (1 << 62) (0x4000000000000000)  video    unknown
> (unknown)
>

>  core show translation
>          Translation times between formats (in microseconds) for one
> second of data
>           Source Format (Rows) Destination Format (Columns)
>
>            g723    gsm   ulaw   alaw g726aal2  adpcm   slin  lpc10  g729
> speex  ilbc   g726  g722 siren7 siren14 slin16  g719 speex16 testlaw
>      g723     -      -      -      -        -      -      -      -
> -      -     -      -     -      -       -      -     -       -       -
>       gsm     -      -      2      2    10001      2      1  20001     -
> 90001     -  10001     2      -       -  70001     -  130001       2
>      ulaw     -  10001      -      1    10001      2      1  20001     -
> 90001     -  10001     2      -       -  70001     -  130001       2
>      alaw     -  10001      1      -    10001      2      1  20001     -
> 90001     -  10001     2      -       -  70001     -  130001       2
>  g726aal2     -  20000  10001  10001        -  10001  10000  30000     -
> 100000     -  20000 10001      -       -  80000     -  140000   10001
>     adpcm     -  10001      2      2    10001      -      1  20001     -
> 90001     -  10001     2      -       -  70001     -  130001       2
>      slin     -  10000      1      1    10000      1      -  20000     -
> 90000     -  10000     1      -       -  70000     -  130000       1
>     lpc10     -  20000  10001  10001    20000  10001  10000      -     -
> 100000     -  20000 10001      -       -  80000     -  140000   10001
>      g729     -      -      -      -        -      -      -      -
> -      -     -      -     -      -       -      -     -       -       -
>     speex     -  20000  10001  10001    20000  10001  10000  30000
> -      -     -  20000 10001      -       -  80000     -  140000   10001
>      ilbc     -      -      -      -        -      -      -      -
> -      -     -      -     -      -       -      -     -       -       -
>      g726     -  10001      2      2    10001      2      1  20001     -
> 90001     -      -     2      -       -  70001     -  130001       2
>      g722     -  20000  10001  10001    20000  10001  10000  30000     -
> 100000     -  20000     -      -       -  10000     -   70000   10001
>    siren7     -      -      -      -        -      -      -      -
> -      -     -      -     -      -       -      -     -       -       -
>   siren14     -      -      -      -        -      -      -      -
> -      -     -      -     -      -       -      -     -       -       -
>    slin16     - 170000 160001 160001   170000 160001 160000 180000     -
> 250000     - 170000 10000      -       -      -     -   60000  160001
>      g719     -      -      -      -        -      -      -      -
> -      -     -      -     -      -       -      -     -       -       -
>   speex16     - 180000 170001 170001   180000 170001 170000 190000     -
> 260000     - 180000 20000      -       -  10000     -       -  170001
>   testlaw     -  10001      2      2    10001      2      1  20001     -
> 90001     -  10001     2      -       -  70001     -  130001       -
>
>Thank you very much for any hint on this!


Best regards
Stefan
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Matthew Jordan

2012-Jun-18 20:31 UTC

head link

[asterisk-users] Error SIP/2.0 488 Not acceptable here

----- Original Message ----- 
> From: "Stefan at WPF" <stefan.at.wpf at googlemail.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Sent: Monday, June 18, 2012 3:04:32 PM
> Subject: [asterisk-users] Error SIP/2.0 488 Not acceptable here
> Hello,
> a person trying to call me by my phone number is getting the error
> 488 Not acceptable here. I googled that error, seems like this error
> is normally caused by a failed codec negotation, though I have no
> clue how I could have read this out of the logs. Anyway, my setup is
> as follows:
> Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider
> The user calling me is also using Sipgate and is calling my landline
> phone number from Sipgate (not [my sip id]@ sipgate.de ).
> My sip.conf including the codec restrictions looks like this (I left
> out my local sip account)
> > [general]
> 
> > port=5060
> 
> > bindaddr=0.0.0.0
> 
> > context=other
> 
> > language=de
> 
> > allowguest=no
> 
> > qualify=no
> 
> > disallow=all
> 
> > allow=alaw
> 
> > allow=ulaw
> 
> > allow=g729
> 
> > allow=gsm
> 
> > allow=slinear
> 
> > srvlookup=yes
> 
> > register => <MY_SIP_ID>:<password>@ sipgate.de/
<MY_SIP_ID>
> 
> > [sipgate]
> 
> > type=friend
> 
> > insecure=invite
> 
> > nat=yes
> 
> > username=<MY_SIP_ID>
> 
> > fromuser=<MY_SIP_ID>
> 
> > fromdomain= sipgate.de
> 
> > secret=<password>
> 
> > host= sipgate.de
> 
> > qualify=yes
> 
> > canreinvite=no
> 
> > dtmfmode=rfc2833
> 
> > context = from_external_voip_provider
> 
> The relevant part from my full asterisk log /var/log/asterisk/full
> including the 488 Not acceptable here error message:
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c:
> 
> > <--- SIP read from UDP: 217.10.79.9:5060 --->
> 
> > INVITE sip:<MY_SIP_ID>@ 192.168.5.11:5060 SIP/2.0
> 
> > Record-Route: <sip:217.10.79.9;lr;ftag=8cgn1bajqb>
> 
> > Record-Route: <sip:172.20.40.3;lr=on>
> 
> > Record-Route: <sip:217.10.79.9;lr;ftag=8cgn1bajqb>
> 
> > Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK8f5c.48627b3.0
> 
> > Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK8f5c.48627b3.0
> 
> > Via: SIP/2.0/UDP
> > 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK-un6p0cm50qse
> 
> > Via: SIP/2.0/UDP
> >
192.168.0.8:2048;received=<CALLING_PARTY_IP_ADDRESS>;branch=z9hG4bK-un6p0cm50qse;rport=2048
> 
> > From: " sipgate.de " <sip:<CALLING_PARTY_SIP_ID>@
sipgate.de
> > >;tag=8cgn1bajqb
> 
> > To: <sip:0049<MY_PHONE_NUMBER>@ sipgate.de ;user=phone>
> 
> > Call-ID: 4fdf703d880d-ywqwnfbbj1h7
> 
> > CSeq: 2 INVITE
> 
> > Max-Forwards: 67
> 
> > Contact:
> >
<sip:<CALLING_PARTY_SIP_ID>@<CALLING_PARTY_IP_ADDRESS>:2048;line=swnt2d3t>;reg-id=1
> 
> > X-Serialnumber: 000413251D76
> 
> > User-Agent: snom300/ 8.7.3.7
> 
> > Accept: application/sdp
> 
> > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> > PRACK, MESSAGE, INFO, UPDATE
> 
> > Allow-Events: talk, hold, refer, call-info
> 
> > Supported: timer, 100rel, replaces, from-change
> 
> > Session-Expires: 3600;refresher=uas
> 
> > Min-SE: 90
> 
> > Content-Type: application/sdp
> 
> > Content-Length: 522
> 
> > P-Asserted-Identity: <sip:<CALLING_PARTY_PHONE_NUMBER>@
sipgate.de
> > >
> 
> > v=0
> 
> > o=root 269390684 269390684 IN IP4 192.168.0.8
> 
> > s=call
> 
> > c=IN IP4 217.10.77.20
> 
> > t=0 0
> 
> > m=audio 62652 RTP/AVP 9 0 8 3 99 108 18 101
> 
> > a=crypto:1 AES_CM_128_HMAC_SHA1_32
> > inline:Ed8iHaP3BXNVeXHj98PRa6sJyImNer3ImjUvDZps
> 
> > a=rtpmap:9 G722/8000
> 
> > a=rtpmap:0 PCMU/8000
> 
> > a=rtpmap:8 PCMA/8000
> 
> > a=rtpmap:3 GSM/8000
> 
> > a=rtpmap:99 G726-32/8000
> 
> > a=rtpmap:108 AAL2-G726-32/8000
> 
> > a=rtpmap:18 G729/8000
> 
> > a=fmtp:18 annexb=no
> 
> > a=rtpmap:101 telephone-event/8000
> 
> > a=fmtp:101 0-15
> 
> > a=ptime:20
> 
> > a=sendrecv
> 
> > a=direction:active
> 
> > a=nortpproxy:yes
> 
> > <------------->
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: --- (25 headers 21
> > lines)
> > ---
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Sending to
> > 217.10.79.9:5060 (NAT)
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Using INVITE request as
> > basis request - 4fdf703d880d-ywqwnfbbj1h7
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found peer
'sipgate'
> > for
> > '<CALLING_PARTY_SIP_ID>' from 217.10.79.9:5060
> 
> > [Jun 18 20:15:26] VERBOSE[1164] netsock2.c: == Using SIP RTP CoS
> > mark
> > 5
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format
> > 9
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format
> > 0
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format
> > 8
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format
> > 3
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format
> > 99
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format
> > 108
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format
> > 18
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format
> > 101
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description
> > format G722 for ID 9
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description
> > format PCMU for ID 0
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description
> > format PCMA for ID 8
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description
> > format GSM for ID 3
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description
> > format G726-32 for ID 99
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description
> > format AAL2-G726-32 for ID 108
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description
> > format G729 for ID 18
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description
> > format telephone-event for ID 101
> 
> > [Jun 18 20:15:26] WARNING[1164] chan_sip.c: We are requesting SRTP,
> > but they responded without it!
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c:
> 
> > <--- Reliably Transmitting (NAT) to 217.10.79.9:5060 --->
> 
> > SIP/2.0 488 Not acceptable here
> 
> > Via: SIP/2.0/UDP
> >
217.10.79.9:5060;branch=z9hG4bK8f5c.48627b3.0;received=217.10.79.9;rport=5060
> 
> > Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK8f5c.48627b3.0
> 
> > Via: SIP/2.0/UDP
> > 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK-un6p0cm50qse
> 
> > Via: SIP/2.0/UDP
> >
192.168.0.8:2048;received=<CALLING_PARTY_IP_ADDRESS>;branch=z9hG4bK-un6p0cm50qse;rport=2048
> 
> > From: " sipgate.de " <sip:<CALLING_PARTY_SIP_ID>@
sipgate.de
> > >;tag=8cgn1bajqb
> 
> > To: <sip:0049<MY_PHONE_NUMBER>@ sipgate.de
> > ;user=phone>;tag=as6364b798
> 
> > Call-ID: 4fdf703d880d-ywqwnfbbj1h7
> 
> > CSeq: 2 INVITE
> 
> > Server: Asterisk PBX 1.8.13.0~dfsg-1
> 
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> > INFO, PUBLISH
> 
> > Supported: replaces, timer
> 
> > Content-Length: 0
> 
> I am having problems to see to what "488 Not acceptable here"
relates
> to? What is not acceptable? Is it maybe about
> > [Jun 18 20:15:26] WARNING[1164] chan_sip.c: We are requesting SRTP,
> > but they responded without it!
> 
Yes, that would be the problem.

The SIP UA is doing something a little wrong here by offering a security
description (crypto) without specifying that the audio/video protocol that
should be used as SRTP (RTP/SAVP).  Because the UA appears to be attempting
to negotiate a SRTP connection, Asterisk is checking if the peer has encryption
enabled.  Since the peer corresponding with the UA does not have encryption
enabled for it, Asterisk is responding with a 488 response.

SRTP security descriptions (such as 'crypto') must only be used with the
SRTP transport specified, e.g., RTP/SAVP or RTP/SAVPF.

--
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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