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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 => : @sipgate.de/
>
>
>
> [sipgate]
> type=friend
> insecure=invite
> nat=yes
> username=

> fromuser=

> fromdomain=sipgate.de
> secret= > 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:@192.168.5.11:5060 SIP/2.0
> Record-Route:
> Record-Route:
> Record-Route:
> 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=;branch=z9hG4bK-un6p0cm50qse;rport=2048
> From: “sipgate.de” @sipgate.de>;tag=8cgn1bajqb
> To: @sipgate.de;user=phone>
> Call-ID: 4fdf703d880d-ywqwnfbbj1h7
> CSeq: 2 INVITE
> Max-Forwards: 67
> Contact:
> @: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: @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
> ‘‘ 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=;branch=z9hG4bK-un6p0cm50qse;rport=2048
> From: “sipgate.de” @sipgate.de>;tag=8cgn1bajqb
> To: @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

No progress tones on transferred call

Asterisk 1.4
We are experiencing an issue on transfers where no progress tones are heard
by the caller:
1. Call from 1593 (SPA525G 0026998D2FFFF) to 1595 (SPA922 000B820AFFFFF).
1595 answers
2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4AFFFF). 1595 hears
MoH.
3. 1597 starts ringing and 1593 presses transfer again. MoH stops but 1595
hears no ringing
When xfer is pressed and the extension is dialled:
U 203.89.001.001:5060 -> 121.98.001.001:1034
INVITE sip:1CDF0F4AFFFF@192.168.1.72:5060 SIP/2.0..Via: SIP/2.0/UDP
203.89.001.001:5060;branch=z9hG4bK5286810e;rport..From: “C Allerid”
;tag=as72616c50..To:
..Contact:
..Call-ID:
59ba10300b9b8cb5684eba2368c90a77@203.89.001.001..CSeq: 102
INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70..Date: Tue, 05 Jun 2012
08:05:02 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO..Supported: replaces..Content-Type:
application/sdp..Content-Length: 262….v=0..o=root 3031 3031 IN IP4
203.89.001.001..s=session..c=IN IP4 203.89.001.001..t=0 0..m=audio 13728
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:1010-16..a=ptime:20..a=sendrecv..
U 121.98.001.001:1034 -> 203.89.001.001:5060
SIP/2.0 100 Trying..To: ..From: “C
Allerid” ;tag=as72616c50..Call-ID:
59ba10300b9b8cb5684eba2368c90a77@203.89.001.001..CSeq: 102 INVITE..Via:
SIP/2.0/UDP 203.89.001.001:5060;branch=z9hG4bK5286810e..Server:
Cisco/SPA508G-7.4.9c..Content-Length: 0….
U 121.98.001.001:1034 -> 203.89.001.001:5060
SIP/2.0 180 Ringing..To:
;tag=53e23c5265d60f06i0..From: “C
Allerid”
;tag=as72616c50..Call-ID:59ba10300b9b8cb5684eba2368
c90a77@203.89.001.001..CSeq: 102 INVITE..Via: SIP/2.0/UDP
203.89.001.001:5060;branch=z9hG4bK5286810e..Contact: “$USER”
..Server:
Cisco/SPA508G-7.4.9c..Content-Length: 0….
After transfer is pressed the second time there is no further SIP messages
with

Asterisk CLI

No extension found ?

Hi

I have a small problems with incoming call.

I have a peer actually configured for outcall :

sip.conf:

[Trunk-Telco]
type=peer
host=domaineofmysupplier.net
outboundproxy=domaineofmysupplier.net
session-timers=originate
session-expires=7200
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
dtmfmode=rfc2833
disallow=all
allow=alaw
insecure=port,invite
context=incoming

This SIP account work for outgoing call. when i want receive call from
this sipplier, i have a “extension not found”.

In extensions.conf for incoming:

[incoming]
exten => _X.,1,Dial(IAX2/VoIP/${EXTEN},180,rt)

in dialplan show incoming, no problems i see the dialplan.

when i call, i have:

< --- SIP read from UDP://84.xx.xx.72:5060 —>
INVITE sip:331NUMNOFOUND@78.IPOFMYSERVER:5060 SIP/2.0
Record-Route:
Record-Route:
Record-Route:

Record-Route:

Via: SIP/2.0/UDP 84.xx.xx.72;branch=z9hG4bK10e4.e7f23f11.0
Via: SIP/2.0/UDP 172.16.21.67;branch=z9hG4bK10e4.bbf4c444.0
Via: SIP/2.0/UDP 172.16.20.119;branch=z9hG4bK10e4.9fe53c91.0
Via: SIP/2.0/UDP 172.16.21.11:5060;branch=z9hG4bK00151747E2606DB6CA39464AF542
From: “+331MYCLID”
;tag=2RUVP51HBW30000E1D00001u0K4NFQC0QNAN31
To:

Call-ID:
60471500e217-4f924d2c-477df10c-66ea6f8-140732f@127.0.0.1
CSeq: 20114 INVITE
Contact:
Allow-Events: refer
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
Content-Type: application/sdp
Max-Forwards: 67
P-Asserted-Identity:

Supported: timer, replaces
Content-Length: 369
Min-SE: 90
Session-Expires: 300
P-Charging-Vector: icid-value=”4f924d2c1e20abe1d@172.16.20.119″
X-PSN-Trunk: ME

v=0
o=- 18406958643964291255 1 IN IP4 172.16.21.11
s=session
c=IN IP4 84.xx.xx.34
t=0 0
m=audio 64296 RTP/AVP 8 18 4 0 105 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=fmtp:4 bitrate=6.3
a=rtpmap:0 PCMU/8000
a=rtpmap:105 X-CCD/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
a=nortpproxy:yes

< ------------->

No IVR audio. Jump in RTP sequence number

My users dial *120 get to an IVR menu that plays their balance and then
ask them for a voucher. Ater the balance is played and the request for
the voucher is played the user don’t hear any other audio from the
asterisk box. I can see the asterisk server playing the files to ask
for the voucher again but the user cannot hear any thing.

Has any one seens this issue with IVRs. I notice a change in RTP
sequence when voucher is being requested again.

sip debug
< --- SIP read from UDP:x.x.x.x:5060 --->
INVITE sip:*120@a.b.c.d SIP/2.0
Via: SIP/2.0/UDP x.x.x.x;rport;branch=z9hG4bK2vr8B7r60vS7j
Max-Forwards: 70
From: “14735201326″ ;tag=0K219XHeF7K2j
To:
Call-ID: 00ddbda6-d9b1-122f-e7a7-00259025b560
CSeq: 24716447 INVITE
Contact:
User-Agent: Wireless Call Manager
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 231
Remote-Party-ID: “14735201326″
;party=calling;screen=yes;privacy=off

v=0
o=wCM 1330087502 1330087503 IN IP4 x.x.x.x
s=wCM
c=IN IP4 x.x.x.x
t=0 0
m=audio 17520 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
< ------------->

Presence subscription from other pbx systems

Hi members,

I have a question regarding presence in asterisk.

I have two PBX systems and would like to connect them. After configuring
each other as sip providers calls between users of the pbx systems are
possible.

Now I’m trying to implement presence between the systems. PBX1 sends
dialog-event SUBSCRIBE messages to PBX2. Asterisk just answers 404 not found
although user 410 exists. I think this is for security reasons. Is there an
option to allow presence subscription from configured providers?

Sincerely

Jan

PS: Here are sample sip messages:

< --- SIP read from UDP:10.99.10.2:5060 --->

SUBSCRIBE sip:410@10.99.10.14;sipx-noroute=VoiceMail;sipx-userforward=false
SIP/2.0

Record-Route:
e9d680ffaa83f6db4234704>

From: ;tag=XHK1Gy

To:

Call-Id: eZwhSebwLCc187

Cseq: 2 SUBSCRIBE

Contact:

Event: dialog

Accept: application/dialog-info+xml

Expires: 3153

Date: Mon, 13 Feb 2012 09:45:50 GMT

Max-Forwards: 19

User-Agent: sipXecs/4.4.0 sipXecs/rls (Linux)

Accept-Language: en

Proxy-Authorization: Digest username=”~~id~sipXrls”,
realm=”voip.mydomain.local”,
nonce=”3998fbca7da46e21895d383a16356f424f38dbce”,
uri=”sip:410@10.99.10.14;sipx-noroute=VoiceMail;sipx-userforward=false”,
response=”53ae73a9ce6a3a6acbe35deda3f731be”, cnonce=”a42sMg”, qop=auth,
nc=00000001

Via: SIP/2.0/UDP 10.99.10.2;branch=z9hG4bK-XX-18ddpkccVUQr6IO02D7a9Q5x0A

Via: SIP/2.0/UDP
10.99.10.1:51829;branch=z9hG4bK-XX-f75bT2Zlly8RPJBMDcOw5dyOxw

Content-Length: 0

< ------------->

TLS bug in asterisk?

Hi folks.

I’ve got a problem dialing with my new Snom M9 via TLS on asterisk 1.8.7.1 .
Registration works like a charm – the phone becomes ‘AVAILABLE’.
An INVITE is responded by a 401 to be expected, but then asterisk closes the TLS connection upon the Snom’s ACK.

The authenticated INVITE the Snom sends immediately after the ACK meets a closed socket and merely triggers a TCP RST packet on asterisk’s behalf.

There’s no ERROR or WARNING put out on the asterisk CLI.
The only hint I get is asterisk complaining about not finding the CSeq anymore it used a second ago for the beginning of the dialog.

I couldn’t really figure a reason for asterisk to close the connection when it should wait for an authenticated INVITE, so I posted the problem details in the bug tracker under
https://issues.asterisk.org/jira/browse/ASTERISK-19003?focusedCommentId=186012#comment-186012

I’d be very happy though, if someone could show me that this is not a bug, or how to work around it (I’ve got the Snom for about one more week, and then I’ll have to decide whether to return it ;) ).

Cheers,
Gregor.