* You are viewing the archive for June 18th, 2012

fritzbox

Hi,

Couple of moments ago my asteriskbox with a bri-card went down.
(burn-out)

I’ve heard that it seems to be possible to use an fritz!box as an
isdn-gateway (isdn < --> sip)

Anyone around who has good/bad experiences with those AVM-boxes?

(yeah, i know it is tech overkill, but i’ll get an dualband wifi router,
that is Ipv6-ready with it)

Hans.

TDM410 PTSN line setup with 1 analog phone

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds
1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1
(not trying to use the gui, want to do everything by hand) with a TDM410 with
2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a
analog phone connected to port 3 (FXS). I compiled asterisk with asterisk
samples so I realize that may have messed me up.

This is all running on Ubuntu Server 12.04. I have been googling/researching
reading the book, etc. Everything I find is for SIP softphones etc. I just
want to start by getting the asterisk machine to provide dialtone to the analog
phone, and ring that phone when I call the PTSN line.

I must be missing something in the basic dahdi and dialplan to simple get the
analog phone to work. Can someone point me to a example of what I am trying to
accomplish? Not wanting handholding but a push in the right direction.

Thanks.

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

Asterisk 1.8.13.0 / problem with cdr logging (mysql, odbc)

Hi,

I am trying now for over 4 hours setting up cdr-logging via odbc into a
mysql database. But with no success. Do you have any hint for me?

cat /etc/odbc.ini

asterisk-users Digest, Vol 95, Issue 20

Anybody, can you please share your thoughts to overcome this issue?

> Hi,
>
> I’m getting error: ‘ FAX session ‘9’ is complete, result: ‘FAILED’
> (FAX_FAILURE_PARTIAL), error: ‘3RD_T2_TIMEOUT’, pages: 1, resolution:
> ‘204×196′, transfer rate: ‘9600’, remoteSID: ” ‘ when I tried send fax
> more than 2 pages to Asterisk using T.38.
>
> First I set speed rate to 14400 which I was getting same error message
> while sending 2 fax pages document. Later I set the speed rate for sending
> fax machine to 9600(which is the lowest speed rate available fax machine),
> I was able to send 2 pages document fax but tried to send 3 pages document,
> I’m getting this error message.
>
> The Asterisk version I’m using is 10.4.2. Please advise me at earliest to
> overcome this issue
>
> Note: Logs can also be provided as per request
>
> –
> Regards,
>
> Ahmed Munir Chohan
>