* You are viewing Posts Tagged ‘sipgate’

Sipgate Outgoing Calls

Hello

i am trying to setup sipgate gateway

i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line

– Called 01179248615@sipgate
[Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885
handle_response_invite: Failed to authenticate on INVITE to
‘”01179553708″ ;tag=as30eb9dd1′
— SIP/sipgate-0000014d is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)


here is my sip.conf file


[general]
port = 5060
bindaddr = 0.0.0.0
context

Sip Trunk Failing To Register Causes Sip Phones To Become Unreachable

Hi,

I have only seen this problem when using SIPgate SIP trunks which actually “register”. If the ADSL connection goes down that the sip trunk uses, the sip phones registered locally become unreachable. This happens on any 1.6.x or 1.8 version of asterisk I’ve tried. Is there a work around that doesn’t involve putting an opensips server between the asterisk server and the sip trunk?

Thanks.

Regards, John

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

Missing voicemail prompt beginning

Hello,

I am using the voicemail module of asterisk. When I did some test calls
from my mobile phone, sometimes the beginning of the prompt was missing,
e.g. instead of something like “number 12345 not available” I was only
hearing “345 not available”. Verbose level 5 on the asterisk console didn’t
give me any hint on this, it only shows that playback of the prompt started
correctly in every test case. Any hints on how I can debug this? I think
it’s some problem on my local configuration, I doubt it’s a problem with my
SIP provider or mobile phone provider, they are both very reliable (Sipgate
and T-Mobile).

Thanks for any hint!

Best regards
Stefan

asterisk not connecting to sipgate / NAT related issue?

Hello,

I configured asterisk in sip.conf like that:

=====

register => username:secret@sipgate.de:5060/number

[sipgate-out]
port=5060
type=friend
insecure=invite
nat=yes
username=username
fromuser=username
fromdomain=sipgate.de
secret=secret
host=sipgate.de
qualify=5000
canreinvite=no

=====

But all I get on CLI is:

[Jan 19 15:59:55] NOTICE[1928]: chan_sip.c:12104 sip_reg_timeout: –
Registration for ‘username@sipgate.de’ timed out, trying again (Attempt #1)

My first impression was that there must be a NAT related issue, so I
captured some SIP debug packets and got:

=====

Retransmitting #6 (NAT) to 217.10.79.9:5060:
OPTIONS sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP 192.168.23.30:5060;branch=z9hG4bK4c5e9a7f;rport
Max-Forwards: 70
From: “asterisk” ;tag=as669e138a
To:
Contact:
Call-ID: 383e3671142f9ec4312d5a263b225629@192.168.23.30
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.21
Date: Thu, 19 Jan 2012 15:00:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

=====

As you can see, asterisk writes the local ip address in the VIA header.
Is this the supposed behaviour?

As far as my understanding of the NAT problem goes, this is exactly what
I don’t want to be happening.

I’m familiar with the general NAT problem with layer 7 protocols relying
on ip addresses in the content of a layer 3 packet, but I thought,
asterisk would replace the local ip with the WAN ip.

Even setting “externip=xx.xx.xx.xx” does not change anything.

Anyone can give me advice?

kind regards,
Ruben

rtp.conf and Asterisk as a sip agent/client

Hello list,

I use Asterisk with one sipgate.co.uk trunk. Asterisk connects to
sipgate.co.uk as a sip agent/client (with “register =>” statement in
sip.conf).

If I restrict the number of ports used in rtp.conf (to 10000-10005 for
example) – will that affect the sip sessions to sipgate.co.uk as well -
or only those sessions where Asterisk acts as a sip proxy/server?

Many thanks,

Sebastian