Error SIP/2.0 488 Not acceptable here

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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…

Asterisk Users 3.2 years ago 3 Answers

No progress tones on transferred call

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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:

Asterisk Users 3.2 years ago 0 Answers

No extension found ?

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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: < ---…

Asterisk Users 3.3 years ago 5 Answers

No IVR audio. Jump in RTP sequence number

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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…

Asterisk Users 3.5 years ago 0 Answers

Presence subscription from other pbx systems

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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…

Asterisk Users 3.5 years ago 0 Answers

TLS bug in asterisk?

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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…

Asterisk Users 3.7 years ago 0 Answers

Trying to send customer mwi updates

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Hi all, I'm trying to send message waiting updates to my phones vai perl and sipsak. What I've got so far is: ====================================================
#!/usr/bin/perl $user="phone_login";
$DOMAIN="phone_ip";
$port=phone_port;
$SIP_FROM="Asterisk";
$SIP_SERVER="sipserver.example.com";
$SEQUENCE=1;
$CONTENT_LENGTH=300;
$NEW_MESSAGES=3;
$OLD_MESSAGES=4;
$HAS_NEW=1; $s .= "NOTIFY sip:${user}@${DOMAIN} SIP/2.0rl";
$s .= "From: rl";
$s .= "To:
rl";
$s .= "Contact:
rl";
$s .= "Call-ID: ${SEQUENCE}@${SIP_SERVER}rl";
$s .= "CSeq: ${SEQUENCE} NOTIFYrl";
$s .= "Event: message-summaryrl";
$s .= "Content-Type: application/simple-message-summaryrl";
$s .= "Content-Length: ${CONTENT_LENGTH}rl";
$s .= "rl";
$s .=…

Asterisk Users 3.7 years ago 2 Answers

OPTIONS support for SDP

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I have been sending OPTIONS requests 1) programatically (my own code),
2)manually via SIP VERIFY PEER x and 3)automatcially by setting verify=yes
in sip.conf. The trouble is I do not see anything except an ACK 200 come
back from endpoints and it does not contain any SDP/codec info. . My goal is
to determine audio and video codec capability in advance of a call INVITE. I
notice in both 2 and 3 examples the Asterisk generated OPTIONS does not
specify any ACCEPT header (ie Accept=application/sdp). I was thinking maybe
that is why I…

Asterisk Users 3.8 years ago 0 Answers

OPTIONS to determine codec capability before an INVITE

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I have been sending OPTIONS requests 1) programatically (my own code),
2)manually via SIP VERIFY PEER x and 3)automatcially by setting verify=yes
in sip.conf. The trouble is I do not see anything except an ACK 200 come
back from endpoints and it does not contain any SDP/codec info. . My goal is
to determine audio and video codec capability in advance of a call INVITE. I
notice in both 2 and 3 examples the Asterisk generated OPTIONS does not
specify any ACCEPT header (ie Accept=application/sdp). I was thinking maybe
that is why I…

Asterisk Users 3.8 years ago 0 Answers

OPTIONS to query endpoint capability

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I have been sending OPTIONS requests both programatically (my own code),
manually via SIP VERIFY PEER x and automatcially by setting verify=yes in
sip.conf. The trouble is I do not see anything except an ACK 200 come back
from endpoints and it does not contain any SDP/codec info. . My goal is to
determine audio and video codec capability in advance of a call INVITE. I
notice the Asterisk generated OPTIONS does not specify any Accept header (ie
Accept=application/sdp). I was thinking maybe that is why I don't get any
SDP coming back.…

Asterisk Users 3.8 years ago 0 Answers