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 3.3 years ago 0 Answers

Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error

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Hi list, we are upgrading our Asterisk production server from 1.6.24 to 1.8.12
version and face the following problem: one of our peer
(voicetrading.com) doesn't accept our calls anymore, we receive a
timeout error "Packet timed out after 32000ms with no response". Switching back to 1.6 make things working again! In sip.conf we have nat=no, peer conf is: [myPeerDef]
type=peer
host=111.111.1.111
context=honeypot
insecure=invite
directmedia=no
disallow=all
allow=ulaw,alaw
dtmfmode=inband We aren't registered, our IP is authorized by their system. Debug of sessions (222.222.22.22 is our server…

Asterisk Users 3.3 years ago 7 Answers

When CALL-ID were same , I could hijack another session

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Hello all. I want to know this issue is bug or not. My Asterisk version is 1.6.2.6.
I used "nat=yes" on sip.conf. ##################################
Issue 1. SDP session handring by Asterisk
################################## I used 2 clients , A and B. 2 UAC under another NAT. ///////////////////////////////

Asterisk Users 3.4 years ago 0 Answers

Problem while sending SIP NOTIFY via AMI in 1.8.10-rc2

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Hi, while testing asterisk 1.8.10-rc2 I stumbled across a weird behavior. I
want to notify a snom phone to reload its configuration. For this to
happen, I use the NOTIFY mechanism. I started the notify via AMI
command. Asterisk is bound to udp 25060, because all phones are
registered with a local opensips proxy which uses 5060. The expected
behavior would be:
asterisk send SIP NOTIFY to the proxy, the proxy sends it to the phone. Actually asterisk sends the packet to the proxy, but the contact header
contains something invalid (IMHO): On…

Asterisk Users 3.5 years ago 1 Answer

Call going into s-extension

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Hello list, any idea why this call goes to the extension 3292000101 : /INVITE sip:3292000101@IP.IP.IP.IP:5060 SIP/2.0
Call-ID: OTRC74RLS5C2PBYULB3HSJZBR4@IP.IP.IP.IP
CSeq: 102 INVITE
From: "32433885116" ;tag=74706
Via: SIP/2.0/UDP
IP.IP.IP.IP:5060;rport;branch=z9hG4bKcc90522a02bb6591fd8a807e66
Via: SIP/2.0/UDP
IP.IP.IP.IP:5060;branch=z9hG4bKd636e00c235e59dbb2d4a5eff83fdd96
Max-Forwards: 68
Content-Type: application/sdp
Contact:

User-agent: Vox Callcontrol
To:
Content-Length: 309
Record-Route: /
and this call goes to the s-extension : /< --- SIP read from UDP:IP.IP.IP.IP:5060 --->
INVITE sip:s@IP.IP.IP.IP:5060 SIP/2.0
Record-Route:

Via: SIP/2.0/UDP IP.IP.IP.IP;branch=z9hG4bK918e.b1b09812.0
Via: SIP/2.0/UDP
217.111.202.80:5060;received=217.111.202.80;branch=z9hG4bK5cb88830;rport=5060
Max-Forwards:…

Asterisk Users 3.7 years ago 3 Answers

Contexts and Extensions

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Hello Everyone, For inbound, I am trying to specify a specific context. Everything
works fine using the IP address, however with domain name
it's not working at all. I tried changing the: Via: SIP/2.0/UDP test.com, and the
Record-Route: If I have a peer with the host, fromdomain, and outboundprxy set as
the IP address the correct context is found "context-from-test",
but not using the domain name test.com. Asterisk still knows that the call is coming from IP address: chan_sip.c:22081 handle_request_invite: Call from ''
(192.168.2.102:5060) to extension '1001' rejected because extension
not…

Asterisk Users 3.8 years ago 0 Answers

Chan_sip How to store Register Call ID?

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Hello, I am trying to find a way to store the Register Call ID along with the peer
info, or at least extract it from a log. What can be tweak in chan_sip to
accomplish this? To illustrate, if the phone REGISTER message Call-ID
header was something like 002584a2-58e40003-5b7b478e-f56e8005@192.168.1.200,
then I would like to retrieve it somehow for that peer. I guess this might have some complications if multiple SIP clients register
from the same IP, but the contact header can be used to differenciate. I
see this is store along the Peers…

Asterisk Users 3.8 years ago 0 Answers

Tips & best practices for asterisk troubleshooting & parsing logs

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Hi Douglas, You;re right, that method is useful only for one-to-one call but as soon as
the call gets transferred etc etc as you mentioned everything will get
mixed and confusing. Any way I this can be done? Can’t a call be passed off from one channel to
> another, which would leave me with only seeing a part of the logs for the
> life of the call if I only grep the logs based on one channel id?
Yes, is the answer if you want this to implement. You need to do…

Asterisk Users 3.9 years ago 0 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 3.9 years ago 0 Answers