* You are viewing Posts Tagged ‘call id’

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

Upgrade from version 1.6.24 to 1.8.12 – Retransmission timeout error

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 111.111.1.111 is their)

Working one with 1.6:

Audio is at 222.222.22.22 port 26002
Adding codec 0×4 (ulaw) to SDP
Adding codec 0×8 (alaw) to SDP
Reliably Transmitting (no NAT) to 111.111.1.111:5060:
INVITE sip:0000033666666666@111.111.1.111 SIP/2.0
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport
Max-Forwards: 70
From: “TOOTAi” ;tag=as52190c5c
To:
Contact:
Call-ID: 2c974a0a2b08abe320ed388433e47d7e@222.222.22.22
CSeq: 102 INVITE
User-Agent: TOOTAiAudio
Date: Sun, 27 May 2012 16:10:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 284043376 284043376 IN IP4 222.222.22.22
s=TOOTAiAudio PBX
c=IN IP4 222.222.22.22
t=0 0
m=audio 26002 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

Invite + decreasing sequence number => 500 Error?

This only applies to the same dialog. The question here is if it is the same dialog. If it is, then the server indeed has a bug. Check the Call-ID and the from tag of both requests.

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

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.

///////////////////////////////

Problem while sending SIP NOTIFY via AMI in 1.8.10-rc2

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 Manager Interface:
T 127.0.0.1:57530 -> 127.0.0.1:5038 [AP]
Action: SIPnotify.
Channel: SIP/max.
Variable: Event=check-sync;reboot=false.

Leads to:
U 192.168.10.72:25060 -> 192.168.10.72:5060
NOTIFY sip:max@192.168.10.72 SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.72:0;branch=z9hG4bK1dff6efe.
Max-Forwards: 70.
From: “asterisk” ;tag=as66766c2a.
To: .
Contact: .
Call-ID: 412a8eff76bd7ac56ac06831256fd6aa@192.168.10.72.
CSeq: 102 NOTIFY.
Subscription-State: terminated.
Event: check-sync;reboot=false.
Content-Length: 0.

The weird thing is the port number 0 in the contact header.

Is this a bug or do I something wrong?

Thanks,

Karsten

Call going into s-extension

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: 69
From: “32433885116″ ;tag=as6aa259f4
To:
Contact:
Call-ID: 092fd61b1e3aad5e670dfa94578db96d@217.111.202.80:5060
CSeq: 102 INVITE
User-Agent: TLDOSIPS
Date: Tue, 27 Dec 2011 19:26:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 296/

What is the difference ??