Can not hear each other from sip phone!

Asterisk Asterisk Support
1

hello, asteriskers:
i make a call between the two xlite, the two phones are ring, but we can not hear each other. anybody knows that issue? please give me a hand!
thanks!
zhu8080

2

Explain a bit more on how your environment look like.

Is the x-lite’s behind different firewalls ?

Is your asterisk behind a NAT and have you configured sip.conf accordigly ?

Open the debug in the x-lite, what do you see!

Is there any errors in the asterisk log/console ?

Othervise is the answer, you have done something wrong :wink:

/Mats

3

hello, friends:
the two xlite phones are in LAN. there is no NAT in my environment. no error
comes out. i am also confusing it!
zhu8080

4

you need to start giving more information and log outputs. are the machines running X-Lite also running a personal firewall ? do you need to allow UDP ports that correspond to your rtp.conf settings on these firewalls ?

5

BYE sip:800@192.168.2.138 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.159:26461;branch=z9hG4bK-d87543-b943b607e119db10-1–d87543-;rport
Max-Forwards: 70
Contact: sip:900@192.168.2.159:26461
To: "800"sip:800@192.168.2.138;tag=as3026466a
From: “900"sip:900@192.168.2.138;tag=7742f13c
Call-ID: NmRjMmM4MTBjMzY0NzNmN2Q0ZjJjOTAyMDQzNDgzYzk.
CSeq: 3 BYE
Proxy-Authorization: Digest username=“900”,realm=“asterisk”,nonce=“78e11eaa”,uri="sip:800@192.168.2.138”,response=“8fed91c57166f149d27ce4d6139cd35e”,algorithm=MD5
User-Agent: X-Lite release 1006e stamp 34025
Reason: SIP;description="User Hung Up"
Content-Length: 0

— (12 headers 0 lines) —
Sending to 192.168.2.159 : 26461 (NAT)
Transmitting (NAT) to 192.168.2.159:26461:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.159:26461;branch=z9hG4bK-d87543-b943b607e119db10-1–d87543-;rport;received=192.168.2.159
From: "900"sip:900@192.168.2.138;tag=7742f13c
To: "800"sip:800@192.168.2.138;tag=as3026466a
Call-ID: NmRjMmM4MTBjMzY0NzNmN2Q0ZjJjOTAyMDQzNDgzYzk.
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:800@192.168.2.138
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing

set_destination: Parsing sip:800@192.168.2.155:39299;rinstance=c4e6dad35f881257 for address/port to send to
set_destination: set destination to 192.168.2.155, port 39299
We’re at 192.168.2.138 port 18222
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 192.168.2.155:39299:
INVITE sip:800@192.168.2.155:39299;rinstance=c4e6dad35f881257 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.138:5060;branch=z9hG4bK28dbb76d
From: “900” sip:800@192.168.2.138;tag=as25206d58
To: sip:800@192.168.2.155:39299;rinstance=c4e6dad35f881257;tag=6b74074e
Contact: sip:800@192.168.2.138
Call-ID: 52a06b602d023c7a7ad6ed4417440c0e@192.168.2.138
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp

6

instead of posting partial sip debugs, why not turn on debug logging in logger.conf and post that for a complete call ?

are you going to answer the question about the firewall ?

7

baconbuttie,
I’m having a similar problem but only in calls using my VoiP provider, not between internal extensions. The solution for the lack of sound was using canreinvite=no. I’m using version 1.4.1 and the SVN 1.4 branch.
My firewall is forwarding UDP 5060 and 10000-20000 to my Asterisk box.
Do you have an idea of what is happening here? I have an AsteriskNow appliance that doesn’t have this problem.