Reciver cant here me

Asterisk Asterisk Support
1

Hello,

I have a problem. When i calling to someone they cant here me talk but i can here them talk. What can be the problem ?

I hope someone can help me out of this problem.

sip settings info

[quote]Global Settings:

UDP Bindaddress: 91.189.125.106:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: No
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: Asterisk PBX 10.11.0
SDP Session Name: Asterisk PBX 10.11.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No

Network QoS Settings:

IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No

Network Settings:

SIP address remapping: Disabled, no localnet list
Externhost:
Externaddr: (null)
Externrefresh: 10

Global Signalling Settings:

Codecs: (gsm|ulaw|alaw|h263|testlaw)
Codec Order: ulaw:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70

Default Settings:

Allowed transports: UDP
Outbound transport: UDP
Context: default
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk[/quote]

2

Please provide the actual sip.conf and a sip debug trace containing the SDP exchange.

You haven’t even provided the device specific SIP configuration, and your printout will contain too many default settings.

3

Hi,

Here is my sip.conf

4

Still no SIP debug trace. Likely that there’s NAT involved and that’s what’s causing the one-way audio.

5

Hello, Here is all data i get from asterisk.

[quote] == Using SIP RTP CoS mark 5
– Executing [4733284007@from-didww:1] Dial(“SIP/46.19.210.14-0000000e”, “SIP/4733284007,60”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/4733284007
– SIP/4733284007-0000000f is ringing
== Spawn extension (from-didww, 4733284007, 1) exited non-zero on ‘SIP/46.19.210.14-0000000e’
== Using SIP RTP CoS mark 5
– Executing [4733284007@from-didww:1] Dial(“SIP/46.19.210.14-00000010”, “SIP/4733284007,60”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/4733284007
– SIP/4733284007-00000011 is ringing
– SIP/4733284007-00000011 answered SIP/46.19.210.14-00000010
– Locally bridging SIP/46.19.210.14-00000010 and SIP/4733284007-00000011
== Spawn extension (from-didww, 4733284007, 1) exited non-zero on ‘SIP/46.19.210.14-00000010’
[/quote]

6

try adding this lines to general section in the sip.conf

nat=yes
externip=91.189.125.106
localnet=192.168.1.0/255.255.255.0
canreinvite=nonat

  • replace 192.168.1.0 with your local network and 255.255.255.0 with your local network mask if u have a local network. externip should be your public IP to your pbx server.
7

If nat is needed, note that nat=yes is deprecated in the most recent versions. If the above works, I would then try it without nat=, as I’m not convinced that it is needed for simple inbound traversals.