No rtp transmission in asterisk until i press hold and release

When i call from my mobile to the server it works fine but on call from asterisk peer no rtp transmission occurs in asterisk server unless i send a hold request from my soft-phone ( i confirm it by rtp debugging in asterisk).
After hold , release asterisk keep transmitting rtp packet and everything works fine.

My server and peer is separated by a firewall. I checked udp dump in my server and it is receiving rtp packet from my peer though rtp debug on asterisk cli does not show anything.

In my dialplan i execute a MixMonitor command first then Dial command.

My Server: centos-release-7-3.1611.el7.centos.x86_64
**Asterisk:**Asterisk 1.8.7.1

Sip Settings:
localhost*CLI> sip show settings

Global Settings:

UDP Bindaddress: 0.0.0.0:5060
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: No
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 1.8.7.1
SDP Session Name: Asterisk PBX 1.8.7.1
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: 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: Enabled
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: 0x80000008000e (gsm|ulaw|alaw|h263|testlaw)
Codec Order: none
Relax DTMF: Yes
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 60
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
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: bogus
Force rport: No
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk

Realtime SIP Settings:

Realtime Peers: Yes
Realtime Regs: No
Cache Friends: Yes
Update: Yes
Ignore Reg. Expire: No
Save sys. name: Yes
Auto Clear: 120 (Disabled)


Peer settings:

name:3001
host:dynamic
type:peer
dtmfmode:rfc2833
directmedia:yes
nat:yes
disallow:all
allow: alaw,ulaw,gsm,g729,ilbc
qualify:no
call-limit:1
dynamic:yes