User=phone in FROM header field of uri-parameter of SIP URL

Asterisk Asterisk General
1

Version: Asterisk 1.8
Trunk: SIP Trunk with provider

I had created a trunk to over service provider who is providing SIP trunk. They need to make a call to their server with user=phone in FROM field of SIP URL.

I added usereqphone=yes in peer details. It provides user=phone in TO field, but not in FROM.

Any body can help me out in this case ?

My trunk details.

Attached sip and rtp debug details.

SIP trunk Peer details;

host=X.X.X.X
port=5060
type=peer
qualify=yes
nat=yes
insecure=invite,port
dtmfmode=rfc2833
disallow=all
canreinvite=no
allow=alaw
usereqphone=yes

  • Name : sip-trunk
    Secret :
    MD5Secret :
    Remote Secret:
    Context : from-trunk-sip-mobily
    Subscr.Cont. :
    Language :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    MOH Suggest :
    Mailbox :
    VM Extension : *97
    LastMsgsSent : 32767/65535
    Call limit : 0
    Max forwards : 0
    Dynamic : No
    Callerid : “” <>
    MaxCallBR : 384 kbps
    Expire : -1
    Insecure : port,invite
    Force rport : Yes
    ACL : No
    DirectMedACL : No
    T.38 support : No
    T.38 EC mode : Unknown
    T.38 MaxDtgrm: -1
    DirectMedia : No
    PromiscRedir : No
    User=Phone : Yes
    Video Support: Yes
    Text Support : No
    Ign SDP ver : No
    Trust RPID : Yes
    Send RPID : Yes
    Subscriptions: Yes
    Overlap dial : Yes
    DTMFmode : rfc2833
    Timer T1 : 500
    Timer B : 32000
    ToHost : X.X.X.X
    Addr->IP : X.X.X.X:5060
    Defaddr->IP : (null)
    Prim.Transp. : UDP
    Allowed.Trsp : UDP
    Def. Username:
    SIP Options : (none)
    Codecs : 0x8 (alaw)
    Codec Order : (alaw:20)
    Auto-Framing : No
    Status : UNREACHABLE
    Useragent :
    Reg. Contact :
    Qualify Freq : 60000 ms
    Sess-Timers : Accept
    Sess-Refresh : uas
    Sess-Expires : 1800 secs
    Min-Sess : 90 secs
    RTP Engine : asterisk
    Parkinglot :
    Use Reason : No
    Encryption : No
2

Can any body help in this case,

SIP Provider where using SIP in 3GPP !!

3

I have hit the same problem here, with user=phone being added to the invite sip uri, the TO sip header but not the FROM sip header.

Did you, or anyone, have a pointer as to what causes this behaviour, and how to fix it?

I can see an older request with the same problem, but no answer, so hoping that I can get a bit more success. See User=phone in FROM header field of uri-parameter of SIP URL

Thanks in advance