[Asterisk-Users] Grandstream and CallerID not working

[Andrew Joakimsen]

I am having similar issues, except that I get the phones extension when
it its called, I tried to set the caller id number, and asterisk
recognizes the callers number, as well as defines it, it just does not
end up on the phones display.

    -- Executing SetCallerID("SIP/-08114498", "3057400221") in new stack
    -- Executing Dial("SIP/-0811e340", "SIP/318|30|Ttm") in new stack
We're at 64.36.104.205 port 6052
Answering with capability 2
Answering with capability 4
Answering with capability 8
Answering with non-codec capability 1
11 headers, 11 lines
Reliably Transmitting:
INVITE sip:318 at 64.36.104.203 SIP/2.0
Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK34720c6f
From: "3057400221" <sip:318 at 64.36.104.205>;tag=as478b8300
To: <sip:318 at 64.36.104.203>
Contact: <sip:318 at 64.36.104.205>
Call-ID: 61cc14e70e829f9233099b292e46d2e3 at 64.36.104.205
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 16316 16316 IN IP4 64.36.104.205
s=session
c=IN IP4 64.36.104.205
t=0 0
m=audio 6052 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
 (no NAT) to 64.36.104.203:5060
    -- Called 318
    -- Started music on hold, class 'default', on SIP/-0811e340
Sip read:
SIP/2.0 100 trying
Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK34720c6f
From: "3057400221" <sip:318 at 64.36.104.205>;tag=as478b8300
To: <sip:318 at 64.36.104.203>
Call-ID: 61cc14e70e829f9233099b292e46d2e3 at 64.36.104.205
CSeq: 102 INVITE
User-Agent: Grandstream SIP UA 1.0.3.81
Content-Length: 0


8 headers, 0 lines
Sip read:
SIP/2.0 180 ringing
Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK34720c6f
From: "3057400221" <sip:318 at 64.36.104.205>;tag=as478b8300
To: <sip:318 at 64.36.104.203>
Call-ID: 61cc14e70e829f9233099b292e46d2e3 at 64.36.104.205
CSeq: 102 INVITE
User-Agent: Grandstream SIP UA 1.0.3.81
Content-Length: 0


8 headers, 0 lines
    -- SIP/318-2600 is ringing

*CLI>
*CLI>
*CLI>


-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of John Brown
Sent: Sunday, August 24, 2003 12:49 AM
To: asterisk-users at lists.digium.com
Cc: w_w_zhang at yahoo.com
Subject: Re: [Asterisk-Users] Grandstream and CallerID not working

numeric

${CALLERIDNUM}


On Sat, Aug 23, 2003 at 09:38:22PM -0700, William Zhang wrote:
> Are those caller ID numeric or have some alpha characters? GS LCD can
> display only some of those characters.
> 
> --- John Brown <jmbrown at chagresventures.com> wrote:
> > I have the following:
> > 
> > Call -> PSTN -> * -> GrandStream 101  1.0.3.81
> > 
> > The GS displays  "ohn ro n2600"  when the call
> > is past to the GS.
> > 
> > If I pass the call to a XTEN client, Caller ID
> > shows up.
> > 
> > 
> > Any ideas ??
> > 
> > 
> > 
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> =====
> 
> William Zhang
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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