[Asterisk-Users] Grandstream and CallerID not working

[John Brown]

Yup, pretty much the same SIP flow I have.  If I send this to a XTEN
client life is happy


On Sun, Aug 24, 2003 at 01:46:33AM -0400, Andrew Joakimsen wrote:
> 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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> 
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