Re: [Asterisk-Users] Grandstream and CallerID
On Monday 20 December 2004 20:24, David Ishmael wrote: Jon, I went in and setup my extensions.conf just like you stated, and I can see in the incoming call number in the log file but the phone does not display the number (it's just showing the phone's extension number). Can you post your sip.conf context for the GS so I can compare? sip.conf [2002] type=friend host=dynamic username=2002 dtmfmode=inband mailbox=2002 context=remote callerid=Name 2002 canreinvite=no secret=xx Jon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream and CallerID
On Monday 20 December 2004 14:59, David Ishmael wrote:
I'm having similar problems with my Grandstream BT-100 SIP phone. I've
removed the fromuser=1234 from the sip.conf file but the phone still shows
1234 in the display when getting a call. I can see the incoming PSTN CID
in the log file but for some reason its not passing this to the phone. The
CID looks something like:
Joe Somebody 7035551212
Others have stated that the BT-100 can't take characters, only numbers so I
would assume there's a function like SetCIDNum(${CALLERID}) to extract the
number and send it to the BT-100. Can anyone that has the CallerID working
post their setup/configs so I can see what I'm doing wrong?
I've used SetCallerID(${CALLERIDNUM}) with /gS phones and they display CID
correctly.
ie:
exten = 2000,1,SetCallerID(${CALLERIDNUM})
exten = 2000,2,Dial(SIP/2000,30,Tt)
Jon
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RE: [Asterisk-Users] Grandstream and CallerID
Jon,
I went in and setup my extensions.conf just like you stated, and I can see
in the incoming call number in the log file but the phone does not display
the number (it's just showing the phone's extension number). Can you post
your sip.conf context for the GS so I can compare?
-Dave
-Original Message-
From: Jon Lawrence [mailto:[EMAIL PROTECTED]
Sent: Monday, December 20, 2004 9:41 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Grandstream and CallerID
On Monday 20 December 2004 14:59, David Ishmael wrote:
I'm having similar problems with my Grandstream BT-100 SIP phone. I've
removed the fromuser=1234 from the sip.conf file but the phone still shows
1234 in the display when getting a call. I can see the incoming PSTN CID
in the log file but for some reason its not passing this to the phone.
The
CID looks something like:
Joe Somebody 7035551212
Others have stated that the BT-100 can't take characters, only numbers so
I
would assume there's a function like SetCIDNum(${CALLERID}) to extract the
number and send it to the BT-100. Can anyone that has the CallerID
working
post their setup/configs so I can see what I'm doing wrong?
I've used SetCallerID(${CALLERIDNUM}) with /gS phones and they display CID
correctly.
ie:
exten = 2000,1,SetCallerID(${CALLERIDNUM})
exten = 2000,2,Dial(SIP/2000,30,Tt)
Jon
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RE: [Asterisk-Users] Grandstream and CallerID
We recently went through this pain. Here is the resolution.
On the phone set the following field to no
User ID is phone number
On the PBX delete the following field in sip.conf for each extension
associated with a GS BT101 phone
Fromuser=ext#
Regards,
Goutam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David Ishmael
Sent: December 20, 2004 3:25 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Grandstream and CallerID
Jon,
I went in and setup my extensions.conf just like you stated, and I can see
in the incoming call number in the log file but the phone does not display
the number (it's just showing the phone's extension number). Can you post
your sip.conf context for the GS so I can compare?
-Dave
-Original Message-
From: Jon Lawrence [mailto:[EMAIL PROTECTED]
Sent: Monday, December 20, 2004 9:41 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Grandstream and CallerID
On Monday 20 December 2004 14:59, David Ishmael wrote:
I'm having similar problems with my Grandstream BT-100 SIP phone. I've
removed the fromuser=1234 from the sip.conf file but the phone still shows
1234 in the display when getting a call. I can see the incoming PSTN CID
in the log file but for some reason its not passing this to the phone.
The
CID looks something like:
Joe Somebody 7035551212
Others have stated that the BT-100 can't take characters, only numbers so
I
would assume there's a function like SetCIDNum(${CALLERID}) to extract the
number and send it to the BT-100. Can anyone that has the CallerID
working
post their setup/configs so I can see what I'm doing wrong?
I've used SetCallerID(${CALLERIDNUM}) with /gS phones and they display CID
correctly.
ie:
exten = 2000,1,SetCallerID(${CALLERIDNUM})
exten = 2000,2,Dial(SIP/2000,30,Tt)
Jon
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RE: [Asterisk-Users] Grandstream and CallerID
I went into the phone and made sure that 'User ID is phone number' was set
to 'No' and made sure that Fromuser=ext# was not present in the sip.conf
file. When a call comes in, the log shows the incoming number but the phone
still reads the extension number. I also have the sip.conf dtmfmode=inband
and the SIP phone set to in-audio. Maybe this is a problem with the
firmware version (1.0.5.16) or my phone is broken.
-Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Goutam Shaw
Sent: Monday, December 20, 2004 2:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Grandstream and CallerID
We recently went through this pain. Here is the resolution.
On the phone set the following field to no
User ID is phone number
On the PBX delete the following field in sip.conf for each extension
associated with a GS BT101 phone
Fromuser=ext#
Regards,
Goutam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David Ishmael
Sent: December 20, 2004 3:25 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Grandstream and CallerID
Jon,
I went in and setup my extensions.conf just like you stated, and I can see
in the incoming call number in the log file but the phone does not display
the number (it's just showing the phone's extension number). Can you post
your sip.conf context for the GS so I can compare?
-Dave
-Original Message-
From: Jon Lawrence [mailto:[EMAIL PROTECTED]
Sent: Monday, December 20, 2004 9:41 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Grandstream and CallerID
On Monday 20 December 2004 14:59, David Ishmael wrote:
I'm having similar problems with my Grandstream BT-100 SIP phone. I've
removed the fromuser=1234 from the sip.conf file but the phone still shows
1234 in the display when getting a call. I can see the incoming PSTN CID
in the log file but for some reason its not passing this to the phone.
The
CID looks something like:
Joe Somebody 7035551212
Others have stated that the BT-100 can't take characters, only numbers so
I
would assume there's a function like SetCIDNum(${CALLERID}) to extract the
number and send it to the BT-100. Can anyone that has the CallerID
working
post their setup/configs so I can see what I'm doing wrong?
I've used SetCallerID(${CALLERIDNUM}) with /gS phones and they display CID
correctly.
ie:
exten = 2000,1,SetCallerID(${CALLERIDNUM})
exten = 2000,2,Dial(SIP/2000,30,Tt)
Jon
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RE: [Asterisk-Users] Grandstream and CallerID
At 04:42 PM 12/20/04, you wrote:
I went into the phone and made sure that 'User ID is phone number' was set
to 'No' and made sure that Fromuser=ext# was not present in the sip.conf
file. When a call comes in, the log shows the incoming number but the phone
still reads the extension number. I also have the sip.conf dtmfmode=inband
and the SIP phone set to in-audio. Maybe this is a problem with the
firmware version (1.0.5.16) or my phone is broken.
-Dave
did you try this in your extensions.conf
...
exten = context,6,NoOp(${CALLERID})
exten = context,7,Dial(SIP/${Ext}SIP/${Ext2},15,Ttr)
...
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[Asterisk-Users] Grandstream and CallerID
I would be grateful for any pointers in the right direction. In short, I get CallerID to display on Xten and a SipTone II; but have failed miserably to get my BudgeTone 101 to display anything other than the phone's own number. I've been through the Wiki and have followed (I think) the suggestions given in a couple of articles (e.g. Tips for Granstream Budgetone and Asterisk). Is there a particular trick I'm missing? My sip.conf contains the following: [101] type=friend username=101 fromuser=101 callerid=Study 101 host=dynamic defaultip=192.168.1.80 nat=no dtmfmode=info [EMAIL PROTECTED] disallow=all ;allow=gsm allow=ulaw allow=alawallow=ilbc I have also tried this with callerid=Study 101 and callerid=101 101. Many thanks. George ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream and CallerID
Hi George Are you using a Voicetronix card, if so the driver has not yet been updated for caller ID. They are working on it. :) Regards Doug Reid Director Stormcorp Network Solutions (Pty) Ltd Tel:+27 11 807 1141 Fax:+27 11 807 3504 Mobile: +27 83 989 0008 E-Mail: [EMAIL PROTECTED] Web:www.stormcorp.co.za --- NOTICE - This message contains privileged and confidential information intended only for the use of the addressee named above. If you are not the intended recipient of this message, you are hereby notified that you must not disseminate, copy or take any action in reliance on it. If you have received this message in error, please notify Stormcorp Network Solutions, its subsidiaries or associates, immediately. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the view of Stormcorp, its subsidiaries or associates. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of George Gardiner Sent: Wednesday, October 27, 2004 11:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Grandstream and CallerID I would be grateful for any pointers in the right direction. In short, I get CallerID to display on Xten and a SipTone II; but have failed miserably to get my BudgeTone 101 to display anything other than the phone's own number. I've been through the Wiki and have followed (I think) the suggestions given in a couple of articles (e.g. Tips for Granstream Budgetone and Asterisk). Is there a particular trick I'm missing? My sip.conf contains the following: [101] type=friend username=101 fromuser=101 callerid=Study 101 host=dynamic defaultip=192.168.1.80 nat=no dtmfmode=info [EMAIL PROTECTED] disallow=all ;allow=gsm allow=ulaw allow=alawallow=ilbc I have also tried this with callerid=Study 101 and callerid=101 101. Many thanks. George ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream and CallerID
George Gardiner wrote: I would be grateful for any pointers in the right direction. In short, I get CallerID to display on Xten and a SipTone II; but have failed miserably to get my BudgeTone 101 to display anything other than the phone's own number. The BT101 can only display callerid number. It's a number only display. begin:vcard fn:Eric Wileing n:Wileing;Eric email;internet:[EMAIL PROTECTED] tel;work:504-899-1387 x2120 x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream and CallerID
On Wed, 2004-10-27 at 08:15 -0500, Eric Wieling wrote: George Gardiner wrote: I would be grateful for any pointers in the right direction. In short, I get CallerID to display on Xten and a SipTone II; but have failed miserably to get my BudgeTone 101 to display anything other than the phone's own number. The BT101 can only display callerid number. It's a number only display. Not quite - when someone calls from out of the country (no caller ID) - then the BT100 tries to display'Trl' or something like that... -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream and CallerID
I've also seen my 101 try to display 'grandstream', but with missing letters. I think it showed gra str m once - all lowercase letters. At least the display is capable of crude letter display. Getting it to work the way you want is another matter altogether. - Original Message - From: Mark Elkins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, October 27, 2004 8:47 AM Subject: Re: [Asterisk-Users] Grandstream and CallerID On Wed, 2004-10-27 at 08:15 -0500, Eric Wieling wrote: George Gardiner wrote: I would be grateful for any pointers in the right direction. In short, I get CallerID to display on Xten and a SipTone II; but have failed miserably to get my BudgeTone 101 to display anything other than the phone's own number. The BT101 can only display callerid number. It's a number only display. Not quite - when someone calls from out of the country (no caller ID) - then the BT100 tries to display'Trl' or something like that... -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream and CallerID
We just received a couple Grandstream's and also notice some problems with the display. When dialing a number, they will display the digits dialed except that it seems to clear in the middle so if dialing 5804 it might show 5 then 8, then clear, then show the 0 and 4. I've also not been able to figure out how to dial numbers that have characters in them. For example, how would I dial the extension gonzo or for that matter, how do I dial [EMAIL PROTECTED] from the Grandstream 101? Rick Petersen VP, Engineering WhiteHorse Communications, Inc. El Paso, TX [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roger Hanson Sent: Wednesday, October 27, 2004 8:06 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Grandstream and CallerID I've also seen my 101 try to display 'grandstream', but with missing letters. I think it showed gra str m once - all lowercase letters. At least the display is capable of crude letter display. Getting it to work the way you want is another matter altogether. - Original Message - From: Mark Elkins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, October 27, 2004 8:47 AM Subject: Re: [Asterisk-Users] Grandstream and CallerID On Wed, 2004-10-27 at 08:15 -0500, Eric Wieling wrote: George Gardiner wrote: I would be grateful for any pointers in the right direction. In short, I get CallerID to display on Xten and a SipTone II; but have failed miserably to get my BudgeTone 101 to display anything other than the phone's own number. The BT101 can only display callerid number. It's a number only display. Not quite - when someone calls from out of the country (no caller ID) - then the BT100 tries to display'Trl' or something like that... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream and CallerID - sorted
Thanks to everyone for their help. I sorted out my CallerID problem - I had a stray fromuser=101 command in my sip.conf which was overwriting any CallerID info. It was a process of elimination (on my part) helped by all the comments I had back. Regards, George ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream and CallerID not working
On Saturday 23 August 2003 11:09 pm, John Brown 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 ?? Mine GS (callerid) works fine without any special configuration: extensions.conf ; Extension 202 - Grandstream exten = 202,1,Playback,transfer|skip ; Please hold while... exten = 202,2,Dial,sip/202|20|t; Ring, 20 secs max exten = 202,3,Voicemail,u202 ; Send to voicemail... exten = 202,5,Goto,s|6 ; Start over exten = 202,103,Voicemail,b202 ; (2 + 101) I'm on the phone exten = 202,104,Goto,5 ; Go to voicemail, etc. sip.conf [general] port = 5060 ; Port to bind to bindaddr = 10.125.65.10 ; Address to bind to context = default ; Default for incoming calls disallow=all allow=ulaw allow=alaw [202] type=friend ;;context=local context=routing host=dynamic insecure=yes defaultip=10.125.65.8 callerid=Steve 202 mailbox=202 dtmfmode=inband canreinvite=no ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve __ You actually need to constantly be alert and willing to handle things, or life will find a way to get you good! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream and CallerID not working
Are those caller ID numeric or have some alpha characters? GS LCD can display only some of those characters. --- John Brown [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = William Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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 [EMAIL PROTECTED] 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 ??
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William Zhang
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RE: [Asterisk-Users] Grandstream and CallerID not working
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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK34720c6f
From: 3057400221 sip:[EMAIL PROTECTED];tag=as478b8300
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED];tag=as478b8300
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED];tag=as478b8300
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Brown
Sent: Sunday, August 24, 2003 12:49 AM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
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 [EMAIL PROTECTED] 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 ??
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William Zhang
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Re: [Asterisk-Users] Grandstream and CallerID not working
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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK34720c6f
From: 3057400221 sip:[EMAIL PROTECTED];tag=as478b8300
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED];tag=as478b8300
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED];tag=as478b8300
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Brown
Sent: Sunday, August 24, 2003 12:49 AM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
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 [EMAIL PROTECTED] 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 ??
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William Zhang
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