[asterisk-users] Remote-Party-ID party=called
Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten = 10,1,NoOp() exten = 10,n,SIPAddHeader(Remote-Party-ID: eric sip:1...@192.168.1.150;party=called ) exten = 10,n,Dial(SIP/test2) This is what the CLI shows : /[Jul 12 14:56:19] -- Executing [...@from-test:1] NoOp(SIP/test6-0094, ) in new stack [Jul 12 14:56:19] -- Executing [...@from-test:2] SIPAddHeader(SIP/test6-0094, Remote-Party-ID: eric sip:1...@192.168.1.150) in new stack [Jul 12 14:56:19] -- Executing [...@from-test:3] Dial(SIP/test6-0094, SIP/test2) in new stack/ SIP debug : /asterisk*CLI sip set debug peer test6 SIP Debugging Enabled for IP: 192.168.1.104:5063 [Jul 12 15:02:42] --- SIP read from 192.168.1.104:5063 --- INVITE sip:1...@192.168.1.150 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5063;branch=z9hG4bK-fe158095 From: test 6 sip:te...@192.168.1.150;tag=adbbedf0959298ddo3 To: sip:1...@192.168.1.150 *Remote-Party-ID: test 6 sip:te...@192.168.1.150;screen=yes;party=calling* Call-ID: fb31bee7-94a6a...@192.168.1.104 CSeq: 101 INVITE Max-Forwards: 70 Contact: test 6 sip:te...@192.168.1.104:5063 Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 397 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp/ In all the other SIP-messages there is no trace of the Remote-Party-ID header... Shouldn't there be a /*Remote-Party-ID: eric sip:1...@192.168.1.150;party=called */somewhere ?? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote-Party-ID party=called
Hello, escape the semicolons with a backslash! At least in astersik-1.6.X this works fine. I.e. replace in the SIP-Header-command all ; by \; Regards, Roger. Jonas Kellens schrieb: Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten = 10,1,NoOp() exten = 10,n,SIPAddHeader(Remote-Party-ID: eric sip:1...@192.168.1.150;party=called ) exten = 10,n,Dial(SIP/test2) This is what the CLI shows : /[Jul 12 14:56:19] -- Executing [...@from-test:1] NoOp(SIP/test6-0094, ) in new stack [Jul 12 14:56:19] -- Executing [...@from-test:2] SIPAddHeader(SIP/test6-0094, Remote-Party-ID: eric sip:1...@192.168.1.150) in new stack [Jul 12 14:56:19] -- Executing [...@from-test:3] Dial(SIP/test6-0094, SIP/test2) in new stack/ SIP debug : /asterisk*CLI sip set debug peer test6 SIP Debugging Enabled for IP: 192.168.1.104:5063 [Jul 12 15:02:42] --- SIP read from 192.168.1.104:5063 --- INVITE sip:1...@192.168.1.150 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5063;branch=z9hG4bK-fe158095 From: test 6 sip:te...@192.168.1.150;tag=adbbedf0959298ddo3 To: sip:1...@192.168.1.150 *Remote-Party-ID: test 6 sip:te...@192.168.1.150;screen=yes;party=calling* Call-ID: fb31bee7-94a6a...@192.168.1.104 CSeq: 101 INVITE Max-Forwards: 70 Contact: test 6 sip:te...@192.168.1.104:5063 Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 397 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp/ In all the other SIP-messages there is no trace of the Remote-Party-ID header... Shouldn't there be a /*Remote-Party-ID: eric sip:1...@192.168.1.150;party=called */somewhere ?? Jonas. -- Roger Schreiter Spindelberg 11 D-74354 Besigheim Tel.: +49 7143 36476 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote-Party-ID party=called
-Original Message- From: Jonas Kellens jonas.kell...@telenet.be To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Mon, Jul 12, 2010 3:09 pm Subject: [asterisk-users] Remote-Party-ID party=called Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten = 10,1,NoOp() exten = 10,n,SIPAddHeader(Remote-Party-ID: eric sip:1...@192.168.1.150;party=called ) exten = 10,n,Dial(SIP/test2) This is what the CLI shows : [Jul 12 14:56:19] -- Executing [...@from-test:1] NoOp(SIP/test6-0094, ) in new stack [Jul 12 14:56:19] -- Executing [...@from-test:2] SIPAddHeader(SIP/test6-0094, Remote-Party-ID: eric sip:1...@192.168.1.150) in new stack [Jul 12 14:56:19] -- Executing [...@from-test:3] Dial(SIP/test6-0094, SIP/test2) in new stack SIP debug : asterisk*CLI sip set debug peer test6 SIP Debugging Enabled for IP: 192.168.1.104:5063 [Jul 12 15:02:42] --- SIP read from 192.168.1.104:5063 --- INVITE sip:1...@192.168.1.150 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5063;branch=z9hG4bK-fe158095 From: test 6 sip:te...@192.168.1.150;tag=adbbedf0959298ddo3 To: sip:1...@192.168.1.150 Remote-Party-ID: test 6 sip:te...@192.168.1.150;screen=yes;party=calling Call-ID: fb31bee7-94a6a...@192.168.1.104 CSeq: 101 INVITE Max-Forwards: 70 Contact: test 6 sip:te...@192.168.1.104:5063 Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 397 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp In all the other SIP-messages there is no trace of the Remote-Party-ID header... Shouldn't there be a Remote-Party-ID: eric sip:1...@192.168.1.150;party=called somewhere ?? Jonas. -- i, I am trying to solve the same issue but can't get it work. I use Asterisk 1.6.1.20 and use a Siemens Optipoint 410 client which should capable of displaying Remote Party ID but it doesn't. Does anyone know a Softclient which is able to display Remote Party ID? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote-Party-ID party=called
Roger, your answer did resolve something : /[Jul 12 15:51:24] -- Executing [...@from-test:2] SIPAddHeader(SIP/test6-009a, Remote-Party-ID: eric sip:1...@192.168.1.150;party=called ) in new stack/ However this SIP-header is never send as a SIP-message to the phone from where I'm placing the call. The name eric is not displayed on the screen. This is a Cisco SPA 941 and supports the Remote-Party-ID. Regards, Jonas. On 07/12/2010 03:28 PM, Roger Schreiter wrote: Hello, escape the semicolons with a backslash! At least in astersik-1.6.X this works fine. I.e. replace in the SIP-Header-command all ; by \; Regards, Roger. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote-Party-ID party=called
It doesn't work for me too..
exten = 1400,1,SIPAddHeader(Remote-Party-ID: Test
sip:2...@192.168.1.150:5060\;party=called)
exten = 1400,n,Dial(SIP/${EXTEN},15)
leads to
-- Executing [1...@default:1] SIPAddHeader(SIP/1401-0159,
Remote-Party-ID: Test sip:2...@192.168.1.150:5060\;party=called) in new
stack
-- Executing [1...@default:2] Dial(SIP/1401-0159, SIP/1400,15) in
new stack
but does not show anything on the callee's screen.
-Original Message-
From: Jonas Kellens jonas.kell...@telenet.be
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Mon, Jul 12, 2010 3:58 pm
Subject: Re: [asterisk-users] Remote-Party-ID party=called
Roger,
your answer did resolve something :
[Jul 12 15:51:24] -- Executing [...@from-test:2]
SIPAddHeader(SIP/test6-009a, Remote-Party-ID: eric
sip:1...@192.168.1.150;party=called ) in new stack
However this SIP-header is never send as a SIP-message to the phone from where
I'm placing the call. The name eric is not displayed on the screen.
This is a Cisco SPA 941 and supports the Remote-Party-ID.
Regards,
Jonas.
On 07/12/2010 03:28 PM, Roger Schreiter wrote:
Hello,
escape the semicolons with a backslash! At least in astersik-1.6.X
his works fine.
I.e. replace in the SIP-Header-command all ; by \;
Regards,
oger.
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Re: [asterisk-users] Remote-Party-ID party=called
Hello, the SIP header now should be sent. What the remote device is doing with this header, or whether the syntax of the header is as the remote device expects it, is another question. You can check with sip set debug on whether the header is now sent as you expect! If it does, I cannot tell you, why your Cisco device is not displaying it. Regards, Roger. Jonas Kellens schrieb: Roger, your answer did resolve something : /[Jul 12 15:51:24] -- Executing [...@from-test:2] SIPAddHeader(SIP/test6-009a, Remote-Party-ID: eric sip:1...@192.168.1.150;party=called ) in new stack/ However this SIP-header is never send as a SIP-message to the phone from where I'm placing the call. The name eric is not displayed on the screen. This is a Cisco SPA 941 and supports the Remote-Party-ID. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote-Party-ID party=called
In my case, it shows the name eric and number 20 on the receiving
phone. As if the From-header is overwritten...
That's off course not what I'm trying to accomplish. Therefore I can use
the P-Asserted-Identity (which works well if I may add).
Jonas.
On 07/12/2010 04:05 PM, unsero...@aol.com wrote:
It doesn't work for me too..
exten = 1400,1,SIPAddHeader(Remote-Party-ID: Test
sip:2...@192.168.1.150:5060\;party=called)
exten = 1400,n,Dial(SIP/${EXTEN},15)
leads to
-- Executing [1...@default:1] SIPAddHeader(SIP/1401-0159,
Remote-Party-ID: Test sip:2...@192.168.1.150:5060\;party=called)
in new stack
-- Executing [1...@default:2] Dial(SIP/1401-0159,
SIP/1400,15) in new stack
but does not show anything on the callee's screen.
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Re: [asterisk-users] Remote-Party-ID party=called
No, the receiving side shows name and number as it should.
But as calling person I only see the number of the called person instead of
name and number.
So we seem to struggle with the same issue.
-Original Message-
From: Jonas Kellens jonas.kell...@telenet.be
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Mon, Jul 12, 2010 4:42 pm
Subject: Re: [asterisk-users] Remote-Party-ID party=called
In my case, it shows the name eric and number 20 on the receiving phone. As
if the From-header is overwritten...
That's off course not what I'm trying to accomplish. Therefore I can use the
P-Asserted-Identity (which works well if I may add).
Jonas.
On 07/12/2010 04:05 PM, unsero...@aol.com wrote:
It doesn't work for me too..
exten = 1400,1,SIPAddHeader(Remote-Party-ID: Test
sip:2...@192.168.1.150:5060\;party=called)
exten = 1400,n,Dial(SIP/${EXTEN},15)
leads to
-- Executing [1...@default:1] SIPAddHeader(SIP/1401-0159,
Remote-Party-ID: Test sip:2...@192.168.1.150:5060\;party=called) in new
stack
-- Executing [1...@default:2] Dial(SIP/1401-0159, SIP/1400,15) in
new stack
but does not show anything on the callee's screen.
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Re: [asterisk-users] Remote-Party-ID party=called
On 12 Jul 2010, at 14:09, Jonas Kellens wrote: I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. Am I missing something or is this waht CALLERID(name) and sendrpid is for?.. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote-Party-ID party=called
On 07/12/2010 05:01 PM, Steve Howes wrote: On 12 Jul 2010, at 14:09, Jonas Kellens wrote: I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. Am I missing something or is this waht CALLERID(name) and sendrpid is for?.. If I'm not mistaken, sendrpid is an option in the sip.conf file that only can be set on/off. Can it be dynamically set in the dialplan ? Does this 'sendrpid' display the name of the person I am calling on my phone (in stead of the extension I am calling) ? To be clear : when I dial 20, I want to see 'eric' on my display and not '20'. The dialed number needs to be transformed to a name without me having to use the phonebook-option of the IP-phone. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote-Party-ID party=called
On 12 Jul 2010, at 16:35, Jonas Kellens wrote: On 07/12/2010 05:01 PM, Steve Howes wrote: On 12 Jul 2010, at 14:09, Jonas Kellens wrote: I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. Am I missing something or is this waht CALLERID(name) and sendrpid is for?.. If I'm not mistaken, sendrpid is an option in the sip.conf file that only can be set on/off. Can it be dynamically set in the dialplan ? No. It merely makes it generate it from CALLERID Does this 'sendrpid' display the name of the person I am calling on my phone (in stead of the extension I am calling) ? No, that wasn't what you asked for.. display the name of the calling party on my phone To be clear : when I dial 20, I want to see 'eric' on my display and not '20'. The dialed number needs to be transformed to a name without me having to use the phonebook-option of the IP-phone. So you walled CALLED party displayed on your phone. There are a number of threads about this recently. You need asterisk trunk (soon to be 1.8). Doesn't exist in 1.6 S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote-Party-ID party=called
The only thing I read about changes in trunk is : / * The sendrpid parameter has been expanded to include the options 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID header to be sent (equivalent to setting sendrpid=yes) and setting sendrpid to 'pai' will cause P-Asserted-Identity header to be sent./ How will this show the name of the callee on the phone display ?? Isn't this the same as what I'm doing now with manually adding the SIP-header in the dialplan with the function SIPAddHeader() ?? Jonas. On 07/12/2010 05:54 PM, Steve Howes wrote: On 12 Jul 2010, at 16:35, Jonas Kellens wrote: On 07/12/2010 05:01 PM, Steve Howes wrote: On 12 Jul 2010, at 14:09, Jonas Kellens wrote: I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. Am I missing something or is this waht CALLERID(name) and sendrpid is for?.. If I'm not mistaken, sendrpid is an option in the sip.conf file that only can be set on/off. Can it be dynamically set in the dialplan ? No. It merely makes it generate it from CALLERID Does this 'sendrpid' display the name of the person I am calling on my phone (in stead of the extension I am calling) ? No, that wasn't what you asked for.. display the name of the calling party on my phone To be clear : when I dial 20, I want to see 'eric' on my display and not '20'. The dialed number needs to be transformed to a name without me having to use the phonebook-option of the IP-phone. So you walled CALLED party displayed on your phone. There are a number of threads about this recently. You need asterisk trunk (soon to be 1.8). Doesn't exist in 1.6 S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote-Party-ID party=called
On Mon, 2010-07-12 at 10:48 -0400, unsero...@aol.com wrote: No, the receiving side shows name and number as it should. But as calling person I only see the number of the called person instead of name and number. So we seem to struggle with the same issue. This is something that is not supported in any current version of Asterisk. However, a large amount of work has gone into connected party ID support which will be included in Asterisk 1.8. I expect the first beta of 1.8 to be available this month. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW -Huntsville, AL 35806 - USA jabber: rbry...@digium.com-=-skype: russell-bryant www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote-Party-ID party=called
On Mon, 2010-07-12 at 10:48 -0400, unsero...@aol.com wrote: No, the receiving side shows name and number as it should. But as calling person I only see the number of the called person instead of name and number. So we seem to struggle with the same issue. This is something that is not supported in any current version of Asterisk. However, a large amount of work has gone into connected party ID support which will be included in Asterisk 1.8. I expect the first beta of 1.8 to be available this month. Russell Bryant -- Ok, thanks a lot for clarifying. So I can stop investigating any further with 1.6. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
