Re: [asterisk-users] Update the LCD with the callee's name after dialing
Hello, using asterisk 1.4.30. No patch or anything else. I just do the following in dialplan : exten = 20,n,SIPAddHeader(Remote-Party-ID: Testing sip:2...@192.168.1.150:5060) When my Cisco calls my Grandstream, the name Testing appears on the screen of my Cisco. When my Grandstream calls my Cisco, only 20 appears. Is this where this discussion is about ? I guess my Cisco SPA 941 supports the Remote-Party-ID header and my Grandstream GXP2010 does not. Kind regards, 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] Update the LCD with the callee's name after dialing
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.
exten =
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})
Ryan
If you installed Asterisk from source you just need to patch and
recompile / install.
cd asterisk-version
patch -p1 ../asterisk-verson-called-
rpid.patch
make install
Otherwise if your using trixbox, etc you would probably want to grab
their SRPMS, add the patch to the spec file, and rebuild them. However
that is outside of the scope of this mailing list.
Ryan
Which version of Asterisk? The patches were made against the latest
releases. If you are running an earlier version you might need to
manually patch your install.
Ryan
--
Version 1.6.1.20
But it was my individual problem. Installing from scratch solved the
patching issue.
Now the application SIPCalledRPID is active and gets executed but i still
don't get the name of the called person
on my display. Maybe this is client dependent? I am using 3CX Softphone. Or
is somethins else missing?
The client needs to support the Remote-Party-ID SIP header. If you
ant to verify the header is being added run tcpdump and analyze it
ith Wireshark. I know that Polycom phones have support for this. I
ust put a modified version of the Asterisk 1.6.1 patch into
roduction for 25 Polycom phones, soon to be 150 phones. I changed the
eturn -1 to return 0 so that the call continues even if they
IPCalledRPID args are invalid.
Ryan
--
ust to make sure that we are talking about the same issue.
What I want is that when two users are registered at the same peer that
when user A calls user B user A gets the name of user B displayed on his client.
Is this what you are trying to fix with the patch?
Because from my understanding as an absolute newbie to SIP and Asterisk, the
header
should already contain the let's call it displayname and look something like
INVITE sip:2...@192.168.1.10:5060 SIP/2.0
ia: SIP/2.0/TCP
192.168.1.149:3822;branch=z9hG4bK-d8754z-9f01b74a4b708b04-1---d8754z-;rport
ax-Forwards: 70
ontact: sip:1...@192.168.1.149:3823;rinstance=8f3067c0aac0abc4;transport=TCP
o: Callee Name sip:2...@192.168.1.10:5060
rom: Caller Name sip:1...@192.168.1.10:5060;tag=cf41cd30
according to SIP rfc 3261 http://tools.ietf.org/html/rfc3261
--
_
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asterisk-users mailing list
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Re: [asterisk-users] Update the LCD with the callee's name after dialing
On Tue, Jul 6, 2010 at 10:19 AM, unsero...@aol.com wrote:
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.
exten =
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)
})
Ryan
If you installed Asterisk from source you just need to patch and
recompile / install.
cd asterisk-version
patch -p1 ../asterisk-verson-called-
rpid.patch
make install
Otherwise if your using trixbox, etc you would probably want to grab
their SRPMS, add the patch to the spec file, and rebuild them. However
that is outside of the scope of this mailing list.
Ryan
Which version of Asterisk? The patches were made against the latest
releases. If you are running an earlier version you might need to
manually patch your install.
Ryan
--
Version 1.6.1.20
But it was my individual problem. Installing from scratch solved the
patching issue.
Now the application SIPCalledRPID is active and gets executed but i still
don't get the name of the called person
on my display. Maybe this is client dependent? I am using 3CX Softphone.
Or
is somethins else missing?
The client needs to support the Remote-Party-ID SIP header. If you
want to verify the header is being added run tcpdump and analyze it
with Wireshark. I know that Polycom phones have support for this. I
just put a modified version of the Asterisk 1.6.1 patch into
production for 25 Polycom phones, soon to be 150 phones. I changed the
return -1 to return 0 so that the call continues even if they
SIPCalledRPID args are invalid.
Ryan
--
Just to make sure that we are talking about the same issue.
What I want is that when two users are registered at the same peer that
when user A calls user B user A gets the name of user B displayed on his
client.
Is this what you are trying to fix with the patch?
Because from my understanding as an absolute newbie to SIP and Asterisk, the
header
should already contain the let's call it displayname and look something
like
INVITE sip:2...@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/TCP
192.168.1.149:3822;branch=z9hG4bK-d8754z-9f01b74a4b708b04-1---d8754z-;rport
Max-Forwards: 70
Contact:
sip:1...@192.168.1.149:3823;rinstance=8f3067c0aac0abc4;transport=TCP
To: Callee Name sip:2...@192.168.1.10:5060
From: Caller Name sip:1...@192.168.1.10:5060;tag=cf41cd30
according to SIP rfc 3261 http://tools.ietf.org/html/rfc3261
Yes that is what the patch addresses. The phones will only display the
name of the called extension if Remote-Party-ID or P-Asserted-Identity
is set.
Ryan
--
_
-- 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] Update the LCD with the callee's name after dialing
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.
exten =
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)
})
Ryan
If you installed Asterisk from source you just need to patch and
recompile / install.
cd asterisk-version
patch -p1 ../asterisk-verson-called-
rpid.patch
make install
Otherwise if your using trixbox, etc you would probably want to grab
their SRPMS, add the patch to the spec file, and rebuild them. However
that is outside of the scope of this mailing list.
Ryan
Which version of Asterisk? The patches were made against the latest
releases. If you are running an earlier version you might need to
manually patch your install.
Ryan
--
Version 1.6.1.20
But it was my individual problem. Installing from scratch solved the
patching issue.
Now the application SIPCalledRPID is active and gets executed but i still
don't get the name of the called person
on my display. Maybe this is client dependent? I am using 3CX Softphone.
Or
is somethins else missing?
The client needs to support the Remote-Party-ID SIP header. If you
want to verify the header is being added run tcpdump and analyze it
with Wireshark. I know that Polycom phones have support for this. I
just put a modified version of the Asterisk 1.6.1 patch into
production for 25 Polycom phones, soon to be 150 phones. I changed the
return -1 to return 0 so that the call continues even if they
SIPCalledRPID args are invalid.
Ryan
--
Just to make sure that we are talking about the same issue.
What I want is that when two users are registered at the same peer that
when user A calls user B user A gets the name of user B displayed on his
client.
Is this what you are trying to fix with the patch?
Because from my understanding as an absolute newbie to SIP and Asterisk, the
header
should already contain the let's call it displayname and look something
like
INVITE sip:2...@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/TCP
192.168.1.149:3822;branch=z9hG4bK-d8754z-9f01b74a4b708b04-1---d8754z-;rport
Max-Forwards: 70
Contact:
sip:1...@192.168.1.149:3823;rinstance=8f3067c0aac0abc4;transport=TCP
To: Callee Name sip:2...@192.168.1.10:5060
From: Caller Name sip:1...@192.168.1.10:5060;tag=cf41cd30
according to SIP rfc 3261 http://tools.ietf.org/html/rfc3261
Yes that is what the patch addresses. The phones will only display the
name of the called extension if Remote-Party-ID or P-Asserted-Identity
is set.
Ryan
--
But if the Remote-Party-ID is set or not can only be checked by sniffing with
Wireshark or another sniffer.
It can not be checked by using sip set debug on in Asterisk. Correct?
Because there I cannot see anything added.
--
_
-- 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] Update the LCD with the callee's name after dialing
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.
exten =
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)
})
Ryan
If you installed Asterisk from source you just need to patch and
recompile / install.
cd asterisk-version
patch -p1 ../asterisk-verson-called-
rpid.patch
make install
Otherwise if your using trixbox, etc you would probably want to grab
their SRPMS, add the patch to the spec file, and rebuild them. However
that is outside of the scope of this mailing list.
Ryan
Which version of Asterisk? The patches were made against the latest
releases. If you are running an earlier version you might need to
manually patch your install.
Ryan
--
Version 1.6.1.20
But it was my individual problem. Installing from scratch solved the
patching issue.
Now the application SIPCalledRPID is active and gets executed but i still
don't get the name of the called person
on my display. Maybe this is client dependent? I am using 3CX Softphone.
Or
is somethins else missing?
The client needs to support the Remote-Party-ID SIP header. If you
want to verify the header is being added run tcpdump and analyze it
with Wireshark. I know that Polycom phones have support for this. I
just put a modified version of the Asterisk 1.6.1 patch into
production for 25 Polycom phones, soon to be 150 phones. I changed the
return -1 to return 0 so that the call continues even if they
SIPCalledRPID args are invalid.
Ryan
--
Just to make sure that we are talking about the same issue.
What I want is that when two users are registered at the same peer that
when user A calls user B user A gets the name of user B displayed on his
client.
Is this what you are trying to fix with the patch?
Because from my understanding as an absolute newbie to SIP and Asterisk, the
header
should already contain the let's call it displayname and look something
like
INVITE sip:2...@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/TCP
192.168.1.149:3822;branch=z9hG4bK-d8754z-9f01b74a4b708b04-1---d8754z-;rport
Max-Forwards: 70
Contact:
sip:1...@192.168.1.149:3823;rinstance=8f3067c0aac0abc4;transport=TCP
To: Callee Name sip:2...@192.168.1.10:5060
From: Caller Name sip:1...@192.168.1.10:5060;tag=cf41cd30
according to SIP rfc 3261 http://tools.ietf.org/html/rfc3261
Yes that is what the patch addresses. The phones will only display the
name of the called extension if Remote-Party-ID or P-Asserted-Identity
is set.
Ryan
--
But if the Remote-Party-ID is set or not can only be checked by sniffing with
Wireshark or another sniffer.
It can not be checked by using sip set debug on in Asterisk. Correct?
Because there I cannot see anything added.
--
I am sorry, my fault. It is added and I can see it in Asterisk sip debug.
But comparing the Remote-Party-ID Header of a (displayed) caller and a (not
displayed) callee looks a bit different.
Remote-Party-ID: Callee
sip:2...@192.168.1.10:5060;party=called;id-type=subscriber;screen=yes
Remote-Party-ID: Caller sip:1...@192.168.1.10;privacy=off;screen=yes
Could maybe this be the reason why it does not work for me?
Sorry if I ask stupid questions but this feature is quite important for me.
--
_
-- 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:
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Re: [asterisk-users] Update the LCD with the callee's name after dialing
The client needs to support the Remote-Party-ID SIP header. If you want to verify the header is being added run tcpdump and analyze it with Wireshark. I know that Polycom phones have support for this. I just put a modified version of the Asterisk 1.6.1 patch into production for 25 Polycom phones, soon to be 150 phones. I changed the return -1 to return 0 so that the call continues even if they SIPCalledRPID args are invalid. Ryan -- Ok, i did a sniffing and the header is added by asterisk, but marked as unrecognised header. So this means the client is not able to deal with it, correct? pHSIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.149:1245;branch=z9hG4bK-d8754z-1f09b71f6073d335-1---d8754z-;received=192.168.1.149;rport=1245 From: sip:1...@192.168.1.10:5060;tag=485db579 To: sip:2...@192.168.1.10:5060;tag=as71ddd62f Call-ID: NGJjNDEyNmQxNGUyYTI1YTdhN2MwZWQwYzVhMTA2ZjA. CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.20 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: sip:2...@192.168.1.10 Date: Sat, 03 Jul 2010 09:25:11 GMT Remote-Party-ID: Test2 sip:2...@192.168.1.10:5060;party=called;id-type=subscriber;screen=yes Expert Info (Note/Undecoded): Unrecognised SIP header (Remote-Party-ID) Message: Unrecognised SIP header (Remote-Party-ID) Severity level: Note Group: Undecoded -- _ -- 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] Update the LCD with the callee's name after dialing
On Thu, Jul 1, 2010 at 3:26 PM, unsero...@aol.com wrote:
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 6:19 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after
dialing
On Thu, Jul 1, 2010 at 11:52 AM, unsero...@aol.com wrote:
Thanks a lot.
Applying the patch gives me a
Hunk #5 failed at 9881
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:37 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after
dialing
On Thu, Jul 1, 2010 at 11:29 AM, unsero...@aol.com wrote:
Sounds great.
Could you please give me a hint how to install the patch?
Sorry for my stupid question but I'm a newbie to Asterisk.
Thanks.
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:06 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after
dialing
On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info wrote:
Ryan Wagoner wrote:
together one for 1.4 that compiles. I'll post both to the list
hopefully later today.
Thank you!
Doug
--
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.
exten =
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})
Ryan
If you installed Asterisk from source you just need to patch and
recompile / install.
cd asterisk-version
patch -p1 ../asterisk-verson-called-
rpid.patch
make install
Otherwise if your using trixbox, etc you would probably want to grab
their SRPMS, add the patch to the spec file, and rebuild them. However
that is outside of the scope of this mailing list.
Ryan
Which version of Asterisk? The patches were made against the latest
releases. If you are running an earlier version you might need to
manually patch your install.
Ryan
--
Version 1.6.1.20
But it was my individual problem. Installing from scratch solved the
patching issue.
Now the application SIPCalledRPID is active and gets executed but i still
don't get the name of the called person
on my display. Maybe this is client dependent? I am using 3CX Softphone. Or
is somethins else missing?
The client needs to support the Remote-Party-ID SIP header. If you
want to verify the header is being added run tcpdump and analyze it
with Wireshark. I know that Polycom phones have support for this. I
just put a modified version of the Asterisk 1.6.1 patch into
production for 25 Polycom phones, soon to be 150 phones. I changed the
return -1 to return 0 so that the call continues even if they
SIPCalledRPID args are invalid.
Ryan
--
_
-- 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:
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Re: [asterisk-users] Update the LCD with the callee's name after dialing
On Wed, Jun 30, 2010 at 12:10 PM, CunningPike cunningp...@gmail.com wrote: On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com wrote: Thank you Andrew, I will check it out. We are currently running 1.4. -Matt On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham lath...@gmail.com wrote: Remote Party ID in trunk, it works There are hacks for other versions. We use the patch in https://issues.asterisk.org/view.php?id=6643. Works great. CP CP, What version of Asterisk are you running. We are using 1.4. Seems like the patches are for 1.2. -Matt -- _ -- 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] Update the LCD with the callee's name after dialing
On Wed, Jun 30, 2010 at 4:26 PM, Ryan Wagoner rswago...@gmail.com wrote: On Wed, Jun 30, 2010 at 6:10 PM, CunningPike cunningp...@gmail.com wrote: On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com wrote: Thank you Andrew, I will check it out. We are currently running 1.4. -Matt On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham lath...@gmail.com wrote: Remote Party ID in trunk, it works There are hacks for other versions. We use the patch in https://issues.asterisk.org/view.php?id=6643. Works great. CP Until Asterisk 1.8 is released this looks like the easiest way to get remote party id working. I have modified the patch to work with Asterisk 1.6.2.9. I have also attached a patch against FreePBX 2.7 to add the necessary changes to the dialplan. I have verified this works on a Polycom 550. Ryan Ryan, 1.8 is going to be pretty awesome! I know some folks on 1.6.2.9 that will be interested in your patch. I hope it gets stable quick. -Matt -- _ -- 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] Update the LCD with the callee's name after dialing
CunningPike wrote: On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnellmattdarn...@gmail.com wrote: We use the patch in https://issues.asterisk.org/view.php?id=6643. Works great. There is a much newer patch for 1.4 that can be found at: https://issues.asterisk.org/view.php?id=8824 But, it won't apply cleanly on the latest 1.4 series. It's like 4 versions back. Once I get into work, I'll post the version I'm running it on. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Update the LCD with the callee's name after dialing
On Thu, Jul 1, 2010 at 5:55 AM, Doug Lytle supp...@drdos.info wrote: CunningPike wrote: On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnellmattdarn...@gmail.com wrote: We use the patch in https://issues.asterisk.org/view.php?id=6643. Works great. There is a much newer patch for 1.4 that can be found at: https://issues.asterisk.org/view.php?id=8824 But, it won't apply cleanly on the latest 1.4 series. It's like 4 versions back. Once I get into work, I'll post the version I'm running it on. Doug This is the version that went into trunk for 1.8. It should send the remote party id without dialplan changes. I had looked into using it with 1.6.1 and 1.6.2. However due to the number of changes since the patch was merged I was worried that I would introduce bugs. The previous patch is simple, but does require a one line dial plan change. On the previous patch I posted for 1.6.2 I also have a 1.6.1 version. It compiles but hasn't been tested. Let me see if I can quickly put together one for 1.4 that compiles. I'll post both to the list hopefully later today. Ryan -- _ -- 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] Update the LCD with the callee's name after dialing
Ryan Wagoner wrote: together one for 1.4 that compiles. I'll post both to the list hopefully later today. Thank you! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Update the LCD with the callee's name after dialing
On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info wrote:
Ryan Wagoner wrote:
together one for 1.4 that compiles. I'll post both to the list
hopefully later today.
Thank you!
Doug
--
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.
exten =
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})
Ryan
asterisk-1.6.1.20-called-rpid.patch
Description: Binary data
asterisk-1.4.33.1-called-rpid.patch
Description: Binary data
--
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Re: [asterisk-users] Update the LCD with the callee's name after dialing
Sounds great.
Could you please give me a hint how to install the patch?
Sorry for my stupid question but I'm a newbie to Asterisk.
Thanks.
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:06 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after
dialing
On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info wrote:
Ryan Wagoner wrote:
together one for 1.4 that compiles. I'll post both to the list
hopefully later today.
Thank you!
Doug
--
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
ompile but need to be tested to verify that they work. I have the
.6.2.9 version in production and plan to put the 1.6.1.20 version in
ometime this weekend.
In you are just using Asterisk in the dialplan you can set the called
emote party id with something like below. Otherwise check out the
revious FreePBX 2.7 patch.
exten =
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})
Ryan
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asterisk-users mailing list
o UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Update the LCD with the callee's name after dialing
On Thu, Jul 1, 2010 at 11:29 AM, unsero...@aol.com wrote:
Sounds great.
Could you please give me a hint how to install the patch?
Sorry for my stupid question but I'm a newbie to Asterisk.
Thanks.
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:06 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after
dialing
On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info wrote:
Ryan Wagoner wrote:
together one for 1.4 that compiles. I'll post both to the list
hopefully later today.
Thank you!
Doug
--
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.
exten =
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})
Ryan
If you installed Asterisk from source you just need to patch and
recompile / install.
cd asterisk-version
patch -p1 ../asterisk-verson-called-rpid.patch
make install
Otherwise if your using trixbox, etc you would probably want to grab
their SRPMS, add the patch to the spec file, and rebuild them. However
that is outside of the scope of this mailing list.
Ryan
--
_
-- 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] Update the LCD with the callee's name after dialing
Thanks a lot.
Applying the patch gives me a
Hunk #5 failed at 9881
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:37 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after
dialing
On Thu, Jul 1, 2010 at 11:29 AM, unsero...@aol.com wrote:
Sounds great.
Could you please give me a hint how to install the patch?
Sorry for my stupid question but I'm a newbie to Asterisk.
Thanks.
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:06 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after
dialing
On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info wrote:
Ryan Wagoner wrote:
together one for 1.4 that compiles. I'll post both to the list
hopefully later today.
Thank you!
Doug
--
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.
exten =
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})
Ryan
If you installed Asterisk from source you just need to patch and
ecompile / install.
cd asterisk-version
atch -p1 ../asterisk-verson-called-rpid.patch
ake install
Otherwise if your using trixbox, etc you would probably want to grab
heir SRPMS, add the patch to the spec file, and rebuild them. However
hat is outside of the scope of this mailing list.
Ryan
--
- Bandwidth and Colocation Provided by http://www.api-digital.com --
ew to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
o UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- 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] Update the LCD with the callee's name after dialing
On Thu, Jul 1, 2010 at 11:52 AM, unsero...@aol.com wrote:
Thanks a lot.
Applying the patch gives me a
Hunk #5 failed at 9881
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:37 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after
dialing
On Thu, Jul 1, 2010 at 11:29 AM, unsero...@aol.com wrote:
Sounds great.
Could you please give me a hint how to install the patch?
Sorry for my stupid question but I'm a newbie to Asterisk.
Thanks.
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:06 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after
dialing
On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info wrote:
Ryan Wagoner wrote:
together one for 1.4 that compiles. I'll post both to the list
hopefully later today.
Thank you!
Doug
--
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.
exten =
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})
Ryan
If you installed Asterisk from source you just need to patch and
recompile / install.
cd asterisk-version
patch -p1 ../asterisk-verson-called-
rpid.patch
make install
Otherwise if your using trixbox, etc you would probably want to grab
their SRPMS, add the patch to the spec file, and rebuild them. However
that is outside of the scope of this mailing list.
Ryan
Which version of Asterisk? The patches were made against the latest
releases. If you are running an earlier version you might need to
manually patch your install.
Ryan
--
_
-- 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:
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Re: [asterisk-users] Update the LCD with the callee's name after dialing
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 6:19 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after
dialing
On Thu, Jul 1, 2010 at 11:52 AM, unsero...@aol.com wrote:
Thanks a lot.
Applying the patch gives me a
Hunk #5 failed at 9881
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:37 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after
dialing
On Thu, Jul 1, 2010 at 11:29 AM, unsero...@aol.com wrote:
Sounds great.
Could you please give me a hint how to install the patch?
Sorry for my stupid question but I'm a newbie to Asterisk.
Thanks.
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:06 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after
dialing
On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info wrote:
Ryan Wagoner wrote:
together one for 1.4 that compiles. I'll post both to the list
hopefully later today.
Thank you!
Doug
--
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.
exten =
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})
Ryan
If you installed Asterisk from source you just need to patch and
recompile / install.
cd asterisk-version
patch -p1 ../asterisk-verson-called-
rpid.patch
make install
Otherwise if your using trixbox, etc you would probably want to grab
their SRPMS, add the patch to the spec file, and rebuild them. However
that is outside of the scope of this mailing list.
Ryan
Which version of Asterisk? The patches were made against the latest
eleases. If you are running an earlier version you might need to
anually patch your install.
Ryan
--
Version 1.6.1.20
But it was my individual problem. Installing from scratch solved the patching
issue.
Now the application SIPCalledRPID is active and gets executed but i still don't
get the name of the called person
on my display. Maybe this is client dependent? I am using 3CX Softphone. Or is
somethins else missing?
_
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asterisk-users mailing list
o UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Update the LCD with the callee's name after dialing
On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com wrote: Thank you Andrew, I will check it out. We are currently running 1.4. -Matt On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham lath...@gmail.com wrote: Remote Party ID in trunk, it works There are hacks for other versions. We use the patch in https://issues.asterisk.org/view.php?id=6643. Works great. CP -- _ -- 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] Update the LCD with the callee's name after dialing
On Wed, Jun 30, 2010 at 6:10 PM, CunningPike cunningp...@gmail.com wrote: On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com wrote: Thank you Andrew, I will check it out. We are currently running 1.4. -Matt On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham lath...@gmail.com wrote: Remote Party ID in trunk, it works There are hacks for other versions. We use the patch in https://issues.asterisk.org/view.php?id=6643. Works great. CP Until Asterisk 1.8 is released this looks like the easiest way to get remote party id working. I have modified the patch to work with Asterisk 1.6.2.9. I have also attached a patch against FreePBX 2.7 to add the necessary changes to the dialplan. I have verified this works on a Polycom 550. Ryan asterisk-1.6.2.9-called-rpid.patch Description: Binary data freepbx-2.7.0.8-core-called-rpid.patch Description: Binary data -- _ -- 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] Update the LCD with the callee's name after dialing
Thank you Andrew, I will check it out. We are currently running 1.4. -Matt On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham lath...@gmail.com wrote: Remote Party ID in trunk, it works There are hacks for other versions. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Mon, Jun 28, 2010 at 8:40 PM, Matt Darnell mattdarn...@gmail.com wrote: Is is possible with a Polycom phone to update the LCD with the callee's name after dialing them? When you dial ext 103 now, it says 'To:103'...would be nice if could have 'To:Dan Marino' This is the case even when you have a contact for ext 103. None of the phones I have ever tested do this, Polycom, Linksys, Cisco, Grandstream, Yealink, etc. -Matt -- _ -- 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 -- _ -- 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 -- _ -- 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] Update the LCD with the callee's name after dialing
Remote Party ID in trunk, it works There are hacks for other versions. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Mon, Jun 28, 2010 at 8:40 PM, Matt Darnell mattdarn...@gmail.com wrote: Is is possible with a Polycom phone to update the LCD with the callee's name after dialing them? When you dial ext 103 now, it says 'To:103'...would be nice if could have 'To:Dan Marino' This is the case even when you have a contact for ext 103. None of the phones I have ever tested do this, Polycom, Linksys, Cisco, Grandstream, Yealink, etc. -Matt -- _ -- 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 -- _ -- 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
