Re: [asterisk-users] A bit OT - Configure GoIP for Asterisk
On Monday 02 October 2017 at 20:58:33, Steve Edwards wrote: > I recently received a GoIP-32 for a client project -- primarily outbound > calling. > > How should a GoIP be configured for Asterisk? Have you tried http://www.hybertone.com/en/solutionsClass.asp?Id=78 Antony. -- Police have found a cartoonist dead in his house. They say that details are currently sketchy. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connect Two Existing Channels and Stop Listening
Hello all, In my scenario I have two channels connected to Asterisk and in a stasis app. I can put them both in a bridge and audio between them works as expected. However, I would like to free up the resource and no longer have Asterisk involved in the call if possible. I'm currently playing with "redirect" with and without creating the bridge but not have any success. Do you have advice on how to accomplish this? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP add header not working
Thanks all for the help, I got a step ahead. But in this scenario I am not able to deliver call-id of call-leg a to call-leg b. Extension A is going to make an outbound trunk call: 1. extension calls asterisk (call leg a, call-id 1234567890) 2. asterisk makes outbound trunk call (call leg b, call-id abc123def456) Goal: Put call-id of call leg a in a separate header of call leg b or vice versa (I guess that makes more sense because I have more contorl of the headers inside of my pbx, right?). Following your suggestions I have now call-id b in a separate header of the same call-leg, but in call-leg a. regards, andre -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A bit OT - Configure GoIP for Asterisk
I recently received a GoIP-32 for a client project -- primarily outbound calling. How should a GoIP be configured for Asterisk? No fancy shmancy Elastix or FPBX GUI -- just using the configuration files. Single Server Mode, Config By Line, and Trunk Gateway Mode all seem likely suspects. How did you configure your GoIP and why? What do your relevant sip.conf section(s) look like? What does your dial command look like? So far, all I've got out of it is a 503 Declined. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP add header not working
Andre
For this to work we have had to go to using the b() option in the dial
legs for the calls that are pasting up.
You call a context that gets run before the calls are made on each
channel. This allows you to add headers to the new pjsip channels.
It works well. You can also set variables with the _ option to trigger
which headers you want to add..
The example below would add "ThisHeader", "ThatHeader" and "Call-Info" to
the new channel created in the dial. You could use combinations of other
variables and augment these methods to meet almost any need.
Exp
[OutboundDial]
exten => _XX,1,NoOp(Dial Exp)
exten => _XX,n,Set(_var1setinparrent=1) ;;Set Variable so that
when you call the b() option context in your dial the first header is
added
exten => _XX,n,Set(_var2setinparrent=1) ;;Set Variable so that
when you call the b() option context in your dial the second header is
added
exten => _XX,n,Set(_varAddSessionInparrent=1) ;;Set Variable so
that when you call the b() option context in your dial the second header is
added
exten =>
_XX,n,Dial(pjsip/333222@vendortrunk,b(AddpjsipHeaders^s^1))
[AddpjsipHeaders]
exten =>s,1,Gosubif({"$[var1setinparrent}}"="1"]?ThisHeader,1)
exten =>s,n,Gosubif({"$[var2setinparrent}}"="1"]?ThatHeader,1)
exten
=>s,n,Gosubif({"$[varAddSessionInparrent}}"="1"]?addSessionCallInfo,1)
exten => ThisHeader,1,Set(PJSIP_HEADER(add,ThisHeader)=ValueToSet)
exten => ThisHeader,n,Return()
exten => ThatHeader,1,Set(PJSIP_HEADER(add,ThatHeader)=ValuetoSet)
exten => ThatHeader,n,Return()
exten =>
addSessionCallInfo,1,Set(PJSIP_HEADER(add,Call-Info)=\;answ
er-after=0)
exten => addSessionCallInfo,n,Return()
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
From: "Andre Gronwald"
Sent: Monday, October 2, 2017 11:07 AM
To: "asterisk-users"
Subject: [asterisk-users] PJSIP add header not working
Hi,
I am trying to add a custom header to my calls to map several call-legs
into a global call for viewing.
For this to work I read the call-id from pjsip-channel and write it into
X-CID:
##
-- Executing [s@macro-dialout-trunk-predial-hook:4]
Set("PJSIP/10-0006",
"pjsipCallId=313530363933383438363436353930-1gh0bjceo933") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:5]
Set("PJSIP/10-0006",
"PJSIP_HEADER(add,X-CID)=313530363933383438363436353930-1gh0bjceo933") in
new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("PJSIP/10-0006",
"0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("PJSIP/10-0006",
"1?Set(CONNECTEDLINE(num,i)=0xx)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("PJSIP/10-0006",
"1?Set(CONNECTEDLINE(name,i)=CID:3x)") in new stack
-- Executing [s@macro-dialout-trunk:21] ExecIf("PJSIP/10-0006",
"0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)3x)") in new stack
-- Executing [s@macro-dialout-trunk:22] GotoIf("PJSIP/10-0006",
"0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:23] Dial("PJSIP/10-0006",
"PJSIP/0xx@3x,300,T") in new stack
-- Called PJSIP/0xx@3x
<--- Transmitting SIP request (991 bytes) to UDP:217.23.24.100:5060 --->
INVITE sip:0xxx...@sip.provid.er:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.253.185:15070;rport;branch=z9hG4bKPj453d15e0-de58-4945-8b95-d05b16b9
e4c3
From:
;tag=080788ac-7c10-4cf3-86b3-359764ffb5a2
To:
Contact:
Call-ID: de41b93b-51d8-44b5-9c34-f2c0928192b0
CSeq: 1519 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-14.0.1.10(14.6.2)
Content-Type: application/sdp
Content-Length: 308
v=0
o=- 1719768133 1719768133 IN IP4 192.168.253.185
s=Asterisk
c=IN IP4 192.168.253.185
t=0 0
m=audio 55112 RTP/AVP 107 9 8 3 101
a=rtpmap:107 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
<--- Received SIP response (559 bytes) from UDP:217.23.24.100:5060 --->
[...]
##
But I can't see that header anywhere in my call-legs. What am I missing?
kind regards,
andre
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Re: [asterisk-users] PJSIP add header not working
On Mon, Oct 2, 2017, at 12:06 PM, Andre Gronwald wrote:
> Hi,
> I am trying to add a custom header to my calls to map several call-legs
> into a global call for viewing.
>
> For this to work I read the call-id from pjsip-channel and write it into
> X-CID:
>
> ##
> -- Executing [s@macro-dialout-trunk-predial-hook:4]
> Set("PJSIP/10-0006",
> "pjsipCallId=313530363933383438363436353930-1gh0bjceo933") in new stack
> -- Executing [s@macro-dialout-trunk-predial-hook:5]
> Set("PJSIP/10-0006",
> "PJSIP_HEADER(add,X-CID)=313530363933383438363436353930-1gh0bjceo933")
> in new stack
> -- Executing [s@macro-dialout-trunk:18] GotoIf("PJSIP/10-0006",
> "0?bypass,1") in new stack
> -- Executing [s@macro-dialout-trunk:19] ExecIf("PJSIP/10-0006",
> "1?Set(CONNECTEDLINE(num,i)=0xx)") in new stack
> -- Executing [s@macro-dialout-trunk:20] ExecIf("PJSIP/10-0006",
> "1?Set(CONNECTEDLINE(name,i)=CID:3x)") in new stack
> -- Executing [s@macro-dialout-trunk:21] ExecIf("PJSIP/10-0006",
> "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)3x)") in new stack
> -- Executing [s@macro-dialout-trunk:22] GotoIf("PJSIP/10-0006",
> "0?customtrunk") in new stack
> -- Executing [s@macro-dialout-trunk:23] Dial("PJSIP/10-0006",
> "PJSIP/0xx@3x,300,T") in new stack
> -- Called PJSIP/0xx@3x
The PJSIP_HEADER dialplan function operates on the channel it is invoked
on. In this case you are using it on the caller, not the called party.
The wiki documentation[1] includes an example of how to apply it to an
outgoing call.
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_PJSIP_HEADER
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] PJSIP add header not working
Hi,
Following some new behaviour on PJSIP, adding SIP header must be done using
a subrouting.
Please find below my working configuration:
*[subroutine]exten => caller_handler,1,NoOp()same
=>n,Set(PJSIP_HEADER(add,X-CID)=${ARG1})same => n,Return()*
and then, add new parameters on Dial command: *same
=>n,Dial(PJSIP/${EXTEN}@,,b(subroutine^caller_handler^1(${SIPCALLID})))*
The first "*b*" before parenthesis gave direction (if header must be added
to caller or callee). More information on
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_HEADER
.
Regards.
2017-10-02 17:06 GMT+02:00 Andre Gronwald :
> Hi,
> I am trying to add a custom header to my calls to map several call-legs
> into a global call for viewing.
>
> For this to work I read the call-id from pjsip-channel and write it into
> X-CID:
>
> ##
> -- Executing [s@macro-dialout-trunk-predial-hook:4]
> Set("PJSIP/10-0006",
> "pjsipCallId=313530363933383438363436353930-1gh0bjceo933")
> in new stack
> -- Executing [s@macro-dialout-trunk-predial-hook:5]
> Set("PJSIP/10-0006", "PJSIP_HEADER(add,X-CID)=
> 313530363933383438363436353930-1gh0bjceo933") in new stack
> -- Executing [s@macro-dialout-trunk:18] GotoIf("PJSIP/10-0006",
> "0?bypass,1") in new stack
> -- Executing [s@macro-dialout-trunk:19] ExecIf("PJSIP/10-0006",
> "1?Set(CONNECTEDLINE(num,i)=0xx)") in new stack
> -- Executing [s@macro-dialout-trunk:20] ExecIf("PJSIP/10-0006",
> "1?Set(CONNECTEDLINE(name,i)=CID:3x)") in new stack
> -- Executing [s@macro-dialout-trunk:21] ExecIf("PJSIP/10-0006",
> "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)3x)") in new stack
> -- Executing [s@macro-dialout-trunk:22] GotoIf("PJSIP/10-0006",
> "0?customtrunk") in new stack
> -- Executing [s@macro-dialout-trunk:23] Dial("PJSIP/10-0006",
> "PJSIP/0xx@3x,300,T") in new stack
> -- Called PJSIP/0xx@3x
> <--- Transmitting SIP request (991 bytes) to UDP:217.23.24.100:5060 --->
> INVITE sip:0xxx...@sip.provid.er:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.253.185:15070;rport;
> branch=z9hG4bKPj453d15e0-de58-4945-8b95-d05b16b9e4c3
> From:
> ;tag=080788ac-7c10-4cf3-86b3-359764ffb5a2
>
> To:
>
> Contact:
>
> Call-ID: de41b93b-51d8-44b5-9c34-f2c0928192b0
> CSeq: 1519 INVITE
> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
> UPDATE, PRACK, REGISTER, MESSAGE, REFER
> Supported: 100rel, timer, replaces, norefersub
> Session-Expires: 1800
> Min-SE: 90
> Max-Forwards: 70
> User-Agent: FPBX-14.0.1.10(14.6.2)
> Content-Type: application/sdp
> Content-Length: 308
>
> v=0
> o=- 1719768133 1719768133 IN IP4 192.168.253.185
> s=Asterisk
> c=IN IP4 192.168.253.185
> t=0 0
> m=audio 55112 RTP/AVP 107 9 8 3 101
> a=rtpmap:107 opus/48000/2
> a=rtpmap:9 G722/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:20
> a=sendrecv
>
> <--- Received SIP response (559 bytes) from UDP:217.23.24.100:5060 --->
> [...]
>
> ##
>
>
>
>
> But I can't see that header anywhere in my call-legs. What am I missing?
>
>
> kind regards,
> andre
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
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Tel : 0975181257 - Fax : 04.816.801.14
Email : loic.chab...@voxity.fr
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inkedIn
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[asterisk-users] PJSIP add header not working
Hi,
I am trying to add a custom header to my calls to map several call-legs
into a global call for viewing.
For this to work I read the call-id from pjsip-channel and write it into
X-CID:
##
-- Executing [s@macro-dialout-trunk-predial-hook:4]
Set("PJSIP/10-0006",
"pjsipCallId=313530363933383438363436353930-1gh0bjceo933") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:5]
Set("PJSIP/10-0006",
"PJSIP_HEADER(add,X-CID)=313530363933383438363436353930-1gh0bjceo933")
in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("PJSIP/10-0006",
"0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("PJSIP/10-0006",
"1?Set(CONNECTEDLINE(num,i)=0xx)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("PJSIP/10-0006",
"1?Set(CONNECTEDLINE(name,i)=CID:3x)") in new stack
-- Executing [s@macro-dialout-trunk:21] ExecIf("PJSIP/10-0006",
"0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)3x)") in new stack
-- Executing [s@macro-dialout-trunk:22] GotoIf("PJSIP/10-0006",
"0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:23] Dial("PJSIP/10-0006",
"PJSIP/0xx@3x,300,T") in new stack
-- Called PJSIP/0xx@3x
<--- Transmitting SIP request (991 bytes) to UDP:217.23.24.100:5060 --->
INVITE sip:0xxx...@sip.provid.er:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.253.185:15070;rport;branch=z9hG4bKPj453d15e0-de58-4945-8b95-d05b16b9e4c3
From:
;tag=080788ac-7c10-4cf3-86b3-359764ffb5a2
To:
Contact:
Call-ID: de41b93b-51d8-44b5-9c34-f2c0928192b0
CSeq: 1519 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-14.0.1.10(14.6.2)
Content-Type: application/sdp
Content-Length: 308
v=0
o=- 1719768133 1719768133 IN IP4 192.168.253.185
s=Asterisk
c=IN IP4 192.168.253.185
t=0 0
m=audio 55112 RTP/AVP 107 9 8 3 101
a=rtpmap:107 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
<--- Received SIP response (559 bytes) from UDP:217.23.24.100:5060 --->
[...]
##
But I can't see that header anywhere in my call-legs. What am I missing?
kind regards,
andre
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
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Re: [asterisk-users] Gerrit usage?
On Fri, Sep 29, 2017 at 12:27:53PM -0300, Joshua Colp wrote: > > "git checkout -b 13" appears to fix this. > > This did not create a branch from 13. This created a branch named "13" > from the branch you were on, which was most likely master. That is why > your "git review" is not working as you expect, because you are telling > it that you did the work against "13" but it really was against master. > > git checkout -b 13 origin/13 > > Would create a local branch "13" which is from the remote branch "13". > You'll need to do this, or do your "git review" against master and then > cherry pick from inside Gerrit to the appropriate branches. Thank you for your near instant feedback, this fixed my problem and I was able to submit code for a review. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
