[asterisk-users] R: Asterisk and Call Hold
Hi All, I have a problem with asterisk and call hold. In the re-invite package when I take the call to the hold, the SDP value a=sendrecv is present, according to the rfc3264 the sdp value a must be mark with sendonly. I've already tried with Asterisk 1.8 and Asterisk 11, but there is the same problem. I've already read all the information about canreinvite and directmedia Can anybody help me? Thanks a lot Marco -- _ -- 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
[asterisk-users] Call Hold problem
Hello everybody, i have a problem with asterisk 1.8 and Call Hold My problem is that Asterisk don't send re-invite when i pick up the call from hold. I already insert canreinvite=no in all my sip channels, set dtmfmode=info in sip.conf and my Dial() command don't insert option like t, T, h, H, w, W or L (with multiple arguments). I already follow this discussion : http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial I run debug with asterisk, and i see that the re-invite are made by asterisk, but in the TO fields is present the local ip address and not the next hop ip. This is the log : http://pastebin.com/ARUC0j4t The asterisk IP : 87.248.56.101 The next hop IP : 87.248.56.100 Is it a bug? i'm already search on google, but i dont find anything. Let me know, if you need more information. Thanks for all Best Regards Marco -- _ -- 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
[asterisk-users] R: R: R: Asterisk and History-Info
Ok, thanks for all Best Regards Marco -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Joshua Colp Inviato: mercoledì 26 settembre 2012 19:37 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] R: R: Asterisk and History-Info Marco Colombo wrote: Hi, Hola, On my invite trace I don't have history-info. Could you explain me how do I put history-info on SIP INVITE? You can't. That specific RFC (4244) is not implemented within chan_sip. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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 -- _ -- 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
[asterisk-users] Asterisk and History-Info
Hi All, Someone can tell me if asterisk support the SIP History-Info? If it supports, how can enable it? I searched on Google, but I could not find anything... Thanks for all Best Regards MC http://www.connectmi.it/?utm_source=email-signatureutm_medium=emailutm_campaign=Email%2Bsignature http://www.connectmi.it/?utm_source=email-signatureutm_medium=emailutm_campaign=Email%2Bsignature [cid:27bf7ebe6e554490ac12463bf40df103]http://www.connectmi.it/?utm_source=email-signatureutm_medium=emailutm_campaign=Email%2Bsignature inline: connectmi.png-- _ -- 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
[asterisk-users] R: Asterisk and History-Info
Hi, Thanks for reply What do you mean with Using flat or Realtime log files? I need this line in the SIP Invite : History-Info: sip:+3906330xx...@enter.it;user=phone;cause=302;privacy=history;index=1 History-Info: sip:+3906330X@enterSIP/2.0 100 Trying how can I provide the data that you asked before? Thanks Best Regards Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Danny Nicholas Inviato: mercoledì 26 settembre 2012 17:34 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: Re: [asterisk-users] Asterisk and History-Info That may depend on the flavor of Asterisk you are using and whether you are using flat or realtime log files. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]mailto:[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Colombo Sent: Wednesday, September 26, 2012 10:33 AM To: Asterisk-Users Subject: [asterisk-users] Asterisk and History-Info Hi All, Someone can tell me if asterisk support the SIP History-Info? If it supports, how can enable it? I searched on Google, but I could not find anything... Thanks for all Best Regards MC [cid:image001.png@01CD9C0E.A3204A50]http://www.connectmi.it/?utm_source=email-signatureutm_medium=emailutm_campaign=Email%2Bsignature inline: image001.png-- _ -- 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
[asterisk-users] R: R: Asterisk and History-Info
Hi, On my invite trace I don't have history-info. Could you explain me how do I put history-info on SIP INVITE? -- Executing [+39@trunk-squire-incoming:1] Dial(SIP/trunk-squire-outcoming-0045, SIP/) in new stack == Using SIP RTP CoS mark 5 Audio is at 11186 Adding codec 14 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.20.1.2:5060: INVITE sip:@10.20.1.2;uniq=73A845E0147AC676B88F6EC07EFF8 SIP/2.0 Via: SIP/2.0/UDP yyy:5060;branch=z9hG4bK56839522;rport Max-Forwards: 70 From: +39zzz sip:+39zzz@yyy;tag=as3e3ef2cf To: sip:@10.20.1.2;uniq=73A845E0147AC676B88F6EC07EFF8 Contact: sip:+39zzz@yyy:5060 Call-ID: 4320ac5e6895a1c40e809ee973c7bed6@yyy:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 10.9.0-rc1 Date: Wed, 26 Sep 2012 18:37:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: +39zzz sip:+39zzz@yyy;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 239 Thanks a lot! Marco Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Danny Nicholas Inviato: mercoledì 26 settembre 2012 17:48 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: Re: [asterisk-users] R: Asterisk and History-Info Versions 1.8 and 11 (probably 10 as well) let you query SIP information. 1.2 and 1.4 (1.6 also I think) do not. If you are in a small environment, you can turn on SIP debug and put that in a separate log (would eat up the disk in a few days in most real environments). From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]mailto:[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Colombo Sent: Wednesday, September 26, 2012 10:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] R: Asterisk and History-Info Hi, Thanks for reply What do you mean with Using flat or Realtime log files? I need this line in the SIP Invite : History-Info: sip:+3906330xx...@enter.it;user=phone;cause=302;privacy=history;index=1 History-Info: sip:+3906330X@enterSIP/2.0 100 Trying how can I provide the data that you asked before? Thanks Best Regards Da: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]mailto:[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Danny Nicholas Inviato: mercoledì 26 settembre 2012 17:34 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: Re: [asterisk-users] Asterisk and History-Info That may depend on the flavor of Asterisk you are using and whether you are using flat or realtime log files. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]mailto:[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Colombo Sent: Wednesday, September 26, 2012 10:33 AM To: Asterisk-Users Subject: [asterisk-users] Asterisk and History-Info Hi All, Someone can tell me if asterisk support the SIP History-Info? If it supports, how can enable it? I searched on Google, but I could not find anything... Thanks for all Best Regards MC [cid:image001.png@01CD9C1A.F24F52E0]http://www.connectmi.it/?utm_source=email-signatureutm_medium=emailutm_campaign=Email%2Bsignature inline: image001.png-- _ -- 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
[asterisk-users] R: SIP CANCEL, Reason
Hi Jordan,
Thanks for all, but i found this bug in Asterisk :
https://issues.asterisk.org/jira/browse/ASTERISK-16465
Attached the patch to fix the problem, if the online site does not work.
Thanks for all
Best Regards
-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Matthew Jordan
Inviato: giovedì 20 settembre 2012 13:42
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] SIP CANCEL, Reason
- Original Message -
From: Marco Colombo mcolo...@enter.it
To: asterisk-users@lists.digium.com
Sent: Wednesday, September 19, 2012 10:51:43 AM
Subject: [asterisk-users] SIP CANCEL, Reason
Hi All!
i have a problem with asterisk 1.8.11.
I must have in the SIP cancel message, the line “Reason”
Example : Reason : SIP;cause=16;text=”Normal Call Clearing”
I have already enable “use_q850_reason=yes”, but this not work.
In my dialplan I have already add : exten =
_X.,n,Hangup(${HANGUPCAUSE})
Can anyone help me?
I don’t know what to do
The use_q850_reason settings applies globally. If you execute sip show
settings, what is the value of the Q.850 Reason header?
If you enable 'sip set debug on', what is the actual CANCEL request sent to the
UA?
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
http://digium.com http://asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
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Index: chan_sip.c
===
--- chan_sip.c (revision 280339)
+++ chan_sip.c (working copy)
@@ -12514,8 +12514,19 @@
}
reqprep(resp, p, sipmethod, seqno, newbranch);
- if (sipmethod == SIP_CANCEL p-answered_elsewhere) {
- add_header(resp, Reason, SIP;cause=200;text=\Call
completed elsewhere\);
+ if (sipmethod == SIP_CANCEL) {
+ if (p-answered_elsewhere) {
+ if (ast_test_flag(p-flags[1], SIP_PAGE2_Q850_REASON))
+ add_header(resp, Reason,
Q.850;cause=200;text=\Call completed elsewhere\);
+ else
+ add_header(resp, Reason,
SIP;cause=200;text=\Call completed elsewhere\);
+ }
+ else if (ast_test_flag(p-flags[1], SIP_PAGE2_Q850_REASON)
p-hangupcause) {
+ char buf[50];
+
+ sprintf(buf, Q.850;cause=%i, p-hangupcause 0x7f);
+ add_header(resp, Reason, buf);
+ }
}
return send_request(p, resp, reliable, seqno ? seqno : p-ocseq);
--
_
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[asterisk-users] SIP CANCEL, Reason
Hi All!
i have a problem with asterisk 1.8.11.
I must have in the SIP cancel message, the line Reason
Example : Reason : SIP;cause=16;text=Normal Call Clearing
I have already enable use_q850_reason=yes, but this not work.
In my dialplan I have already add : exten = _X.,n,Hangup(${HANGUPCAUSE})
Can anyone help me?
I don't know what to do
Thanks for all
Best Regards
--
_
-- 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
