[asterisk-bugs] [JIRA] (ASTERISK-24569) user=phone is not added to From, Contact and Diversion header
Richard Mudgett (JIRA)
noreply at issues.asterisk.org
Thu May 21 09:31:33 CDT 2015
[ https://issues.asterisk.org/jira/browse/ASTERISK-24569?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226310#comment-226310 ]
Richard Mudgett edited comment on ASTERISK-24569 at 5/21/15 9:29 AM:
---------------------------------------------------------------------
Excellent its working fine
That provider also request for with Media Connection information
These are sip debugs from their end server
{noformat}
----------------------------------------------------------------------------Sip debugs from their end by an out going call from our asterisk server-----------------------
=== Session Description Protocol ===
Protocol Version : v=0
Session Owner/Creator and session identi : o=root 377537064 377537064 IN IP4 10.100.20.10
Session Name : s=Asterisk PBX 1.8.20.0
Session Connection Information : c=IN IP4 10.100.20.10
Time the session is active : t=0 0
Media Name : m=audio 14830 RTP/AVP 8 101
Media : audio
Port : 14830
Protocol type : RTP/AVP
Media Format : 8 = PCMA
Media Format : 101 = dynamic
Media Attribute : a=rtpmap:8 PCMA/8000
Media Attribute : a=rtpmap:101 telephone-event/8000
Media Attribute : a=fmtp:101 0-16
Media Attribute : a=ptime:20
Media Attribute : a=sendrecv
---------------------------------------------------Sip debugs from their end by an out going call from an server what they exepect-----------------------
Protocol Version : v=0
Session Owner/Creator and session identi : o=CiscoSystemsSIP-GW-UserAgent 6070 8175 IN IP4 178.16.8.7
Session Name : s=SIP Call
Session Connection Information : c=IN IP4 178.16.8.7
Time the session is active : t=0 0
Media Name : m=audio 32706 RTP/AVP 8 101
Media : audio
Port : 32706
Protocol type : RTP/AVP
Media Format : 8 = PCMA
Media Format : 101 = dynamic
Media Connection information : c=IN IP4 178.16.8.7 <- with Media Connection information
Media Attribute : a=rtpmap:8 PCMA/8000
Media Attribute : a=rtpmap:101 telephone-event/8000
Media Attribute : a=fmtp:101 0-15
Media Attribute : a=ptime:20
------------------------------------------------------------------------------------------------------------
{noformat}
Sip debug from our server as follows
{noformat}
NVITE sip:0553625772 at 172.16.84.6;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.100.20.10:5060;branch=z9hG4bK3b61bdfd
Max-Forwards: 70
From: <sip:576981000 at 10.100.20.10;user=phone>;tag=as2f333a3c
To: <sip:0553625772 at 172.16.84.6;user=phone>
Contact: <sip:576981000 at 10.100.20.10:5060>
Call-ID: 65161339488954b43193f8562f8ac7d4 at 10.100.20.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Wed, 21 Jan 2015 09:32:12 GMT
Session-Expires: 1800
Min-SE: 1800
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Allow-Events: telephone-event
Content-Disposition: session;handling=required
P-Asserted-Identity: <sip:576981000 at 10.100.20.10;user=phone>
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 576210786 576210786 IN IP4 10.100.20.10
s=Asterisk PBX 1.8.20.0
c=IN IP4 10.100.20.10
t=0 0
m=audio 15996 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
{noformat}
was (Author: fas143):
Excellent its working fine
That provider also request for with Media Connection information
These are sip debugs from their end server
----------------------------------------------------------------------------Sip debugs from their end by an out going call from our asterisk server-----------------------
=== Session Description Protocol ===
Protocol Version : v=0
Session Owner/Creator and session identi : o=root 377537064 377537064 IN IP4 10.100.20.10
Session Name : s=Asterisk PBX 1.8.20.0
Session Connection Information : c=IN IP4 10.100.20.10
Time the session is active : t=0 0
Media Name : m=audio 14830 RTP/AVP 8 101
Media : audio
Port : 14830
Protocol type : RTP/AVP
Media Format : 8 = PCMA
Media Format : 101 = dynamic
Media Attribute : a=rtpmap:8 PCMA/8000
Media Attribute : a=rtpmap:101 telephone-event/8000
Media Attribute : a=fmtp:101 0-16
Media Attribute : a=ptime:20
Media Attribute : a=sendrecv
---------------------------------------------------Sip debugs from their end by an out going call from an server what they exepect-----------------------
Protocol Version : v=0
Session Owner/Creator and session identi : o=CiscoSystemsSIP-GW-UserAgent 6070 8175 IN IP4 178.16.8.7
Session Name : s=SIP Call
Session Connection Information : c=IN IP4 178.16.8.7
Time the session is active : t=0 0
Media Name : m=audio 32706 RTP/AVP 8 101
Media : audio
Port : 32706
Protocol type : RTP/AVP
Media Format : 8 = PCMA
Media Format : 101 = dynamic
Media Connection information : c=IN IP4 178.16.8.7 <- with Media Connection information
Media Attribute : a=rtpmap:8 PCMA/8000
Media Attribute : a=rtpmap:101 telephone-event/8000
Media Attribute : a=fmtp:101 0-15
Media Attribute : a=ptime:20
------------------------------------------------------------------------------------------------------------
Sip debug from our server as follows
NVITE sip:0553625772 at 172.16.84.6;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.100.20.10:5060;branch=z9hG4bK3b61bdfd
Max-Forwards: 70
From: <sip:576981000 at 10.100.20.10;user=phone>;tag=as2f333a3c
To: <sip:0553625772 at 172.16.84.6;user=phone>
Contact: <sip:576981000 at 10.100.20.10:5060>
Call-ID: 65161339488954b43193f8562f8ac7d4 at 10.100.20.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Wed, 21 Jan 2015 09:32:12 GMT
Session-Expires: 1800
Min-SE: 1800
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Allow-Events: telephone-event
Content-Disposition: session;handling=required
P-Asserted-Identity: <sip:576981000 at 10.100.20.10;user=phone>
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 576210786 576210786 IN IP4 10.100.20.10
s=Asterisk PBX 1.8.20.0
c=IN IP4 10.100.20.10
t=0 0
m=audio 15996 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
> user=phone is not added to From, Contact and Diversion header
> -------------------------------------------------------------
>
> Key: ASTERISK-24569
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-24569
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/General
> Affects Versions: 11.14.1
> Environment: cento 6.5
> Reporter: Mark Petersen
> Attachments: chan_sip.24569.2.patch
>
>
> the problem is that user=phone is not added to the From, Contact and Diversion header, but is correctly added to the INVITE and To header
> This is a major problem as our Provider is switching to a new platform where they require these header, in order for os to set the outgoing CALLERID on our trunk
> {noformat}
> [general]
> usereqphone=yes
> {noformat}
> {noformat}
> Set(CALLERID(name)=77777777);
> Set(CALLERID(num)=88888888);
> Set(CALLERID(ANI-num)=99999999);
> Set(CALLERID(rdnis)=66666666);
> Dial(SIP/55555555 at 192.168.0.2);
> {noformat}
> {noformat}
> INVITE sip:55555555 at 192.168.0.1;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1b298676;rport
> From: "77777777" <sip:88888888 at 192.168.0.1>;tag=as14ad576b
> To: <sip:55555555 at 192.168.0.2;user=phone>
> Contact: <sip:88888888 at 192.168.0.1:5060>
> Call-ID: 566480180a198f053ee9ba1016c0aef8 at 192.168.0.1
> CSeq: 102 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Diversion: <sip:66666666 at 192.168.0.1>;reason=unknown
> {noformat}
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