FreeSWITCH

SipXecs 4.4 unattended transfers to parking lot fail

Details

  • Type: Bug Bug
  • Status: Closed Closed
  • Priority: Minor Minor
  • Resolution: Fixed
  • Affects Version/s: 1.2.0
  • Fix Version/s: None
  • Component/s: freeswitch-core
  • Labels:
    None
  • Environment:
  • Platform:
    Linux x86/gcc
  • CPU Info:
    x86_64 inside vmware
  • FreeSWITCH GIT Revision:
    FreeSWITCH Version 1.2.0 (git-879a798 2012-05-24 17-27-34 +0000)
  • Reproduced with GIT HEAD?:
    FreeSWITCH Version 1.2.0 (git-879a798 2012-05-24 17-27-34 +0000)

Description

removed existing very long description and created log.txt
  1. Text File
    log.txt.txt
    24/May/12 9:01 PM
    172 kB
    Jeff Lenk
  2. Text File
    trace-debug.still-not-working.txt
    30/May/12 4:51 PM
    157 kB
    Chris Rawlings

Activity

Ascending order - Click to sort in descending order
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Chris Rawlings added a comment -
ITSP --- FS --- SipXecs --- Endpoints(polycom)
i make or receive a call
i dial transfer - blind - 11 - dial (this is auto done using EFK)
 - this puts the call on hold inside of a parking lot at 11@bluecloudconsultants.com
 - this parking lot is located on the SipXecs server
i dial *411 to pick up the parked call (this is auto done using EFK)
 - this takes the call and refer / reinvites it to the phone

everything at this point is working but i do not believe it is working properly as you will see in the trace...

now i go to place to call on park again and the call with drop - dial transfer - blind - 11 - dial


here is the trace from the console with siptrace enabled
Show
Chris Rawlings added a comment - ITSP --- FS --- SipXecs --- Endpoints(polycom) i make or receive a call i dial transfer - blind - 11 - dial (this is auto done using EFK)  - this puts the call on hold inside of a parking lot at 11@bluecloudconsultants.com  - this parking lot is located on the SipXecs server i dial *411 to pick up the parked call (this is auto done using EFK)  - this takes the call and refer / reinvites it to the phone everything at this point is working but i do not believe it is working properly as you will see in the trace... now i go to place to call on park again and the call with drop - dial transfer - blind - 11 - dial here is the trace from the console with siptrace enabled
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Chris Rawlings added a comment -
forgot to mention... inside sipx we are setup as an unmanaged gateway --- this is so that SipXecs does not anchor the media to the SipXecs server... if we setup this as a SIP Trunk in SipXecs everything works great... problem is media is then anchored to the SipXecs device which is very undesirable

Show
Chris Rawlings added a comment - forgot to mention... inside sipx we are setup as an unmanaged gateway --- this is so that SipXecs does not anchor the media to the SipXecs server... if we setup this as a SIP Trunk in SipXecs everything works great... problem is media is then anchored to the SipXecs device which is very undesirable
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Jeff Lenk added a comment -
edited for readability. Please do not add such large logs to the description. Added log.txt and deleted description.
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Jeff Lenk added a comment - edited for readability. Please do not add such large logs to the description. Added log.txt and deleted description.
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Jeff Lenk added a comment -
Please leave the default assignee or otherwise no one may look at this.
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Jeff Lenk added a comment - Please leave the default assignee or otherwise no one may look at this.
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Anthony Minessale II added a comment -
recv 1244 bytes from udp/[10.100.1.18]:5060 at 01:31:42.898171:
   ------------------------------------------------------------------------
   REFER sip:mod_sofia@192.168.1.246:5090;x-sipX-nonat SIP/2.0
   Via: SIP/2.0/UDP 10.100.1.18;branch=z9hG4bK-XX-10c6_QeSsBJWYbTYxckOfCLTqA
   Via: SIP/2.0/UDP 10.2.1.25:5060;branch=z9hG4bKe378a97aE4F9B041;received=98.237.80.232;rport=34178;id=11768-21
    From: "Chris Rawlings" <sip:201@pbx1.bluecloudconsultants.com>;tag=39C1534B-DE43B41A
    To: "Outbound Call" <sip:6107413324@192.168.1.246>;tag=rQNgcHtmHXQ3Q
   Cseq: 3 REFER
   Call-Id: 35ee8a8b-20ac-1230-e3b3-0050568b4427
   Contact: <sip:201@10.2.1.25:5060>
   User-Agent: PolycomVVX-VVX_500-UA/4.0.2.11307
   Accept-Language: en
   Refer-To: <sip:11@pbx1.bluecloudconsultants.com:5060;user=phone?X-sipX-Authidentity=%3Csip%3A201%40bluecloudconsultants.com%3Bsignature%3D4FBEE0FF%253A%253A3b72cd967ae4508c5c447c560ff7ebbf%3E&REFERENCES=35ee8a8b-20ac-1230-e3b3-0050568b4427%3Brel%3Drefer>
    Referred-By: <sip:201@pbx1.bluecloudconsultants.com>
   Proxy-Authorization: Digest username="201", realm="bluecloudconsultants.com", nonce="c724d7a23c518199581772d8d107f22b4fbee0ff", qop=auth, cnonce="8uFTlm2IugohLGY", nc=00000001, uri="sip:mod_sofia@192.168.1.246:5090;x-sipX-nonat", response="e75053770b6d6102e989e3ea3a43fb23", algorithm=MD5
    Max-Forwards: 20
   Content-Length: 0
   Date: Fri, 25 May 2012 01:31:43 GMT

This tells FS to send INVITE to sip:11@pbx1.bluecloudconsultants.com with Replaces: header

send 1534 bytes to udp/[10.100.1.18]:5060 at 01:31:42.928183:
   ------------------------------------------------------------------------
   INVITE sip:11@10.100.1.18 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.246:5090;rport;branch=z9hG4bKBDQ5etXBvX47c
   Max-Forwards: 12
   From: "Outbound Call" <sip:6107413324@192.168.1.246>;tag=t971F7UUBF48e
   To: <sip:11@10.100.1.18>
   Call-ID: 39826e1c-20ac-1230-e3b3-0050568b4427
   CSeq: 28618687 INVITE
   Contact: <sip:mod_sofia@192.168.1.246:5090>
   User-Agent: FreeSWITCH-mod_sofia/1.2.0-git-879a798 2012-05-24 17-27-34 +0000
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
    Require: replaces
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
    Referred-By: <sip:201@pbx1.bluecloudconsultants.com>
   Replaces: 5f8bdbb1-6e0204b0-e05e4227@10.2.1.25;to-tag=6CCDC365-9E51E004;from-tag=QULh83
    Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 207
   X-Sipx-Handled: X10.100.1.18-207.228.237.16
   X-sipX-Authidentity: <sip:~~id~park@bluecloudconsultants.com;signature=4FBEE0F9%3A%3A9ba582535d14bdae3ce9e8501269d2dd>

Server responds 481

recv 740 bytes from udp/[10.100.1.18]:5060 at 01:31:43.040573:
   ------------------------------------------------------------------------
   SIP/2.0 481 Transaction Does Not Exist
   Record-Route: <sip:10.100.1.18:5060;lr;sipXecs-CallDest=PARK;sipXecs-rs=%2Aauth%7E.%2Afrom%7EdDk3MUY3VVVCRjQ4ZQ%60%60.900_ntap%2Aid%7EMTE3NjgtMjk%60%21075bef198d9f88172e5dd2d00798987f;x-sipX-done>
    From: "Outbound Call" <sip:6107413324@192.168.1.246>;tag=t971F7UUBF48e
   To: <sip:11@10.100.1.18>;tag=jpEaam
   Call-Id: 39826e1c-20ac-1230-e3b3-0050568b4427
   Cseq: 28618687 INVITE
   Via: SIP/2.0/UDP 192.168.1.246:5090;rport=5090;branch=z9hG4bKBDQ5etXBvX47c
   Contact: sip:10.100.1.18:5120
   Date: Fri, 25 May 2012 01:31:44 GMT
   Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE
   User-Agent: sipXecs/4.4.0 sipXecs/park (Linux)
   Accept-Language: en
   Supported: replaces
   Content-Length: 0


So you need to go see why that server says 481 to solve your problem

Show
Anthony Minessale II added a comment - recv 1244 bytes from udp/[10.100.1.18]:5060 at 01:31:42.898171:    ------------------------------------------------------------------------    REFER sip: mod_sofia@192.168.1.246 :5090;x-sipX-nonat SIP/2.0    Via: SIP/2.0/UDP 10.100.1.18;branch=z9hG4bK-XX-10c6_QeSsBJWYbTYxckOfCLTqA    Via: SIP/2.0/UDP 10.2.1.25:5060;branch=z9hG4bKe378a97aE4F9B041;received=98.237.80.232;rport=34178;id=11768-21     From: "Chris Rawlings" <sip: 201@pbx1.bluecloudconsultants.com >;tag=39C1534B-DE43B41A     To: "Outbound Call" <sip: 6107413324@192.168.1.246 >;tag=rQNgcHtmHXQ3Q    Cseq: 3 REFER    Call-Id: 35ee8a8b-20ac-1230-e3b3-0050568b4427    Contact: <sip: 201@10.2.1.25 :5060>    User-Agent: PolycomVVX-VVX_500-UA/4.0.2.11307    Accept-Language: en    Refer-To: <sip: 11@pbx1.bluecloudconsultants.com :5060;user=phone?X-sipX-Authidentity=%3Csip%3A201%40bluecloudconsultants.com%3Bsignature%3D4FBEE0FF%253A%253A3b72cd967ae4508c5c447c560ff7ebbf%3E&REFERENCES=35ee8a8b-20ac-1230-e3b3-0050568b4427%3Brel%3Drefer>     Referred-By: <sip: 201@pbx1.bluecloudconsultants.com >    Proxy-Authorization: Digest username="201", realm="bluecloudconsultants.com", nonce="c724d7a23c518199581772d8d107f22b4fbee0ff", qop=auth, cnonce="8uFTlm2IugohLGY", nc=00000001, uri="sip: mod_sofia@192.168.1.246 :5090;x-sipX-nonat", response="e75053770b6d6102e989e3ea3a43fb23", algorithm=MD5     Max-Forwards: 20    Content-Length: 0    Date: Fri, 25 May 2012 01:31:43 GMT This tells FS to send INVITE to sip: 11@pbx1.bluecloudconsultants.com with Replaces: header send 1534 bytes to udp/[10.100.1.18]:5060 at 01:31:42.928183:    ------------------------------------------------------------------------    INVITE sip: 11@10.100.1.18 SIP/2.0    Via: SIP/2.0/UDP 192.168.1.246:5090;rport;branch=z9hG4bKBDQ5etXBvX47c    Max-Forwards: 12    From: "Outbound Call" <sip: 6107413324@192.168.1.246 >;tag=t971F7UUBF48e    To: <sip: 11@10.100.1.18 >    Call-ID: 39826e1c-20ac-1230-e3b3-0050568b4427    CSeq: 28618687 INVITE    Contact: <sip: mod_sofia@192.168.1.246 :5090>    User-Agent: FreeSWITCH-mod_sofia/1.2.0-git-879a798 2012-05-24 17-27-34 +0000    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE     Require: replaces    Supported: timer, precondition, path, replaces    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer     Referred-By: <sip: 201@pbx1.bluecloudconsultants.com >    Replaces: 5f8bdbb1-6e0204b0-e05e4227@10.2.1.25 ;to-tag=6CCDC365-9E51E004;from-tag=QULh83     Content-Type: application/sdp    Content-Disposition: session    Content-Length: 207    X-Sipx-Handled: X10.100.1.18-207.228.237.16    X-sipX-Authidentity: <sip:~~id~ park@bluecloudconsultants.com ;signature=4FBEE0F9%3A%3A9ba582535d14bdae3ce9e8501269d2dd> Server responds 481 recv 740 bytes from udp/[10.100.1.18]:5060 at 01:31:43.040573:    ------------------------------------------------------------------------    SIP/2.0 481 Transaction Does Not Exist    Record-Route: <sip:10.100.1.18:5060;lr;sipXecs-CallDest=PARK;sipXecs-rs=%2Aauth%7E.%2Afrom%7EdDk3MUY3VVVCRjQ4ZQ%60%60.900_ntap%2Aid%7EMTE3NjgtMjk%60%21075bef198d9f88172e5dd2d00798987f;x-sipX-done>     From: "Outbound Call" <sip: 6107413324@192.168.1.246 >;tag=t971F7UUBF48e    To: <sip: 11@10.100.1.18 >;tag=jpEaam    Call-Id: 39826e1c-20ac-1230-e3b3-0050568b4427    Cseq: 28618687 INVITE    Via: SIP/2.0/UDP 192.168.1.246:5090;rport=5090;branch=z9hG4bKBDQ5etXBvX47c    Contact: sip:10.100.1.18:5120    Date: Fri, 25 May 2012 01:31:44 GMT    Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE    User-Agent: sipXecs/4.4.0 sipXecs/park (Linux)    Accept-Language: en    Supported: replaces    Content-Length: 0 So you need to go see why that server says 481 to solve your problem
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Chris Rawlings added a comment -
i was working with TJ and another consultant from freeSWITCH Consulting company... they have informed me that i needed to put a bug / jira request into the FreeSWITCH system for this issue....

Ingate / Patton / Sangoma / AudioCodes all work for this issue when SipXecs is setup to use one of these devices as an unmanaged Gatway...

If TJ could post more data on what they had found when we hired them that would be great because i am not exactly sure what the conclusion was that they had come to
Show
Chris Rawlings added a comment - i was working with TJ and another consultant from freeSWITCH Consulting company... they have informed me that i needed to put a bug / jira request into the FreeSWITCH system for this issue.... Ingate / Patton / Sangoma / AudioCodes all work for this issue when SipXecs is setup to use one of these devices as an unmanaged Gatway... If TJ could post more data on what they had found when we hired them that would be great because i am not exactly sure what the conclusion was that they had come to
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Anthony Minessale II added a comment -
You need to ask sipx why its giving a 481 to this request and if its an issue I can address I will.
This is what the intention was of the detailed post.
Show
Anthony Minessale II added a comment - You need to ask sipx why its giving a 481 to this request and if its an issue I can address I will. This is what the intention was of the detailed post.
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Chris Rawlings added a comment -
This as sent to me from my PBX Vender eZuce... it was a different call log since i had to take an internal sip trace and log.... but all other details are the same....



Chris,
 
I need to confirm with Engineering, but it appears to me that there is an issue with the device 192.168.1.246. The following is the INVITE for the the first pickup by extension 201. Note the replaces header.
 
INVITE sip:201@bluecloudconsultants.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.246:5080;rport;branch=z9hG4bKpacpBKBBUBBmm
Max-Forwards: 70
From: "Outbound Call" <sip:6107413324@192.168.1.246>;tag=5QD36N10g5UUB
To: <sip:201@bluecloudconsultants.com>
Call-ID: 921d62d2-2446-1230-50a0-d5dcba68eeac
CSeq: 28816759 INVITE
Contact: <sip:mod_sofia@192.168.1.246:5080>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7f17ad7 2011-10-01 14-05-14 +0200
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Require: replaces
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Replaces: 4ad097c7-79db5340-469a90bd@192.168.1.19;to-tag=DA1D9F8B-53766694;from-tag=ueVII3
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 203
X-FS-Support: update_display
X-sipX-Authidentity: <sip:~~id~park@bluecloudconsultants.com;signature=4FC4EC6F%3A%3Ad9369c8b014a72df3d985b79ec348a1e>
REFERENCES: 8f0aa0e0-2446-1230-50a0-d5dcba68eeac;rel=refer
Remote-Party-ID: "Outbound Call" <sip:6107413324@192.168.1.246>;party=calling;screen=yes;privacy=off
 
v=0
o=FreeSWITCH 1338278851 1338278852 IN IP4 192.168.1.246
s=FreeSWITCH
c=IN IP4 192.168.1.246
t=0 0
m=audio 26796 RTP/AVP 0 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
 
Then when the call is parked again, here is the INVITE:
 
INVITE sip:11@10.100.1.18 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.246:5080;rport;branch=z9hG4bKUQajmtZvcrtHe
Max-Forwards: 14
From: "Outbound Call" <sip:6107413324@192.168.1.246>;tag=606U8gj4DejeQ
To: <sip:11@10.100.1.18>
Call-ID: 96bbdbe6-2446-1230-50a0-d5dcba68eeac
CSeq: 28816763 INVITE
Contact: <sip:mod_sofia@192.168.1.246:5080>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7f17ad7 2011-10-01 14-05-14 +0200
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Require: replaces
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Referred-By: <sip:201@bluecloudconsultants.com>
Replaces: 4ad097c7-79db5340-469a90bd@192.168.1.19;to-tag=DA1D9F8B-53766694;from-tag=ueVII3
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 203
X-Sipx-Handled: X10.100.1.18-207.228.237.16
X-sipX-Authidentity: <sip:~~id~park@bluecloudconsultants.com;signature=4FC4EC6F%3A%3Ad9369c8b014a72df3d985b79ec348a1e>
REFERENCES: 8f0aa0e0-2446-1230-50a0-d5dcba68eeac;rel=refer
X-FS-Support: update_display
Remote-Party-ID: "Outbound Call" <sip:6107413324@192.168.1.246>;party=calling;screen=yes;privacy=off
 
v=0
o=FreeSWITCH 1338287863 1338287864 IN IP4 192.168.1.246
s=FreeSWITCH
c=IN IP4 192.168.1.246
t=0 0
m=audio 17792 RTP/AVP 0 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
 
The second INVITE contains again contains the same Replaces: header. The Park Server would have terminated that call and would have no reference to the Call-ID. So I believe that is why the Park server issues the 481 Transaction Does Not Exist.
Show
Chris Rawlings added a comment - This as sent to me from my PBX Vender eZuce... it was a different call log since i had to take an internal sip trace and log.... but all other details are the same.... Chris,   I need to confirm with Engineering, but it appears to me that there is an issue with the device 192.168.1.246. The following is the INVITE for the the first pickup by extension 201. Note the replaces header.   INVITE sip: 201@bluecloudconsultants.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.246:5080;rport;branch=z9hG4bKpacpBKBBUBBmm Max-Forwards: 70 From: "Outbound Call" <sip: 6107413324@192.168.1.246 >;tag=5QD36N10g5UUB To: <sip: 201@bluecloudconsultants.com > Call-ID: 921d62d2-2446-1230-50a0-d5dcba68eeac CSeq: 28816759 INVITE Contact: <sip: mod_sofia@192.168.1.246 :5080> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7f17ad7 2011-10-01 14-05-14 +0200 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: replaces Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Replaces: 4ad097c7-79db5340-469a90bd@192.168.1.19 ;to-tag=DA1D9F8B-53766694;from-tag=ueVII3 Content-Type: application/sdp Content-Disposition: session Content-Length: 203 X-FS-Support: update_display X-sipX-Authidentity: <sip:~~id~ park@bluecloudconsultants.com ;signature=4FC4EC6F%3A%3Ad9369c8b014a72df3d985b79ec348a1e> REFERENCES: 8f0aa0e0-2446-1230-50a0-d5dcba68eeac;rel=refer Remote-Party-ID: "Outbound Call" <sip: 6107413324@192.168.1.246 >;party=calling;screen=yes;privacy=off   v=0 o=FreeSWITCH 1338278851 1338278852 IN IP4 192.168.1.246 s=FreeSWITCH c=IN IP4 192.168.1.246 t=0 0 m=audio 26796 RTP/AVP 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20   Then when the call is parked again, here is the INVITE:   INVITE sip: 11@10.100.1.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.246:5080;rport;branch=z9hG4bKUQajmtZvcrtHe Max-Forwards: 14 From: "Outbound Call" <sip: 6107413324@192.168.1.246 >;tag=606U8gj4DejeQ To: <sip: 11@10.100.1.18 > Call-ID: 96bbdbe6-2446-1230-50a0-d5dcba68eeac CSeq: 28816763 INVITE Contact: <sip: mod_sofia@192.168.1.246 :5080> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7f17ad7 2011-10-01 14-05-14 +0200 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: replaces Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Referred-By: <sip: 201@bluecloudconsultants.com > Replaces: 4ad097c7-79db5340-469a90bd@192.168.1.19 ;to-tag=DA1D9F8B-53766694;from-tag=ueVII3 Content-Type: application/sdp Content-Disposition: session Content-Length: 203 X-Sipx-Handled: X10.100.1.18-207.228.237.16 X-sipX-Authidentity: <sip:~~id~ park@bluecloudconsultants.com ;signature=4FC4EC6F%3A%3Ad9369c8b014a72df3d985b79ec348a1e> REFERENCES: 8f0aa0e0-2446-1230-50a0-d5dcba68eeac;rel=refer X-FS-Support: update_display Remote-Party-ID: "Outbound Call" <sip: 6107413324@192.168.1.246 >;party=calling;screen=yes;privacy=off   v=0 o=FreeSWITCH 1338287863 1338287864 IN IP4 192.168.1.246 s=FreeSWITCH c=IN IP4 192.168.1.246 t=0 0 m=audio 17792 RTP/AVP 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20   The second INVITE contains again contains the same Replaces: header. The Park Server would have terminated that call and would have no reference to the Call-ID. So I believe that is why the Park server issues the 481 Transaction Does Not Exist.
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Anthony Minessale II added a comment -
ok this is the data I needed.
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Anthony Minessale II added a comment - ok this is the data I needed.
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Git added a comment -
Repository: freeswitch
Branch: refs/heads/master
Commit: 3e1d68b http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=3e1d68b
Closed By: anthm@freeswitch.org

Comment:
FS-4250 --resolve erase replaces header after using it


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Git added a comment - Repository: freeswitch Branch: refs/heads/master Commit: 3e1d68b http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=3e1d68b Closed By: anthm@freeswitch.org Comment: FS-4250 --resolve erase replaces header after using it FreeSWITCH Support Contracts and Consulting Services available! Contact us: Email: consulting@freeswitch.org Web: http://www.freeswitch.org Phone: +1-918-420-9266 Tollfree: +1-877-742-2583 Fax: +1-918-420-9267 iNum: +883 5100 1420 9266 Come To ClueCon in August to learn more about FreeSWITCH and Internet Telephony! http://www.cluecon.com
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Chris Rawlings added a comment - - edited
same issue after make current... i am on version 39bad5b13ef8a85608a6783a9ebdaed3713da04d

posting a new SIP Trace

Looks like the replaces still stays the same
Although i have maybe misunderstood the issue


Replaces: 1de6b2e6-1719e653-b4b457e8@10.2.1.16
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Chris Rawlings added a comment - - edited same issue after make current... i am on version 39bad5b13ef8a85608a6783a9ebdaed3713da04d posting a new SIP Trace Looks like the replaces still stays the same Although i have maybe misunderstood the issue Replaces: 1de6b2e6-1719e653-b4b457e8@10.2.1.16
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Anthony Minessale II added a comment -
I need the whole trace, can you post one?

if you still have the same replaces header in the invite with latest it would suggest the same replaces header is also in the REFER
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Anthony Minessale II added a comment - I need the whole trace, can you post one? if you still have the same replaces header in the invite with latest it would suggest the same replaces header is also in the REFER
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Chris Rawlings added a comment -
trace-debug.still-not-working posted above
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Chris Rawlings added a comment - trace-debug.still-not-working posted above
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Git added a comment -
Repository: freeswitch
Branch: refs/heads/master
Commit: 693e04a http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=693e04a
Updated By: anthm@freeswitch.org

Comment:
FS-4250 please post a new log if this does not work


FreeSWITCH Support Contracts and Consulting Services available!

Contact us:
Email: consulting@freeswitch.org
Web: http://www.freeswitch.org
Phone: +1-918-420-9266
Tollfree: +1-877-742-2583
Fax: +1-918-420-9267
iNum: +883 5100 1420 9266


Come To ClueCon in August to learn more about FreeSWITCH and Internet Telephony!
http://www.cluecon.com

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Git added a comment - Repository: freeswitch Branch: refs/heads/master Commit: 693e04a http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=693e04a Updated By: anthm@freeswitch.org Comment: FS-4250 please post a new log if this does not work FreeSWITCH Support Contracts and Consulting Services available! Contact us: Email: consulting@freeswitch.org Web: http://www.freeswitch.org Phone: +1-918-420-9266 Tollfree: +1-877-742-2583 Fax: +1-918-420-9267 iNum: +883 5100 1420 9266 Come To ClueCon in August to learn more about FreeSWITCH and Internet Telephony! http://www.cluecon.com
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Chris Rawlings added a comment -
this is fixed...

we have new issues to report now that we have made it passed this issue i have found 3 more... 2 are call drop issues that need addressing... the others is no MoH while blind transfering... i will post new tickets about these then

you may close this unless you want more data from me on this
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Chris Rawlings added a comment - this is fixed... we have new issues to report now that we have made it passed this issue i have found 3 more... 2 are call drop issues that need addressing... the others is no MoH while blind transfering... i will post new tickets about these then you may close this unless you want more data from me on this
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Due to a long period of inactivity (13 or more days), this issue is due to be automatically close within 24 hours.
If this issue is not actually resolved, please reopen it and add appropriate comments to help developers fix the issue.

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  Jira Admin
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Auto Admin added a comment - Due to a long period of inactivity (13 or more days), this issue is due to be automatically close within 24 hours. If this issue is not actually resolved, please reopen it and add appropriate comments to help developers fix the issue. Thanks,   Jira Admin
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Since there has been no change to the status of this issue, it is being closed for inactivity

Thanks,
  Jira Admin


FreeSWITCH Support Contracts and Consulting Services available!

Contact us:
Email: consulting@freeswitch.org
Web: http://www.freeswitch.org
Phone: +1-918-420-9266
Tollfree: +1-877-742-2583
Fax: +1-918-420-9267
iNum: +883 5100 1420 9266


Come To ClueCon in August to learn more about FreeSWITCH and Internet Telephony!
http://www.cluecon.com

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Auto Admin added a comment - Since there has been no change to the status of this issue, it is being closed for inactivity Thanks,   Jira Admin FreeSWITCH Support Contracts and Consulting Services available! Contact us: Email: consulting@freeswitch.org Web: http://www.freeswitch.org Phone: +1-918-420-9266 Tollfree: +1-877-742-2583 Fax: +1-918-420-9267 iNum: +883 5100 1420 9266 Come To ClueCon in August to learn more about FreeSWITCH and Internet Telephony! http://www.cluecon.com
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Git added a comment -
Repository: freeswitch
Branch: refs/heads/FS-3432
Commit: 3e1d68b http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=3e1d68b
Closed By: d@d-man.org

Comment:
FS-4250 --resolve erase replaces header after using it


FreeSWITCH Support Contracts and Consulting Services available!

Contact us:
Email: consulting@freeswitch.org
Web: http://www.freeswitch.org
Phone: +1-918-420-9266
Tollfree: +1-877-742-2583
Fax: +1-918-420-9267
iNum: +883 5100 1420 9266


Come To ClueCon in August to learn more about FreeSWITCH and Internet Telephony!
http://www.cluecon.com

Show
Git added a comment - Repository: freeswitch Branch: refs/heads/ FS-3432 Commit: 3e1d68b http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=3e1d68b Closed By: d@d-man.org Comment: FS-4250 --resolve erase replaces header after using it FreeSWITCH Support Contracts and Consulting Services available! Contact us: Email: consulting@freeswitch.org Web: http://www.freeswitch.org Phone: +1-918-420-9266 Tollfree: +1-877-742-2583 Fax: +1-918-420-9267 iNum: +883 5100 1420 9266 Come To ClueCon in August to learn more about FreeSWITCH and Internet Telephony! http://www.cluecon.com
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Git added a comment -
Repository: freeswitch
Branch: refs/heads/FS-3432
Commit: 693e04a http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=693e04a
Updated By: d@d-man.org

Comment:
FS-4250 please post a new log if this does not work


FreeSWITCH Support Contracts and Consulting Services available!

Contact us:
Email: consulting@freeswitch.org
Web: http://www.freeswitch.org
Phone: +1-918-420-9266
Tollfree: +1-877-742-2583
Fax: +1-918-420-9267
iNum: +883 5100 1420 9266


Come To ClueCon in August to learn more about FreeSWITCH and Internet Telephony!
http://www.cluecon.com

Show
Git added a comment - Repository: freeswitch Branch: refs/heads/ FS-3432 Commit: 693e04a http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=693e04a Updated By: d@d-man.org Comment: FS-4250 please post a new log if this does not work FreeSWITCH Support Contracts and Consulting Services available! Contact us: Email: consulting@freeswitch.org Web: http://www.freeswitch.org Phone: +1-918-420-9266 Tollfree: +1-877-742-2583 Fax: +1-918-420-9267 iNum: +883 5100 1420 9266 Come To ClueCon in August to learn more about FreeSWITCH and Internet Telephony! http://www.cluecon.com

People

  • Assignee:
    Anthony Minessale II
    Reporter:
    Chris Rawlings
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Dates

  • Created:
    Updated:
    Resolved: