major problem with latest version
hello i expirience a big problem with the latest version (pulled from git). inbound sip calls don't get answered (i do answer them with the answer app). does anybody expirience the same issue? br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl@xpirio.com
What happens when calls come in? Post a console log. -MC Sent from my iPhone On Jun 12, 2010, at 1:34 AM, Christian Löschenkohl <christian.loeschenkohl@xpirio.co m> wrote:
hello
i expirience a big problem with the latest version (pulled from git). inbound sip calls don't get answered (i do answer them with the answer app). does anybody expirience the same issue?
br
-- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP
xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria
T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl@xpirio.com
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org
hello the problem is the following fs sends in actual git head P-Asserted-Identity: "43720570500" <sip:43720570500@(null)>. the git version before (around 7 june) did send (with the same config/server) P-Asserted-Identity: "43720570500" <sip:43720570500@92.63.208.34>. so the call does not get answered pastebin of the two 200 ok packets - http://pastebin.freeswitch.org/13175 fyi: the system at 93.185.139.77 is a sonus switch br On 2010-06-12 10:34, Christian Löschenkohl wrote:
hello
i expirience a big problem with the latest version (pulled from git). inbound sip calls don't get answered (i do answer them with the answer app). does anybody expirience the same issue?
br
-- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl@xpirio.com
I experienced the same problem. It also happened to Remote-Party-ID. I have opened Jira for this issue and 2 more related issues as well: http://jira.freeswitch.org/browse/SFSIP-214. You can post your log to the same Jira if you like since I believe it's the similar issue. -djbinter 2010/6/13 Christian Löschenkohl <christian.loeschenkohl@xpirio.com>
hello
the problem is the following
fs sends in actual git head P-Asserted-Identity: "43720570500" <sip:43720570500@(null)>.
the git version before (around 7 june) did send (with the same config/server) P-Asserted-Identity: "43720570500" <sip:43720570500@92.63.208.34<sip%3A43720570500@92.63.208.34>
.
so the call does not get answered
pastebin of the two 200 ok packets - http://pastebin.freeswitch.org/13175 fyi: the system at 93.185.139.77 is a sonus switch
br
On 2010-06-12 10:34, Christian Löschenkohl wrote:
hello
i expirience a big problem with the latest version (pulled from git). inbound sip calls don't get answered (i do answer them with the answer app). does anybody expirience the same issue?
br
-- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP
xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria
T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl@xpirio.com
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
it would work fine if you apply it to the profile and NOT the gateway. /b Sent from my iPad On Jun 13, 2010, at 6:08 PM, DJB International <djbinter@gmail.com> wrote:
I experienced the same problem. It also happened to Remote-Party-ID.
I have opened Jira for this issue and 2 more related issues as well: http://jira.freeswitch.org/browse/SFSIP-214.
You can post your log to the same Jira if you like since I believe it's the similar issue.
-djbinter
2010/6/13 Christian Löschenkohl <christian.loeschenkohl@xpirio.com> hello
the problem is the following
fs sends in actual git head P-Asserted-Identity: "43720570500" <sip:43720570500@(null)>.
the git version before (around 7 june) did send (with the same config/server) P-Asserted-Identity: "43720570500" <sip:43720570500@92.63.208.34>.
so the call does not get answered
pastebin of the two 200 ok packets - http://pastebin.freeswitch.org/13175 fyi: the system at 93.185.139.77 is a sonus switch
br
On 2010-06-12 10:34, Christian Löschenkohl wrote:
hello
i expirience a big problem with the latest version (pulled from git). inbound sip calls don't get answered (i do answer them with the answer app). does anybody expirience the same issue?
br
-- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP
xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria
T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl@xpirio.com
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The issue was we override the cid_type with tech_pvt->cid_type it seems but we already default it to the profile setting which should win if you set it. Patch pushed but it would have worked already if you export sip_cid_type variable before a bridge, Also your use of a gateway is not needed it seems but to each his own. /b On Jun 13, 2010, at 6:08 PM, DJB International wrote:
I experienced the same problem. It also happened to Remote-Party-ID.
I have opened Jira for this issue and 2 more related issues as well: http://jira.freeswitch.org/browse/SFSIP-214.
You can post your log to the same Jira if you like since I believe it's the similar issue.
-djbinter
Brian, Pastebin: http://pastebin.freeswitch.org/13178 git-a95fa59 2010-06-13 19-17-52 -0500 The <param name="caller-id-type" value="pid"/> is now working on sip profile (line 144 - P-Asserted-Identity: "6265101611" <sip:6265101611@ 204.110.12.103>.) ; however, the null value is still showing (line 251, line 305, line 365 - Remote-Party-ID: "6628888888" <sip:6628888888@(null)>;party=calling;privacy=off;screen=no. ) Thank you. -djbinter On Sun, Jun 13, 2010 at 5:08 PM, Brian West <brian@freeswitch.org> wrote:
The issue was we override the cid_type with tech_pvt->cid_type it seems but we already default it to the profile setting which should win if you set it.
Patch pushed but it would have worked already if you export sip_cid_type variable before a bridge, Also your use of a gateway is not needed it seems but to each his own.
/b
On Jun 13, 2010, at 6:08 PM, DJB International wrote:
I experienced the same problem. It also happened to Remote-Party-ID.
I have opened Jira for this issue and 2 more related issues as well: http://jira.freeswitch.org/browse/SFSIP-214.
You can post your log to the same Jira if you like since I believe it's the similar issue.
-djbinter
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Yes a jira already exists for that but I don't see how that is even possible to have the domain null in the rpid because the code tries very hard to not have a null rpid_domain in do_invite in sofia_glue. /b On Jun 13, 2010, at 8:48 PM, DJB International wrote:
Brian,
Pastebin: http://pastebin.freeswitch.org/13178
git-a95fa59 2010-06-13 19-17-52 -0500
The <param name="caller-id-type" value="pid"/> is now working on sip profile (line 144 - P-Asserted-Identity: "6265101611" <sip:6265101611@204.110.12.103>.) ;
however, the null value is still showing (line 251, line 305, line 365 - Remote-Party-ID: "6628888888" <sip:6628888888@(null)>;party=calling;privacy=off;screen=no. )
Thank you.
-djbinter
Brian, Do you mean the jira that I opened: SFSIP-214? If so, I think it was closed. Should I re-open it? Thank you, -djbinter On Sun, Jun 13, 2010 at 6:58 PM, Brian West <brian@freeswitch.org> wrote:
Yes a jira already exists for that but I don't see how that is even possible to have the domain null in the rpid because the code tries very hard to not have a null rpid_domain in do_invite in sofia_glue.
/b
On Jun 13, 2010, at 8:48 PM, DJB International wrote:
Brian,
Pastebin: http://pastebin.freeswitch.org/13178
git-a95fa59 2010-06-13 19-17-52 -0500
The <param name="caller-id-type" value="pid"/> is now working on sip profile (line 144 - P-Asserted-Identity: "6265101611" <sip:6265101611@ 204.110.12.103>.) ;
however, the null value is still showing (line 251, line 305, line 365 - Remote-Party-ID: "6628888888" <sip:6628888888@(null)>;party=calling;privacy=off;screen=no. )
Thank you.
-djbinter
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The issue is outlined in http://jira.freeswitch.org/browse/FSMOD-61 /b On Jun 13, 2010, at 9:12 PM, DJB International wrote:
Brian,
Do you mean the jira that I opened: SFSIP-214? If so, I think it was closed. Should I re-open it?
Thank you, -djbinter
Brian, One thing I make me confused is that if I have <param name="caller-id-type" value="pid"/> in sofia profile, should the FS change both sip message to terminating and originating gateway to use P-Asserted-Identity. However, I only saw the change to terminating gateway to be P-Asserted-Identity, but the message (sip: 180, 200) that FS sends to originating gateway is still using Remote-Party-ID. Thank you, Brian. -djbinter On Sun, Jun 13, 2010 at 7:26 PM, Brian West <brian@freeswitch.org> wrote:
The issue is outlined in http://jira.freeswitch.org/browse/FSMOD-61
/b
On Jun 13, 2010, at 9:12 PM, DJB International wrote:
Brian,
Do you mean the jira that I opened: SFSIP-214? If so, I think it was closed. Should I re-open it?
Thank you, -djbinter
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You can't control what is sent to you with a setting on the sofia profile. /b On Jun 13, 2010, at 10:04 PM, DJB International wrote:
Brian,
One thing I make me confused is that if I have <param name="caller-id-type" value="pid"/> in sofia profile, should the FS change both sip message to terminating and originating gateway to use P-Asserted-Identity. However, I only saw the change to terminating gateway to be P-Asserted-Identity, but the message (sip: 180, 200) that FS sends to originating gateway is still using Remote-Party-ID.
Thank you, Brian.
-djbinter
Brain, I meant was what sofia sent out. For instance, from the previous pastebin: http://pastebin.freeswitch.org/13178, if you look at line 251, 305, and 365. Call Flow is from 75.214.107.143 (originating) -> 204.110.12.103 (FS) -> 209.9.188.91 (terminating) Thank you, -djbinter On Mon, Jun 14, 2010 at 6:47 AM, Brian West <brian@freeswitch.org> wrote:
You can't control what is sent to you with a setting on the sofia profile.
/b
On Jun 13, 2010, at 10:04 PM, DJB International wrote:
Brian,
One thing I make me confused is that if I have <param name="caller-id-type" value="pid"/> in sofia profile, should the FS change both sip message to terminating and originating gateway to use P-Asserted-Identity. However, I only saw the change to terminating gateway to be P-Asserted-Identity, but the message (sip: 180, 200) that FS sends to originating gateway is still using Remote-Party-ID.
Thank you, Brian.
-djbinter
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hey Brain, give neurons to FS community ! :D ----- Original Message ----- From: DJB International To: freeswitch-users@lists.freeswitch.org Sent: Monday, June 14, 2010 12:14 PM Subject: Re: [Freeswitch-users] major problem with latest version Brain, I meant was what sofia sent out. For instance, from the previous pastebin: http://pastebin.freeswitch.org/13178, if you look at line 251, 305, and 365. Call Flow is from 75.214.107.143 (originating) -> 204.110.12.103 (FS) -> 209.9.188.91 (terminating) Thank you, -djbinter On Mon, Jun 14, 2010 at 6:47 AM, Brian West <brian@freeswitch.org> wrote: You can't control what is sent to you with a setting on the sofia profile. /b On Jun 13, 2010, at 10:04 PM, DJB International wrote: Brian, One thing I make me confused is that if I have <param name="caller-id-type" value="pid"/> in sofia profile, should the FS change both sip message to terminating and originating gateway to use P-Asserted-Identity. However, I only saw the change to terminating gateway to be P-Asserted-Identity, but the message (sip: 180, 200) that FS sends to originating gateway is still using Remote-Party-ID. Thank you, Brian. -djbinter _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (5)
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Brian West
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Christian Löschenkohl
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DJB International
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Madovsky
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Michael S Collins