Newbie: Avaya SES <>Freeswitch 407 Proxy Authentication error
Here's my configuration: <include> <gateway name="avaya"> <param name="username" value="3824"/> <param name="password" value="password"/> <param name="proxy" value="10.0.2.154"/> <param name="expire-seconds" value="60"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="retry_seconds" value="30"/> </gateway> </include> - sofia shows successfully registered with SES (SIP Enablement Services) When I call 3824 from another softphone registered with SES, sofia issues a 407 Proxy Authentication error and the call does not go through. I presume their is something wrong in my configuration, but I cannot figure it out. Any ideas? Gerry
Have you added the <extension name="avayaSES"> in dialplan/public.xml ? -- Regards, Gayatri Kulkarni On Thu, Oct 9, 2008 at 1:05 PM, Gerry Hull <gerry@pstn2.net> wrote:
Here's my configuration:
<include> <gateway name="avaya"> <param name="username" value="3824"/> <param name="password" value="password"/> <param name="proxy" value="10.0.2.154"/> <param name="expire-seconds" value="60"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="retry_seconds" value="30"/> </gateway> </include>
- sofia shows successfully registered with SES (SIP Enablement Services)
When I call 3824 from another softphone registered with SES, sofia issues a 407 Proxy Authentication error and the call does not go through.
I presume their is something wrong in my configuration, but I cannot figure it out.
Any ideas?
Gerry
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Of course. On Mon, Oct 13, 2008 at 2:31 AM, Gayatri Kulkarni <xtpl.gayatri@gmail.com>wrote:
Have you added the <extension name="avayaSES"> in dialplan/public.xml ? -- Regards, Gayatri Kulkarni
On Thu, Oct 9, 2008 at 1:05 PM, Gerry Hull <gerry@pstn2.net> wrote:
Here's my configuration:
<include> <gateway name="avaya"> <param name="username" value="3824"/> <param name="password" value="password"/> <param name="proxy" value="10.0.2.154"/> <param name="expire-seconds" value="60"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="retry_seconds" value="30"/> </gateway> </include>
- sofia shows successfully registered with SES (SIP Enablement Services)
When I call 3824 from another softphone registered with SES, sofia issues a 407 Proxy Authentication error and the call does not go through.
I presume their is something wrong in my configuration, but I cannot figure it out.
Any ideas?
Gerry
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Thanks!! On Mon, Oct 13, 2008 at 12:59 PM, Brian West <brian@freeswitch.org> wrote:
You need to add <param name="extension" value="avayaSES"/> otherwise the register contact will be the username. aka 3824, its trying to route to 3824 in context public.
/b
On Oct 13, 2008, at 9:53 AM, Gerry Hull wrote:
Of course.
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On Mon, Oct 13, 2008 at 12:59 PM, Brian West <brian@freeswitch.org> wrote:
You need to add <param name="extension" value="avayaSES"/> otherwise the register contact will be the username. aka 3824, its trying to route to 3824 in context public.
/b
On Oct 13, 2008, at 9:53 AM, Gerry Hull wrote:
Hi Brian, Still no luck. I guess I'm still missing something. Let me explain and provide more details. We have several Avaya extensions registered in FS; we have the Avaya switch direct inbound calls to these extensions. The idea is to park the inbound calls in FS; we will then transfer the calls later using event_socket. However, we cannot get FS to answer the calls due to the proxy authentication error. Here's the configuration: /sip_profiles/internal/AvayaInternal.xml: <include> <gateway name="Avaya"> <param name="extension" value="Avaya"/> <param name="username" value="3823"/> <param name="password" value="xxx"/> <param name="proxy" value="1.2.3.4"/> <param name="realm" value="1.2.3.4"/> <param name="expire-seconds" value="60"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="retry_seconds" value="30"/> </gateway> </include> /dialplan/public.xml: <extension name="Avaya"> <condition field="destination_number" expression="^(3823)$"> <action application="transfer" data="5060 XML default"/> </condition> </extension> and here's the debug info: ------------------------------------------------------------------------ INVITE sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0 Call-ID: 05c8e85a1a5dd14ea549c5a6a00 CSeq: 1 INVITE From: "Station 216" <sip:216@sdc.com:5061
;tag=05c8e85a1a5dd14da549c5a6a00
Record-Route: <sip:10.0.2.154:5060;lr>,<sip:10.0.2.151:5061 ;lr;transport=tls> To: "3823" <sip:3823@sdc.com <sip%3A3823@sdc.com>> Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10 .0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00 Content-Length: 150 Content-Type: application/sdp Contact: "Station 216" <sip:216@10.0.2.151:5061;transport=tls> Max-Forwards: 69 User-Agent: Avaya CM/R014x.00.1.731.2 Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS Accept-Contact: *;+avaya-cm-line=1 Supported: 100rel,timer,replaces,join,histinfo Alert-Info: <cid:internal@sdc.com <cid%3Ainternal@sdc.com>
;avaya-cm-alert-type=internal
Min-SE: 1200 Session-Expires: 1200;refresher=uac P-Asserted-Identity: "Station 216" <sip:216@sdc.com:5061> History-Info: <sip:3823@sdc.com <sip%3A3823@sdc.com>>;index=1,"3823" < sip:3823@sdc.com <sip%3A3823@sdc.com>>;index=1.1 v=0 o=- 1 1 IN IP4 10.0.2.151 s=- c=IN IP4 10.0.2.152 t=0 0 m=audio 3188 RTP/AVP 0 127 a=rtpmap:0 PCMU/8000 a=rtpmap:127 telephone-event/8000 ------------------------------------------------------------------------ tport(01833120): msg 01872960 (1194 bytes) from udp/10.0.2.154:5060/sipnext=00000000 nta: received INVITE sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0 (CSeq 1) nta: canonizing sip:Avaya@10.0.6.114:5060 with contact nta: INVITE (1) going to a default leg nta: timer set to 200 ms soa_clone(static::017FD508, 017FED48, 0188E450) called soa_set_params(static::0189BB50, ...) called nta_leg_tcreate(00FEDDE8) soa_init_offer_answer(static::0189BB50) called soa_set_remote_sdp(static::0189BB50, 00000000, 0189E804, 150) called nua(0188E450): adding session usage tport_tsend(01833120) tpn = UDP/10.0.2.154:5060 tport_resolve addrinfo = 10.0.2.154:5060 tport(01833120): not found by name UDP/10.0.2.154:5060 tport_vsend(01833120): 515 bytes of 515 to UDP/10.0.2.154:5060 tport_vsend returned 515 send 515 bytes to udp/[10.0.2.154]:5060 at 14:09:50.666454: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10 .0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00 Record-Route: <sip:10.0.2.154:5060;lr> Record-Route: <sip:10.0.2.151:5061;lr;transport=tls> From: "Station 216" <sip:216@sdc.com:5061
;tag=05c8e85a1a5dd14da549c5a6a00
To: "3823" <sip:3823@sdc.com <sip%3A3823@sdc.com>> Call-ID: 05c8e85a1a5dd14ea549c5a6a00 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9570M Content-Length: 0 ------------------------------------------------------------------------ nta: sent 100 Trying for INVITE (1) nua(0188E450): event i_invite 100 Trying nua(0188E450): call state changed: init -> received, received offer soa_get_remote_sdp(static::0189BB50, [029EFC10], [029EFC0C], [00000000]) called nua(0188E450): event i_state 100 Trying nua(0188E450): sent signal r_respond nua(0188E450): sent signal r_destroy nua(0188E450): event i_state dropped nua(0188E450): recv signal r_respond 407 Proxy Authentication Required soa_set_params(static::0189BB50, ...) called soa_clear_remote_sdp(static::0189BB50) called tport_tsend(01833120) tpn = UDP/10.0.2.154:5060 tport_resolve addrinfo = 10.0.2.154:5060 tport(01833120): not found by name UDP/10.0.2.154:5060 tport_vsend(01833120): 893 bytes of 893 to UDP/10.0.2.154:5060 tport_vsend returned 893 send 893 bytes to udp/[10.0.2.154]:5060 at 14:09:50.697704: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10 .0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00 From: "Station 216" <sip:216@sdc.com:5061
;tag=05c8e85a1a5dd14da549c5a6a00
To: "3823" <sip:3823@sdc.com <sip%3A3823@sdc.com>>;tag=ptgKFvSeyaDrQ Call-ID: 05c8e85a1a5dd14ea549c5a6a00 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9570M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary Proxy-Authenticate: Digest realm="sdc.com", nonce="21e6dda0-ee03-674b-8df1-08731d42fd76", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ nta: sent 407 Proxy Authentication Required for INVITE (1) nua(0188E450): removing session usage nua(0188E450): call state changed: received -> terminated nua(0188E450): event i_state 407 Proxy Authentication Required nua(0188E450): event i_terminated 407 Proxy Authentication Required soa_destroy(static::0189BB50) called nta_leg_destroy(00FEDDE8) nua(0188E450): recv signal r_destroy nta_leg_destroy(00000000) tport_wakeup_pri(01833120): events IN tport_recv_event(01833120) tport(01833120) msg 018390F8 from (udp/10.0.6.114:5060) has 341 bytes, veclen = 1 recv 341 bytes from udp/[10.0.2.154]:32772 at 14:09:50.697704: ------------------------------------------------------------------------ ACK sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0 From: "Station 216" <sip:216@sdc.com:5061
;tag=05c8e85a1a5dd14da549c5a6a00
To: "3823" <sip:3823@sdc.com <sip%3A3823@sdc.com>>;tag=ptgKFvSeyaDrQ CSeq: 1 ACK Call-ID: 05c8e85a1a5dd14ea549c5a6a00 Max-Forwards: 69 Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0 Content-Length: 0 ------------------------------------------------------------------------ tport(01833120): msg 018390F8 (341 bytes) from udp/10.0.2.154:5060/sip next=00000000 <http://10.0.2.154:5060/sipnext=00000000> nta: received ACK sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0 (CSeq 1) nta: ACK (1) is going to INVITE (1) nta: timer set next to 4844 ms nta: timer I fired, terminate 407 response incoming_reclaim_all(00000000, 00000000, 029EFEB8) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set TIA, Gerry
Every time I have set this up in the past with the avaya I have used ip auth and an ip trusted peer on the avaya side. Mike On Oct 14, 2008, at 10:18 AM, Gerry Hull wrote:
On Mon, Oct 13, 2008 at 12:59 PM, Brian West <brian@freeswitch.org> wrote: You need to add <param name="extension" value="avayaSES"/> otherwise the register contact will be the username. aka 3824, its trying to route to 3824 in context public.
/b
On Oct 13, 2008, at 9:53 AM, Gerry Hull wrote:
Hi Brian,
Still no luck. I guess I'm still missing something. Let me explain and provide more details.
We have several Avaya extensions registered in FS; we have the Avaya switch direct inbound calls to these extensions. The idea is to park the inbound calls in FS; we will then transfer the calls later using event_socket. However, we cannot get FS to answer the calls due to the proxy authentication error.
Here's the configuration:
/sip_profiles/internal/AvayaInternal.xml:
<include>
<gateway name="Avaya">
<param name="extension" value="Avaya"/>
<param name="username" value="3823"/>
<param name="password" value="xxx"/>
<param name="proxy" value="1.2.3.4"/>
<param name="realm" value="1.2.3.4"/>
<param name="expire-seconds" value="60"/>
<param name="register" value="true"/>
<param name="register-transport" value="udp"/>
<param name="retry_seconds" value="30"/>
</gateway>
</include>
/dialplan/public.xml:
<extension name="Avaya">
<condition field="destination_number" expression="^(3823)$">
<action application="transfer" data="5060 XML default"/>
</condition>
</extension>
and here's the debug info:
------------------------------------------------------------------------
INVITE sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0
Call-ID: 05c8e85a1a5dd14ea549c5a6a00
CSeq: 1 INVITE
From: "Station 216" <sip:216@sdc.com: 5061>;tag=05c8e85a1a5dd14da549c5a6a00
Record-Route: <sip:10.0.2.154:5060;lr>,<sip:10.0.2.151:5061;lr;transport=tls
To: "3823" <sip:3823@sdc.com>
Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10
.0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00
Content-Length: 150
Content-Type: application/sdp
Contact: "Station 216" <sip:216@10.0.2.151:5061;transport=tls>
Max-Forwards: 69
User-Agent: Avaya CM/R014x.00.1.731.2
Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS
Accept-Contact: *;+avaya-cm-line=1
Supported: 100rel,timer,replaces,join,histinfo
Alert-Info: <cid:internal@sdc.com>;avaya-cm-alert-type=internal
Min-SE: 1200
Session-Expires: 1200;refresher=uac
P-Asserted-Identity: "Station 216" <sip:216@sdc.com:5061>
History-Info: <sip:3823@sdc.com>;index=1,"3823" <sip: 3823@sdc.com>;index=1.1
v=0
o=- 1 1 IN IP4 10.0.2.151
s=-
c=IN IP4 10.0.2.152
t=0 0
m=audio 3188 RTP/AVP 0 127
a=rtpmap:0 PCMU/8000
a=rtpmap:127 telephone-event/8000
------------------------------------------------------------------------
tport(01833120): msg 01872960 (1194 bytes) from udp/10.0.2.154:5060/ sip next=00000000
nta: received INVITE sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0 (CSeq 1)
nta: canonizing sip:Avaya@10.0.6.114:5060 with contact
nta: INVITE (1) going to a default leg
nta: timer set to 200 ms
soa_clone(static::017FD508, 017FED48, 0188E450) called
soa_set_params(static::0189BB50, ...) called
nta_leg_tcreate(00FEDDE8)
soa_init_offer_answer(static::0189BB50) called
soa_set_remote_sdp(static::0189BB50, 00000000, 0189E804, 150) called
nua(0188E450): adding session usage
tport_tsend(01833120) tpn = UDP/10.0.2.154:5060
tport_resolve addrinfo = 10.0.2.154:5060
tport(01833120): not found by name UDP/10.0.2.154:5060
tport_vsend(01833120): 515 bytes of 515 to UDP/10.0.2.154:5060
tport_vsend returned 515
send 515 bytes to udp/[10.0.2.154]:5060 at 14:09:50.666454:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10
.0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00
Record-Route: <sip:10.0.2.154:5060;lr>
Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
From: "Station 216" <sip:216@sdc.com: 5061>;tag=05c8e85a1a5dd14da549c5a6a00
To: "3823" <sip:3823@sdc.com>
Call-ID: 05c8e85a1a5dd14ea549c5a6a00
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9570M
Content-Length: 0
------------------------------------------------------------------------
nta: sent 100 Trying for INVITE (1)
nua(0188E450): event i_invite 100 Trying
nua(0188E450): call state changed: init -> received, received offer
soa_get_remote_sdp(static::0189BB50, [029EFC10], [029EFC0C], [00000000]) called
nua(0188E450): event i_state 100 Trying
nua(0188E450): sent signal r_respond
nua(0188E450): sent signal r_destroy
nua(0188E450): event i_state dropped
nua(0188E450): recv signal r_respond 407 Proxy Authentication Required
soa_set_params(static::0189BB50, ...) called
soa_clear_remote_sdp(static::0189BB50) called
tport_tsend(01833120) tpn = UDP/10.0.2.154:5060
tport_resolve addrinfo = 10.0.2.154:5060
tport(01833120): not found by name UDP/10.0.2.154:5060
tport_vsend(01833120): 893 bytes of 893 to UDP/10.0.2.154:5060
tport_vsend returned 893
send 893 bytes to udp/[10.0.2.154]:5060 at 14:09:50.697704:
------------------------------------------------------------------------
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10
.0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00
From: "Station 216" <sip:216@sdc.com: 5061>;tag=05c8e85a1a5dd14da549c5a6a00
To: "3823" <sip:3823@sdc.com>;tag=ptgKFvSeyaDrQ
Call-ID: 05c8e85a1a5dd14ea549c5a6a00
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9570M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include- session-description, presence.winfo, message-summary
Proxy-Authenticate: Digest realm="sdc.com", nonce="21e6dda0- ee03-674b-8df1-08731d42fd76", algorithm=MD5, qop="auth"
Content-Length: 0
------------------------------------------------------------------------
nta: sent 407 Proxy Authentication Required for INVITE (1)
nua(0188E450): removing session usage
nua(0188E450): call state changed: received -> terminated
nua(0188E450): event i_state 407 Proxy Authentication Required
nua(0188E450): event i_terminated 407 Proxy Authentication Required
soa_destroy(static::0189BB50) called
nta_leg_destroy(00FEDDE8)
nua(0188E450): recv signal r_destroy
nta_leg_destroy(00000000)
tport_wakeup_pri(01833120): events IN
tport_recv_event(01833120)
tport(01833120) msg 018390F8 from (udp/10.0.6.114:5060) has 341 bytes, veclen = 1
recv 341 bytes from udp/[10.0.2.154]:32772 at 14:09:50.697704:
------------------------------------------------------------------------
ACK sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0
From: "Station 216" <sip:216@sdc.com: 5061>;tag=05c8e85a1a5dd14da549c5a6a00
To: "3823" <sip:3823@sdc.com>;tag=ptgKFvSeyaDrQ
CSeq: 1 ACK
Call-ID: 05c8e85a1a5dd14ea549c5a6a00
Max-Forwards: 69
Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0
Content-Length: 0
------------------------------------------------------------------------
tport(01833120): msg 018390F8 (341 bytes) from udp/10.0.2.154:5060/ sip next=00000000
nta: received ACK sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0 (CSeq 1)
nta: ACK (1) is going to INVITE (1)
nta: timer set next to 4844 ms
nta: timer I fired, terminate 407 response
incoming_reclaim_all(00000000, 00000000, 029EFEB8)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta: timer not set
TIA,
Gerry
Mike, We have the freeswitch box as a trusted host. Still getting the 407. Any other ideas? Gerry On Tue, Oct 14, 2008 at 11:30 AM, Michael Jerris <mike@jerris.com> wrote:
Every time I have set this up in the past with the avaya I have used ip auth and an ip trusted peer on the avaya side. Mike
On Oct 14, 2008, at 10:18 AM, Gerry Hull wrote:
On Mon, Oct 13, 2008 at 12:59 PM, Brian West <brian@freeswitch.org> wrote:
You need to add <param name="extension" value="avayaSES"/> otherwise the register contact will be the username. aka 3824, its trying to route to 3824 in context public.
/b
On Oct 13, 2008, at 9:53 AM, Gerry Hull wrote:
Hi Brian,
Still no luck. I guess I'm still missing something. Let me explain and provide more details.
We have several Avaya extensions registered in FS; we have the Avaya switch direct inbound calls to these extensions. The idea is to park the inbound calls in FS; we will then transfer the calls later using event_socket. However, we cannot get FS to answer the calls due to the proxy authentication error.
Here's the configuration:
/sip_profiles/internal/AvayaInternal.xml:
<include>
<gateway name="Avaya">
<param name="extension" value="Avaya"/>
<param name="username" value="3823"/>
<param name="password" value="xxx"/>
<param name="proxy" value="1.2.3.4"/>
<param name="realm" value="1.2.3.4"/>
<param name="expire-seconds" value="60"/>
<param name="register" value="true"/>
<param name="register-transport" value="udp"/>
<param name="retry_seconds" value="30"/>
</gateway>
</include>
/dialplan/public.xml:
<extension name="Avaya">
<condition field="destination_number" expression="^(3823)$">
<action application="transfer" data="5060 XML default"/>
</condition>
</extension>
and here's the debug info:
------------------------------------------------------------------------
INVITE sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0
Call-ID: 05c8e85a1a5dd14ea549c5a6a00
CSeq: 1 INVITE
From: "Station 216" <sip:216@sdc.com:5061
;tag=05c8e85a1a5dd14da549c5a6a00
Record-Route: <sip:10.0.2.154:5060;lr>,< sip:10.0.2.151:5061;lr;transport=tls>
To: "3823" <sip:3823@sdc.com <sip%3A3823@sdc.com>>
Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10
.0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00
Content-Length: 150
Content-Type: application/sdp
Contact: "Station 216" <sip:216@10.0.2.151:5061;transport=tls>
Max-Forwards: 69
User-Agent: Avaya CM/R014x.00.1.731.2
Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS
Accept-Contact: *;+avaya-cm-line=1
Supported: 100rel,timer,replaces,join,histinfo
Alert-Info: <cid:internal@sdc.com <cid%3Ainternal@sdc.com>
;avaya-cm-alert-type=internal
Min-SE: 1200
Session-Expires: 1200;refresher=uac
P-Asserted-Identity: "Station 216" <sip:216@sdc.com:5061>
History-Info: <sip:3823@sdc.com <sip%3A3823@sdc.com>>;index=1,"3823" < sip:3823@sdc.com <sip%3A3823@sdc.com>>;index=1.1
v=0
o=- 1 1 IN IP4 10.0.2.151
s=-
c=IN IP4 10.0.2.152
t=0 0
m=audio 3188 RTP/AVP 0 127
a=rtpmap:0 PCMU/8000
a=rtpmap:127 telephone-event/8000
------------------------------------------------------------------------
tport(01833120): msg 01872960 (1194 bytes) from udp/10.0.2.154:5060/sipnext=00000000
nta: received INVITE sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0 (CSeq 1)
nta: canonizing sip:Avaya@10.0.6.114:5060 with contact
nta: INVITE (1) going to a default leg
nta: timer set to 200 ms
soa_clone(static::017FD508, 017FED48, 0188E450) called
soa_set_params(static::0189BB50, ...) called
nta_leg_tcreate(00FEDDE8)
soa_init_offer_answer(static::0189BB50) called
soa_set_remote_sdp(static::0189BB50, 00000000, 0189E804, 150) called
nua(0188E450): adding session usage
tport_tsend(01833120) tpn = UDP/10.0.2.154:5060
tport_resolve addrinfo = 10.0.2.154:5060
tport(01833120): not found by name UDP/10.0.2.154:5060
tport_vsend(01833120): 515 bytes of 515 to UDP/10.0.2.154:5060
tport_vsend returned 515
send 515 bytes to udp/[10.0.2.154]:5060 at 14:09:50.666454:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10
.0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00
Record-Route: <sip:10.0.2.154:5060;lr>
Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
From: "Station 216" <sip:216@sdc.com:5061
;tag=05c8e85a1a5dd14da549c5a6a00
To: "3823" <sip:3823@sdc.com <sip%3A3823@sdc.com>>
Call-ID: 05c8e85a1a5dd14ea549c5a6a00
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9570M
Content-Length: 0
------------------------------------------------------------------------
nta: sent 100 Trying for INVITE (1)
nua(0188E450): event i_invite 100 Trying
nua(0188E450): call state changed: init -> received, received offer
soa_get_remote_sdp(static::0189BB50, [029EFC10], [029EFC0C], [00000000]) called
nua(0188E450): event i_state 100 Trying
nua(0188E450): sent signal r_respond
nua(0188E450): sent signal r_destroy
nua(0188E450): event i_state dropped
nua(0188E450): recv signal r_respond 407 Proxy Authentication Required
soa_set_params(static::0189BB50, ...) called
soa_clear_remote_sdp(static::0189BB50) called
tport_tsend(01833120) tpn = UDP/10.0.2.154:5060
tport_resolve addrinfo = 10.0.2.154:5060
tport(01833120): not found by name UDP/10.0.2.154:5060
tport_vsend(01833120): 893 bytes of 893 to UDP/10.0.2.154:5060
tport_vsend returned 893
send 893 bytes to udp/[10.0.2.154]:5060 at 14:09:50.697704:
------------------------------------------------------------------------
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10
.0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00
From: "Station 216" <sip:216@sdc.com:5061
;tag=05c8e85a1a5dd14da549c5a6a00
To: "3823" <sip:3823@sdc.com <sip%3A3823@sdc.com>>;tag=ptgKFvSeyaDrQ
Call-ID: 05c8e85a1a5dd14ea549c5a6a00
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9570M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary
Proxy-Authenticate: Digest realm="sdc.com", nonce="21e6dda0-ee03-674b-8df1-08731d42fd76", algorithm=MD5, qop="auth"
Content-Length: 0
------------------------------------------------------------------------
nta: sent 407 Proxy Authentication Required for INVITE (1)
nua(0188E450): removing session usage
nua(0188E450): call state changed: received -> terminated
nua(0188E450): event i_state 407 Proxy Authentication Required
nua(0188E450): event i_terminated 407 Proxy Authentication Required
soa_destroy(static::0189BB50) called
nta_leg_destroy(00FEDDE8)
nua(0188E450): recv signal r_destroy
nta_leg_destroy(00000000)
tport_wakeup_pri(01833120): events IN
tport_recv_event(01833120)
tport(01833120) msg 018390F8 from (udp/10.0.6.114:5060) has 341 bytes, veclen = 1
recv 341 bytes from udp/[10.0.2.154]:32772 at 14:09:50.697704:
------------------------------------------------------------------------
ACK sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0
From: "Station 216" <sip:216@sdc.com:5061
;tag=05c8e85a1a5dd14da549c5a6a00
To: "3823" <sip:3823@sdc.com <sip%3A3823@sdc.com>>;tag=ptgKFvSeyaDrQ
CSeq: 1 ACK
Call-ID: 05c8e85a1a5dd14ea549c5a6a00
Max-Forwards: 69
Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0
Content-Length: 0
------------------------------------------------------------------------
tport(01833120): msg 018390F8 (341 bytes) from udp/10.0.2.154:5060/sip next=00000000 <http://10.0.2.154:5060/sipnext=00000000>
nta: received ACK sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0 (CSeq 1)
nta: ACK (1) is going to INVITE (1)
nta: timer set next to 4844 ms
nta: timer I fired, terminate 407 response
incoming_reclaim_all(00000000, 00000000, 029EFEB8)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta: timer not set TIA,
Gerry
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On Oct 14, 2008, at 1:24 PM, Gerry Hull wrote:
Mike,
We have the freeswitch box as a trusted host. Still getting the 407. Any other ideas?
Gerry
If your still getting a 407 then either your trusted host is setup wrong on the avaya side or possibly your sending to an extension that it doesn't know what to do with, is the extension "Avaya" really what your sending to and is that setup properly on the avaya side? Mike
Record-Route: <sip:10.0.2.154:5060;lr> Record-Route: <sip:10.0.2.151:5061;lr;transport=tls> what's the 'lr' next to the port number? Did you notice the Content-Length for your INVITE is 0? Usually taking an etheral trace and analyzing it helps in case of Avaya SES - even for administration issues -- Regards, Gayatri Kulkarni On Tue, Oct 14, 2008 at 7:48 PM, Gerry Hull <gerry@pstn2.net> wrote:
On Mon, Oct 13, 2008 at 12:59 PM, Brian West <brian@freeswitch.org> wrote:
You need to add <param name="extension" value="avayaSES"/> otherwise the register contact will be the username. aka 3824, its trying to route to 3824 in context public.
/b
On Oct 13, 2008, at 9:53 AM, Gerry Hull wrote:
Hi Brian,
Still no luck. I guess I'm still missing something. Let me explain and provide more details.
We have several Avaya extensions registered in FS; we have the Avaya switch direct inbound calls to these extensions. The idea is to park the inbound calls in FS; we will then transfer the calls later using event_socket. However, we cannot get FS to answer the calls due to the proxy authentication error.
Here's the configuration:
/sip_profiles/internal/AvayaInternal.xml:
<include>
<gateway name="Avaya">
<param name="extension" value="Avaya"/>
<param name="username" value="3823"/>
<param name="password" value="xxx"/>
<param name="proxy" value="1.2.3.4"/>
<param name="realm" value="1.2.3.4"/>
<param name="expire-seconds" value="60"/>
<param name="register" value="true"/>
<param name="register-transport" value="udp"/>
<param name="retry_seconds" value="30"/>
</gateway>
</include>
/dialplan/public.xml:
<extension name="Avaya">
<condition field="destination_number" expression="^(3823)$">
<action application="transfer" data="5060 XML default"/>
</condition>
</extension>
and here's the debug info:
------------------------------------------------------------------------
INVITE sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0
Call-ID: 05c8e85a1a5dd14ea549c5a6a00
CSeq: 1 INVITE
From: "Station 216" <sip:216@sdc.com:5061
;tag=05c8e85a1a5dd14da549c5a6a00
Record-Route: <sip:10.0.2.154:5060;lr>,<sip:10.0.2.151:5061 ;lr;transport=tls>
To: "3823" <sip:3823@sdc.com <sip%3A3823@sdc.com>>
Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10
.0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00
Content-Length: 150
Content-Type: application/sdp
Contact: "Station 216" <sip:216@10.0.2.151:5061;transport=tls>
Max-Forwards: 69
User-Agent: Avaya CM/R014x.00.1.731.2
Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS
Accept-Contact: *;+avaya-cm-line=1
Supported: 100rel,timer,replaces,join,histinfo
Alert-Info: <cid:internal@sdc.com <cid%3Ainternal@sdc.com>
;avaya-cm-alert-type=internal
Min-SE: 1200
Session-Expires: 1200;refresher=uac
P-Asserted-Identity: "Station 216" <sip:216@sdc.com:5061>
History-Info: <sip:3823@sdc.com <sip%3A3823@sdc.com>>;index=1,"3823" < sip:3823@sdc.com <sip%3A3823@sdc.com>>;index=1.1
v=0
o=- 1 1 IN IP4 10.0.2.151
s=-
c=IN IP4 10.0.2.152
t=0 0
m=audio 3188 RTP/AVP 0 127
a=rtpmap:0 PCMU/8000
a=rtpmap:127 telephone-event/8000
------------------------------------------------------------------------
tport(01833120): msg 01872960 (1194 bytes) from udp/10.0.2.154:5060/sipnext=00000000
nta: received INVITE sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0 (CSeq 1)
nta: canonizing sip:Avaya@10.0.6.114:5060 with contact
nta: INVITE (1) going to a default leg
nta: timer set to 200 ms
soa_clone(static::017FD508, 017FED48, 0188E450) called
soa_set_params(static::0189BB50, ...) called
nta_leg_tcreate(00FEDDE8)
soa_init_offer_answer(static::0189BB50) called
soa_set_remote_sdp(static::0189BB50, 00000000, 0189E804, 150) called
nua(0188E450): adding session usage
tport_tsend(01833120) tpn = UDP/10.0.2.154:5060
tport_resolve addrinfo = 10.0.2.154:5060
tport(01833120): not found by name UDP/10.0.2.154:5060
tport_vsend(01833120): 515 bytes of 515 to UDP/10.0.2.154:5060
tport_vsend returned 515
send 515 bytes to udp/[10.0.2.154]:5060 at 14:09:50.666454:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10
.0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00
Record-Route: <sip:10.0.2.154:5060;lr>
Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
From: "Station 216" <sip:216@sdc.com:5061
;tag=05c8e85a1a5dd14da549c5a6a00
To: "3823" <sip:3823@sdc.com <sip%3A3823@sdc.com>>
Call-ID: 05c8e85a1a5dd14ea549c5a6a00
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9570M
Content-Length: 0
------------------------------------------------------------------------
nta: sent 100 Trying for INVITE (1)
nua(0188E450): event i_invite 100 Trying
nua(0188E450): call state changed: init -> received, received offer
soa_get_remote_sdp(static::0189BB50, [029EFC10], [029EFC0C], [00000000]) called
nua(0188E450): event i_state 100 Trying
nua(0188E450): sent signal r_respond
nua(0188E450): sent signal r_destroy
nua(0188E450): event i_state dropped
nua(0188E450): recv signal r_respond 407 Proxy Authentication Required
soa_set_params(static::0189BB50, ...) called
soa_clear_remote_sdp(static::0189BB50) called
tport_tsend(01833120) tpn = UDP/10.0.2.154:5060
tport_resolve addrinfo = 10.0.2.154:5060
tport(01833120): not found by name UDP/10.0.2.154:5060
tport_vsend(01833120): 893 bytes of 893 to UDP/10.0.2.154:5060
tport_vsend returned 893
send 893 bytes to udp/[10.0.2.154]:5060 at 14:09:50.697704:
------------------------------------------------------------------------
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10
.0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00
From: "Station 216" <sip:216@sdc.com:5061
;tag=05c8e85a1a5dd14da549c5a6a00
To: "3823" <sip:3823@sdc.com <sip%3A3823@sdc.com>>;tag=ptgKFvSeyaDrQ
Call-ID: 05c8e85a1a5dd14ea549c5a6a00
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9570M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary
Proxy-Authenticate: Digest realm="sdc.com", nonce="21e6dda0-ee03-674b-8df1-08731d42fd76", algorithm=MD5, qop="auth"
Content-Length: 0
------------------------------------------------------------------------
nta: sent 407 Proxy Authentication Required for INVITE (1)
nua(0188E450): removing session usage
nua(0188E450): call state changed: received -> terminated
nua(0188E450): event i_state 407 Proxy Authentication Required
nua(0188E450): event i_terminated 407 Proxy Authentication Required
soa_destroy(static::0189BB50) called
nta_leg_destroy(00FEDDE8)
nua(0188E450): recv signal r_destroy
nta_leg_destroy(00000000)
tport_wakeup_pri(01833120): events IN
tport_recv_event(01833120)
tport(01833120) msg 018390F8 from (udp/10.0.6.114:5060) has 341 bytes, veclen = 1
recv 341 bytes from udp/[10.0.2.154]:32772 at 14:09:50.697704:
------------------------------------------------------------------------
ACK sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0
From: "Station 216" <sip:216@sdc.com:5061
;tag=05c8e85a1a5dd14da549c5a6a00
To: "3823" <sip:3823@sdc.com <sip%3A3823@sdc.com>>;tag=ptgKFvSeyaDrQ
CSeq: 1 ACK
Call-ID: 05c8e85a1a5dd14ea549c5a6a00
Max-Forwards: 69
Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0
Content-Length: 0
------------------------------------------------------------------------
tport(01833120): msg 018390F8 (341 bytes) from udp/10.0.2.154:5060/sip next=00000000 <http://10.0.2.154:5060/sipnext=00000000>
nta: received ACK sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0 (CSeq 1)
nta: ACK (1) is going to INVITE (1)
nta: timer set next to 4844 ms
nta: timer I fired, terminate 407 response
incoming_reclaim_all(00000000, 00000000, 029EFEB8)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta: timer not set TIA,
Gerry
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On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote:
Record-Route: <sip:10.0.2.154:5060;lr> Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
what's the 'lr' next to the port number?
short for 'loose routing' - see here for a bit of an explanation: http://www.tech-invite.com/Ti-sip-dialog.html --Dave
Thanks David! Gerry,
From the debug info you have sent, looks like Avaya SES asks for PAI i.e Proxy Authentication Indication - It's a kind of challenge response authentication. After it receives the user's digest in response to this request (again), it authenticates the user. This is the normal behavior of Avaya SES. the users' digest is not sent again it seems!
On Wed, Oct 15, 2008 at 1:52 PM, David Knell <dave@3c.co.uk> wrote:
On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote:
Record-Route: <sip:10.0.2.154:5060;lr>
Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
what's the 'lr' next to the port number?
short for 'loose routing' - see here for a bit of an explanation: http://www.tech-invite.com/Ti-sip-dialog.html
--Dave
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-- Regards, Gayatri Kulkarni
Gayatri, Any idea on how to enable this response in Freeswitch? David, Not sure of the "lr"... On Wed, Oct 15, 2008 at 4:38 AM, Gayatri Kulkarni <xtpl.gayatri@gmail.com>wrote:
Thanks David! Gerry, From the debug info you have sent, looks like Avaya SES asks for PAI i.e Proxy Authentication Indication - It's a kind of challenge response authentication. After it receives the user's digest in response to this request (again), it authenticates the user. This is the normal behavior of Avaya SES. the users' digest is not sent again it seems!
On Wed, Oct 15, 2008 at 1:52 PM, David Knell <dave@3c.co.uk> wrote:
On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote:
Record-Route: <sip:10.0.2.154:5060;lr>
Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
what's the 'lr' next to the port number?
short for 'loose routing' - see here for a bit of an explanation: http://www.tech-invite.com/Ti-sip-dialog.html
--Dave
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-- Regards, Gayatri Kulkarni
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I haven't used Freeswitch with Avaya SES. I have used Avaya SES with Avaya phones! and Freeswitch with xlite separately :( Anyways, if someone has a freeswitch succesfully setup with avaya SES, i suggest, tally your configuration with them, because in that case it's got to be configuration issue Otherwise, we should look into the sofia module. We'll have to tell it to send the digest again if it's an Avaya SES - rather if it's the same Avaya SES box asking user credentials for the second time -- Regards, Gayatri Kulkarni On Wed, Oct 15, 2008 at 7:12 PM, Gerry Hull <gerry@pstn2.net> wrote:
Gayatri,
Any idea on how to enable this response in Freeswitch?
David,
Not sure of the "lr"...
On Wed, Oct 15, 2008 at 4:38 AM, Gayatri Kulkarni <xtpl.gayatri@gmail.com>wrote:
Thanks David! Gerry, From the debug info you have sent, looks like Avaya SES asks for PAI i.e Proxy Authentication Indication - It's a kind of challenge response authentication. After it receives the user's digest in response to this request (again), it authenticates the user. This is the normal behavior of Avaya SES. the users' digest is not sent again it seems!
On Wed, Oct 15, 2008 at 1:52 PM, David Knell <dave@3c.co.uk> wrote:
On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote:
Record-Route: <sip:10.0.2.154:5060;lr>
Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
what's the 'lr' next to the port number?
short for 'loose routing' - see here for a bit of an explanation: http://www.tech-invite.com/Ti-sip-dialog.html
--Dave
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-- Regards, Gayatri Kulkarni
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Gerry, Did you manage to take the ethereal trace? It would be definitely of more help and we can narrow down the actual problem Do you have access to Avaya SES? To take the ethreal trace, you should: *1)telnet* <user>@<Avaya SES> //reduces size of the trace, ssh size usually goes in GBs 2)login 3)tethereal -i eth0 -f <path/to/filename> 4)try to make the call (get the 407) 5)kill tethereal (Ctrl+C) 6) copy the file to a place where you can sit and analyze it You can analyze the trace yourself if you have wireshark installed, or send it over -- Regards, Gayatri Kulkarni On Wed, Oct 15, 2008 at 7:12 PM, Gerry Hull <gerry@pstn2.net> wrote:
Gayatri,
Any idea on how to enable this response in Freeswitch?
David,
Not sure of the "lr"...
On Wed, Oct 15, 2008 at 4:38 AM, Gayatri Kulkarni <xtpl.gayatri@gmail.com>wrote:
Thanks David! Gerry, From the debug info you have sent, looks like Avaya SES asks for PAI i.e Proxy Authentication Indication - It's a kind of challenge response authentication. After it receives the user's digest in response to this request (again), it authenticates the user. This is the normal behavior of Avaya SES. the users' digest is not sent again it seems!
On Wed, Oct 15, 2008 at 1:52 PM, David Knell <dave@3c.co.uk> wrote:
On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote:
Record-Route: <sip:10.0.2.154:5060;lr>
Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
what's the 'lr' next to the port number?
short for 'loose routing' - see here for a bit of an explanation: http://www.tech-invite.com/Ti-sip-dialog.html
--Dave
_______________________________________________ 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
-- Regards, Gayatri Kulkarni
_______________________________________________ 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
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I've solved most of my issues by trunking Freeswitch with Avaya SES. It was more on the SES configuration side. FreeSwitch seems to perform flawlessly. Now the only problem is, if I dial into Freeswitch from an Avaya extension, and hangup the call, I get no SIP BYE or CANCEL from SES! Oh joy! Thanks for all your help on this, guys. Gerry On Thu, Oct 16, 2008 at 1:48 AM, Gayatri Kulkarni <xtpl.gayatri@gmail.com>wrote:
Gerry, Did you manage to take the ethereal trace? It would be definitely of more help and we can narrow down the actual problem Do you have access to Avaya SES?
To take the ethreal trace, you should: *1)telnet* <user>@<Avaya SES> //reduces size of the trace, ssh size usually goes in GBs 2)login 3)tethereal -i eth0 -f <path/to/filename> 4)try to make the call (get the 407) 5)kill tethereal (Ctrl+C) 6) copy the file to a place where you can sit and analyze it
You can analyze the trace yourself if you have wireshark installed, or send it over
-- Regards, Gayatri Kulkarni
On Wed, Oct 15, 2008 at 7:12 PM, Gerry Hull <gerry@pstn2.net> wrote:
Gayatri,
Any idea on how to enable this response in Freeswitch?
David,
Not sure of the "lr"...
On Wed, Oct 15, 2008 at 4:38 AM, Gayatri Kulkarni <xtpl.gayatri@gmail.com
wrote:
Thanks David! Gerry, From the debug info you have sent, looks like Avaya SES asks for PAI i.e Proxy Authentication Indication - It's a kind of challenge response authentication. After it receives the user's digest in response to this request (again), it authenticates the user. This is the normal behavior of Avaya SES. the users' digest is not sent again it seems!
On Wed, Oct 15, 2008 at 1:52 PM, David Knell <dave@3c.co.uk> wrote:
On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote:
Record-Route: <sip:10.0.2.154:5060;lr>
Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
what's the 'lr' next to the port number?
short for 'loose routing' - see here for a bit of an explanation: http://www.tech-invite.com/Ti-sip-dialog.html
--Dave
_______________________________________________ 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
-- Regards, Gayatri Kulkarni
_______________________________________________ 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
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make sure you have all of the nat related params off in FS just in case. edit your sip profile and comment out anything that says nat On Fri, Oct 17, 2008 at 12:51 PM, Gerry Hull <gerry@pstn2.net> wrote:
I've solved most of my issues by trunking Freeswitch with Avaya SES. It was more on the SES configuration side. FreeSwitch seems to perform flawlessly.
Now the only problem is, if I dial into Freeswitch from an Avaya extension, and hangup the call, I get no SIP BYE or CANCEL from SES!
Oh joy!
Thanks for all your help on this, guys.
Gerry
On Thu, Oct 16, 2008 at 1:48 AM, Gayatri Kulkarni <xtpl.gayatri@gmail.com>wrote:
Gerry, Did you manage to take the ethereal trace? It would be definitely of more help and we can narrow down the actual problem Do you have access to Avaya SES?
To take the ethreal trace, you should: *1)telnet* <user>@<Avaya SES> //reduces size of the trace, ssh size usually goes in GBs 2)login 3)tethereal -i eth0 -f <path/to/filename> 4)try to make the call (get the 407) 5)kill tethereal (Ctrl+C) 6) copy the file to a place where you can sit and analyze it
You can analyze the trace yourself if you have wireshark installed, or send it over
-- Regards, Gayatri Kulkarni
On Wed, Oct 15, 2008 at 7:12 PM, Gerry Hull <gerry@pstn2.net> wrote:
Gayatri,
Any idea on how to enable this response in Freeswitch?
David,
Not sure of the "lr"...
On Wed, Oct 15, 2008 at 4:38 AM, Gayatri Kulkarni < xtpl.gayatri@gmail.com> wrote:
Thanks David! Gerry, From the debug info you have sent, looks like Avaya SES asks for PAI i.e Proxy Authentication Indication - It's a kind of challenge response authentication. After it receives the user's digest in response to this request (again), it authenticates the user. This is the normal behavior of Avaya SES. the users' digest is not sent again it seems!
On Wed, Oct 15, 2008 at 1:52 PM, David Knell <dave@3c.co.uk> wrote:
On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote:
Record-Route: <sip:10.0.2.154:5060;lr>
Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
what's the 'lr' next to the port number?
short for 'loose routing' - see here for a bit of an explanation: http://www.tech-invite.com/Ti-sip-dialog.html
--Dave
_______________________________________________ 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
-- Regards, Gayatri Kulkarni
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Thanks Anthony, We have turned off all nat-related anything, but are still having an issue. However, here is more information: -- We are running FreeSwitch on Windows Server 2003 R2 -- We are running FreeSwitch in a console app for testing Calls from Avaya are answered in the dial plan and a javascript is executed: <extension name="5060"> <condition field="destination_number" expression="^5060$"> <action application="javascript" data="Answer.js 5061 welcome.wav"/> </condition> </extension> Here's the js: var sAudioFilePath = "/sounds/"; session.answer(); if(argv.length > 1) { for (var i = 1; i < argv.length; i ++) { session.streamFile(sAudioFilePath + argv[i]); } } session.execute("transfer", argv[0]); exit(); After playing the file, the caller is transferred into a park, using this dialplan code: <extension name="5060_Park"> <condition field="destination_number" expression="^5061$"> <action application="set" data="fifo_music=$${sound_prefix}/sounds/pleasewait.wav"/> <action application="fifo" data="5061@$${domain} in"/> <action application="set" data="hangup_after_bridge=true"/> </condition> </extension> Here's the issue: If we hang up the call while it is parked, we see NOTHING in the sofia debug or freeswitch log. (console window). We deceided to log freeswitch log info to a file. If we shut down FreeSwitch when it is the state with a call that is parked, we do see the SIP BYE message, and everythig shuts down. It's as if some thread is hung. All of this only happens with an inbound Avaya trunk. If we dial from an anonymous SIP connection, everything works fine. Ideas? Regards, Gerry On Fri, Oct 17, 2008 at 3:30 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
make sure you have all of the nat related params off in FS just in case. edit your sip profile and comment out anything that says nat
On Fri, Oct 17, 2008 at 12:51 PM, Gerry Hull <gerry@pstn2.net> wrote:
I've solved most of my issues by trunking Freeswitch with Avaya SES. It was more on the SES configuration side. FreeSwitch seems to perform flawlessly.
Now the only problem is, if I dial into Freeswitch from an Avaya extension, and hangup the call, I get no SIP BYE or CANCEL from SES!
Oh joy!
Thanks for all your help on this, guys.
Gerry
On Thu, Oct 16, 2008 at 1:48 AM, Gayatri Kulkarni <xtpl.gayatri@gmail.com
wrote:
Gerry, Did you manage to take the ethereal trace? It would be definitely of more help and we can narrow down the actual problem Do you have access to Avaya SES?
To take the ethreal trace, you should: *1)telnet* <user>@<Avaya SES> //reduces size of the trace, ssh size usually goes in GBs 2)login 3)tethereal -i eth0 -f <path/to/filename> 4)try to make the call (get the 407) 5)kill tethereal (Ctrl+C) 6) copy the file to a place where you can sit and analyze it
You can analyze the trace yourself if you have wireshark installed, or send it over
-- Regards, Gayatri Kulkarni
On Wed, Oct 15, 2008 at 7:12 PM, Gerry Hull <gerry@pstn2.net> wrote:
Gayatri,
Any idea on how to enable this response in Freeswitch?
David,
Not sure of the "lr"...
On Wed, Oct 15, 2008 at 4:38 AM, Gayatri Kulkarni < xtpl.gayatri@gmail.com> wrote:
Thanks David! Gerry, From the debug info you have sent, looks like Avaya SES asks for PAI i.e Proxy Authentication Indication - It's a kind of challenge response authentication. After it receives the user's digest in response to this request (again), it authenticates the user. This is the normal behavior of Avaya SES. the users' digest is not sent again it seems!
On Wed, Oct 15, 2008 at 1:52 PM, David Knell <dave@3c.co.uk> wrote:
On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote:
Record-Route: <sip:10.0.2.154:5060;lr>
Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
what's the 'lr' next to the port number?
short for 'loose routing' - see here for a bit of an explanation: http://www.tech-invite.com/Ti-sip-dialog.html
--Dave
_______________________________________________ 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
-- Regards, Gayatri Kulkarni
_______________________________________________ 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
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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UA1 discovers P1 as a result of configuration, DHCP assignment or other similar operation, also outside the scope of this document. REGISTRAR has a similar "*default outbound proxy*" relationship with P3. Check if you have this configured on the Avaya SES as your FS. To do that, login to admin interface of Avaya SES and in the setup side bar, there's a link Called 'Hosts' click on the link for this home and check what outbound proxy has been specified Is your SES box a Home or Edge or Home/Edge combo?The Hosts link appears only on Edge or home/Edge combo Thanks Thomas! -- Regards, Gayatri Kulkarni On Wed, Oct 15, 2008 at 7:12 PM, Gerry Hull <gerry@pstn2.net> wrote:
Gayatri,
Any idea on how to enable this response in Freeswitch?
David,
Not sure of the "lr"...
On Wed, Oct 15, 2008 at 4:38 AM, Gayatri Kulkarni <xtpl.gayatri@gmail.com>wrote:
Thanks David! Gerry, From the debug info you have sent, looks like Avaya SES asks for PAI i.e Proxy Authentication Indication - It's a kind of challenge response authentication. After it receives the user's digest in response to this request (again), it authenticates the user. This is the normal behavior of Avaya SES. the users' digest is not sent again it seems!
On Wed, Oct 15, 2008 at 1:52 PM, David Knell <dave@3c.co.uk> wrote:
On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote:
Record-Route: <sip:10.0.2.154:5060;lr>
Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
what's the 'lr' next to the port number?
short for 'loose routing' - see here for a bit of an explanation: http://www.tech-invite.com/Ti-sip-dialog.html
--Dave
_______________________________________________ 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
-- Regards, Gayatri Kulkarni
_______________________________________________ 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
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You could always set up an acl so FS won't send auth to that particular host. On Thu, Oct 16, 2008 at 12:58 AM, Gayatri Kulkarni <xtpl.gayatri@gmail.com>wrote:
UA1 discovers P1 as a result of configuration, DHCP assignment or other similar operation, also outside the scope of this document. REGISTRAR has a similar "*default outbound proxy*" relationship with
P3.
Check if you have this configured on the Avaya SES as your FS. To do that, login to admin interface of Avaya SES and in the setup side bar, there's a link Called 'Hosts' click on the link for this home and check what outbound proxy has been specified Is your SES box a Home or Edge or Home/Edge combo?The Hosts link appears only on Edge or home/Edge combo
Thanks Thomas!
-- Regards, Gayatri Kulkarni
On Wed, Oct 15, 2008 at 7:12 PM, Gerry Hull <gerry@pstn2.net> wrote:
Gayatri,
Any idea on how to enable this response in Freeswitch?
David,
Not sure of the "lr"...
On Wed, Oct 15, 2008 at 4:38 AM, Gayatri Kulkarni <xtpl.gayatri@gmail.com
wrote:
Thanks David! Gerry, From the debug info you have sent, looks like Avaya SES asks for PAI i.e Proxy Authentication Indication - It's a kind of challenge response authentication. After it receives the user's digest in response to this request (again), it authenticates the user. This is the normal behavior of Avaya SES. the users' digest is not sent again it seems!
On Wed, Oct 15, 2008 at 1:52 PM, David Knell <dave@3c.co.uk> wrote:
On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote:
Record-Route: <sip:10.0.2.154:5060;lr>
Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
what's the 'lr' next to the port number?
short for 'loose routing' - see here for a bit of an explanation: http://www.tech-invite.com/Ti-sip-dialog.html
--Dave
_______________________________________________ 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
-- Regards, Gayatri Kulkarni
_______________________________________________ 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
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Didn't get that On Thu, Oct 16, 2008 at 6:13 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You could always set up an acl so FS won't send auth to that particular host.
On Thu, Oct 16, 2008 at 12:58 AM, Gayatri Kulkarni <xtpl.gayatri@gmail.com
wrote:
UA1 discovers P1 as a result of configuration, DHCP assignment or other
similar operation, also outside the scope of this document. REGISTRAR has a similar "*default outbound proxy*" relationship with
P3.
Check if you have this configured on the Avaya SES as your FS. To do that, login to admin interface of Avaya SES and in the setup side bar, there's a link Called 'Hosts' click on the link for this home and check what outbound proxy has been specified Is your SES box a Home or Edge or Home/Edge combo?The Hosts link appears only on Edge or home/Edge combo
Thanks Thomas!
-- Regards, Gayatri Kulkarni
On Wed, Oct 15, 2008 at 7:12 PM, Gerry Hull <gerry@pstn2.net> wrote:
Gayatri,
Any idea on how to enable this response in Freeswitch?
David,
Not sure of the "lr"...
On Wed, Oct 15, 2008 at 4:38 AM, Gayatri Kulkarni < xtpl.gayatri@gmail.com> wrote:
Thanks David! Gerry, From the debug info you have sent, looks like Avaya SES asks for PAI i.e Proxy Authentication Indication - It's a kind of challenge response authentication. After it receives the user's digest in response to this request (again), it authenticates the user. This is the normal behavior of Avaya SES. the users' digest is not sent again it seems!
On Wed, Oct 15, 2008 at 1:52 PM, David Knell <dave@3c.co.uk> wrote:
On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote:
Record-Route: <sip:10.0.2.154:5060;lr>
Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
what's the 'lr' next to the port number?
short for 'loose routing' - see here for a bit of an explanation: http://www.tech-invite.com/Ti-sip-dialog.html
--Dave
_______________________________________________ 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
-- Regards, Gayatri Kulkarni
_______________________________________________ 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
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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-- Regards, Gayatri Kulkarni
ip based authentication using ACL (Access Control List) search on wiki or ask in irc On Fri, Oct 17, 2008 at 12:21 AM, Gayatri Kulkarni <xtpl.gayatri@gmail.com>wrote:
Didn't get that
On Thu, Oct 16, 2008 at 6:13 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You could always set up an acl so FS won't send auth to that particular host.
On Thu, Oct 16, 2008 at 12:58 AM, Gayatri Kulkarni < xtpl.gayatri@gmail.com> wrote:
UA1 discovers P1 as a result of configuration, DHCP assignment or other
similar operation, also outside the scope of this document. REGISTRAR has a similar "*default outbound proxy*" relationship with
P3.
Check if you have this configured on the Avaya SES as your FS. To do that, login to admin interface of Avaya SES and in the setup side bar, there's a link Called 'Hosts' click on the link for this home and check what outbound proxy has been specified Is your SES box a Home or Edge or Home/Edge combo?The Hosts link appears only on Edge or home/Edge combo
Thanks Thomas!
-- Regards, Gayatri Kulkarni
On Wed, Oct 15, 2008 at 7:12 PM, Gerry Hull <gerry@pstn2.net> wrote:
Gayatri,
Any idea on how to enable this response in Freeswitch?
David,
Not sure of the "lr"...
On Wed, Oct 15, 2008 at 4:38 AM, Gayatri Kulkarni < xtpl.gayatri@gmail.com> wrote:
Thanks David! Gerry, From the debug info you have sent, looks like Avaya SES asks for PAI i.e Proxy Authentication Indication - It's a kind of challenge response authentication. After it receives the user's digest in response to this request (again), it authenticates the user. This is the normal behavior of Avaya SES. the users' digest is not sent again it seems!
On Wed, Oct 15, 2008 at 1:52 PM, David Knell <dave@3c.co.uk> wrote:
On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote:
Record-Route: <sip:10.0.2.154:5060;lr>
Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
what's the 'lr' next to the port number?
short for 'loose routing' - see here for a bit of an explanation: http://www.tech-invite.com/Ti-sip-dialog.html
--Dave
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-- Regards, Gayatri Kulkarni
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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-- Regards, Gayatri Kulkarni
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Ofcourse i understood ACL is Access Control Lists, but FS should send the auth info to Avaya SES again otherwise it(SES) does not authenticate it. On Fri, Oct 17, 2008 at 10:19 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
ip based authentication using ACL (Access Control List) search on wiki or ask in irc
On Fri, Oct 17, 2008 at 12:21 AM, Gayatri Kulkarni <xtpl.gayatri@gmail.com
wrote:
Didn't get that
On Thu, Oct 16, 2008 at 6:13 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You could always set up an acl so FS won't send auth to that particular host.
On Thu, Oct 16, 2008 at 12:58 AM, Gayatri Kulkarni < xtpl.gayatri@gmail.com> wrote:
UA1 discovers P1 as a result of configuration, DHCP assignment or other
similar operation, also outside the scope of this document. REGISTRAR has a similar "*default outbound proxy*" relationship with
P3.
Check if you have this configured on the Avaya SES as your FS. To do that, login to admin interface of Avaya SES and in the setup side bar, there's a link Called 'Hosts' click on the link for this home and check what outbound proxy has been specified Is your SES box a Home or Edge or Home/Edge combo?The Hosts link appears only on Edge or home/Edge combo
Thanks Thomas!
-- Regards, Gayatri Kulkarni
On Wed, Oct 15, 2008 at 7:12 PM, Gerry Hull <gerry@pstn2.net> wrote:
Gayatri,
Any idea on how to enable this response in Freeswitch?
David,
Not sure of the "lr"...
On Wed, Oct 15, 2008 at 4:38 AM, Gayatri Kulkarni < xtpl.gayatri@gmail.com> wrote:
Thanks David! Gerry, From the debug info you have sent, looks like Avaya SES asks for PAI i.e Proxy Authentication Indication - It's a kind of challenge response authentication. After it receives the user's digest in response to this request (again), it authenticates the user. This is the normal behavior of Avaya SES. the users' digest is not sent again it seems!
On Wed, Oct 15, 2008 at 1:52 PM, David Knell <dave@3c.co.uk> wrote:
> > On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote: > > Record-Route: <sip:10.0.2.154:5060;lr> > > Record-Route: <sip:10.0.2.151:5061;lr;transport=tls> > > what's the 'lr' next to the port number? > > > short for 'loose routing' - see here for a bit of an explanation: > http://www.tech-invite.com/Ti-sip-dialog.html > > --Dave > > _______________________________________________ > 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 > >
-- Regards, Gayatri Kulkarni
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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-- Regards, Gayatri Kulkarni
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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-- Regards, Gayatri Kulkarni
Hi, it is for loose route for the details see http://www.faqs.org/rfcs/rfc3327.html Thomas On 15 Oct 2008, at 09:00, Gayatri Kulkarni wrote:
Record-Route: <sip:10.0.2.154:5060;lr> Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
what's the 'lr' next to the port number?
Did you notice the Content-Length for your INVITE is 0?
Usually taking an etheral trace and analyzing it helps in case of Avaya SES - even for administration issues
-- Regards, Gayatri Kulkarni
On Tue, Oct 14, 2008 at 7:48 PM, Gerry Hull <gerry@pstn2.net> wrote: On Mon, Oct 13, 2008 at 12:59 PM, Brian West <brian@freeswitch.org> wrote: You need to add <param name="extension" value="avayaSES"/> otherwise the register contact will be the username. aka 3824, its trying to route to 3824 in context public.
/b
On Oct 13, 2008, at 9:53 AM, Gerry Hull wrote:
Hi Brian,
Still no luck. I guess I'm still missing something. Let me explain and provide more details.
We have several Avaya extensions registered in FS; we have the Avaya switch direct inbound calls to these extensions. The idea is to park the inbound calls in FS; we will then transfer the calls later using event_socket. However, we cannot get FS to answer the calls due to the proxy authentication error.
Here's the configuration:
/sip_profiles/internal/AvayaInternal.xml:
<include>
<gateway name="Avaya">
<param name="extension" value="Avaya"/>
<param name="username" value="3823"/>
<param name="password" value="xxx"/>
<param name="proxy" value="1.2.3.4"/>
<param name="realm" value="1.2.3.4"/>
<param name="expire-seconds" value="60"/>
<param name="register" value="true"/>
<param name="register-transport" value="udp"/>
<param name="retry_seconds" value="30"/>
</gateway>
</include>
/dialplan/public.xml:
<extension name="Avaya">
<condition field="destination_number" expression="^(3823)$">
<action application="transfer" data="5060 XML default"/>
</condition>
</extension>
and here's the debug info:
------------------------------------------------------------------------
INVITE sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0
Call-ID: 05c8e85a1a5dd14ea549c5a6a00
CSeq: 1 INVITE
From: "Station 216" <sip:216@sdc.com: 5061>;tag=05c8e85a1a5dd14da549c5a6a00
Record-Route: <sip:10.0.2.154:5060;lr>,<sip:10.0.2.151:5061;lr;transport=tls
To: "3823" <sip:3823@sdc.com>
Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10
.0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00
Content-Length: 150
Content-Type: application/sdp
Contact: "Station 216" <sip:216@10.0.2.151:5061;transport=tls>
Max-Forwards: 69
User-Agent: Avaya CM/R014x.00.1.731.2
Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS
Accept-Contact: *;+avaya-cm-line=1
Supported: 100rel,timer,replaces,join,histinfo
Alert-Info: <cid:internal@sdc.com>;avaya-cm-alert-type=internal
Min-SE: 1200
Session-Expires: 1200;refresher=uac
P-Asserted-Identity: "Station 216" <sip:216@sdc.com:5061>
History-Info: <sip:3823@sdc.com>;index=1,"3823" <sip: 3823@sdc.com>;index=1.1
v=0
o=- 1 1 IN IP4 10.0.2.151
s=-
c=IN IP4 10.0.2.152
t=0 0
m=audio 3188 RTP/AVP 0 127
a=rtpmap:0 PCMU/8000
a=rtpmap:127 telephone-event/8000
------------------------------------------------------------------------
tport(01833120): msg 01872960 (1194 bytes) from udp/10.0.2.154:5060/ sip next=00000000
nta: received INVITE sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0 (CSeq 1)
nta: canonizing sip:Avaya@10.0.6.114:5060 with contact
nta: INVITE (1) going to a default leg
nta: timer set to 200 ms
soa_clone(static::017FD508, 017FED48, 0188E450) called
soa_set_params(static::0189BB50, ...) called
nta_leg_tcreate(00FEDDE8)
soa_init_offer_answer(static::0189BB50) called
soa_set_remote_sdp(static::0189BB50, 00000000, 0189E804, 150) called
nua(0188E450): adding session usage
tport_tsend(01833120) tpn = UDP/10.0.2.154:5060
tport_resolve addrinfo = 10.0.2.154:5060
tport(01833120): not found by name UDP/10.0.2.154:5060
tport_vsend(01833120): 515 bytes of 515 to UDP/10.0.2.154:5060
tport_vsend returned 515
send 515 bytes to udp/[10.0.2.154]:5060 at 14:09:50.666454:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10
.0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00
Record-Route: <sip:10.0.2.154:5060;lr>
Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
From: "Station 216" <sip:216@sdc.com: 5061>;tag=05c8e85a1a5dd14da549c5a6a00
To: "3823" <sip:3823@sdc.com>
Call-ID: 05c8e85a1a5dd14ea549c5a6a00
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9570M
Content-Length: 0
------------------------------------------------------------------------
nta: sent 100 Trying for INVITE (1)
nua(0188E450): event i_invite 100 Trying
nua(0188E450): call state changed: init -> received, received offer
soa_get_remote_sdp(static::0189BB50, [029EFC10], [029EFC0C], [00000000]) called
nua(0188E450): event i_state 100 Trying
nua(0188E450): sent signal r_respond
nua(0188E450): sent signal r_destroy
nua(0188E450): event i_state dropped
nua(0188E450): recv signal r_respond 407 Proxy Authentication Required
soa_set_params(static::0189BB50, ...) called
soa_clear_remote_sdp(static::0189BB50) called
tport_tsend(01833120) tpn = UDP/10.0.2.154:5060
tport_resolve addrinfo = 10.0.2.154:5060
tport(01833120): not found by name UDP/10.0.2.154:5060
tport_vsend(01833120): 893 bytes of 893 to UDP/10.0.2.154:5060
tport_vsend returned 893
send 893 bytes to udp/[10.0.2.154]:5060 at 14:09:50.697704:
------------------------------------------------------------------------
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10
.0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00
From: "Station 216" <sip:216@sdc.com: 5061>;tag=05c8e85a1a5dd14da549c5a6a00
To: "3823" <sip:3823@sdc.com>;tag=ptgKFvSeyaDrQ
Call-ID: 05c8e85a1a5dd14ea549c5a6a00
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9570M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include- session-description, presence.winfo, message-summary
Proxy-Authenticate: Digest realm="sdc.com", nonce="21e6dda0- ee03-674b-8df1-08731d42fd76", algorithm=MD5, qop="auth"
Content-Length: 0
------------------------------------------------------------------------
nta: sent 407 Proxy Authentication Required for INVITE (1)
nua(0188E450): removing session usage
nua(0188E450): call state changed: received -> terminated
nua(0188E450): event i_state 407 Proxy Authentication Required
nua(0188E450): event i_terminated 407 Proxy Authentication Required
soa_destroy(static::0189BB50) called
nta_leg_destroy(00FEDDE8)
nua(0188E450): recv signal r_destroy
nta_leg_destroy(00000000)
tport_wakeup_pri(01833120): events IN
tport_recv_event(01833120)
tport(01833120) msg 018390F8 from (udp/10.0.6.114:5060) has 341 bytes, veclen = 1
recv 341 bytes from udp/[10.0.2.154]:32772 at 14:09:50.697704:
------------------------------------------------------------------------
ACK sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0
From: "Station 216" <sip:216@sdc.com: 5061>;tag=05c8e85a1a5dd14da549c5a6a00
To: "3823" <sip:3823@sdc.com>;tag=ptgKFvSeyaDrQ
CSeq: 1 ACK
Call-ID: 05c8e85a1a5dd14ea549c5a6a00
Max-Forwards: 69
Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0
Content-Length: 0
------------------------------------------------------------------------
tport(01833120): msg 018390F8 (341 bytes) from udp/10.0.2.154:5060/ sip next=00000000
nta: received ACK sip:Avaya@10.0.6.114:5060;transport=udp SIP/2.0 (CSeq 1)
nta: ACK (1) is going to INVITE (1)
nta: timer set next to 4844 ms
nta: timer I fired, terminate 407 response
incoming_reclaim_all(00000000, 00000000, 029EFEB8)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta: timer not set
TIA,
Gerry
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participants (7)
-
Anthony Minessale
-
Brian West
-
David Knell
-
Gayatri Kulkarni
-
Gerry Hull
-
Michael Jerris
-
Thomas Mangin