Hi,<br><br>Changing the line bellow helped to get incoming calls but I add to remove secret= option in sip.conf (otherwise, Patton wouldn't respond to 407 Auth required challenges).<br>If someone could enable secret and still get incoming calls (in any SmartWare 5.X), please, do not hesitate to share here ...<br>
<pre> interface sip IF-ASTERISK<br> bind context sip-gateway ASTERISK<br> route call dest-table calls_from_SIP<br> remote 192.168.100.254 5060</pre><br>Cheers<br><br><div class="gmail_quote">2009/2/25 Olivier <span dir="ltr"><<a href="mailto:oza-4h07@myamail.com">oza-4h07@myamail.com</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi,<br><br>I'm trying to configure a 4638 to pass inbound and outbound to and from ISDN and SIP interfaces.<br>
I'm using web interface at the moment.<br><br>Setup is:<br><br>ISDN -- <BRI> -- Patton 4638 -- <SIP> Asterisk -- <SIP> -- <IP Phone><br>
<br>I can call from IP phone but can't receive any incoming call : I can't see any SIP message coming in when a call comes in.<br><br>Previously, with 4.2 firmware, you just have to edit routing table binding ISDN ports to SIP interface to get calls coming in but now with 5.3, configuration process changed.<br>
Here is an extract from my running config.<br>Any idea ?<br><br>Regards<br><br><pre>context cs switch<br><br> routing-table called-e164 appels_provenance_ISDN<br> route [0-9]+ dest-service ASTERISK_SRV<br> route default dest-service ASTERISK_SRV<br>
<br> routing-table called-uri appels_vers_ISDN<br> route default dest-service isdnports<br><br> mapping-table called-e164 to called-ip transfo<br> map [0-9]+ to 192.168.100.254<br><br> mapping-table called-e164 to called-uri transfo2<br>
<br> interface isdn IF-PBX<br> route call dest-table appels_provenance_ISDN<br><br> interface isdn IF-PBX2<br> route call dest-table appels_provenance_ISDN<br><br> interface isdn IF-PBX3<br> route call dest-table appels_provenance_ISDN<br>
<br> interface isdn IF-PBX4<br> route call dest-table appels_provenance_ISDN<br><br> interface sip IF-ASTERISK<br> bind context sip-gateway ASTERISK<br> route call dest-table appels_vers_ISDN<br><br> service sip-location-service ASTERISK_SRV<br>
bind location-service ASTERISK_SRV<br> mode hunt<br> hunt-timeout 20<br><br> service hunt-group isdnports<br> drop-cause normal-unspecified<br> drop-cause no-circuit-channel-available<br> drop-cause network-out-of-order<br>
drop-cause temporary-failure<br> drop-cause switching-equipment-congestion<br> drop-cause access-info-discarded<br> drop-cause circuit-channel-not-available<br> drop-cause resources-unavailable<br> route call 1 dest-interface IF-PBX<br>
route call 2 dest-interface IF-PBX2<br> route call 3 dest-interface IF-PBX3<br><br>context cs switch<br> no shutdown<br><br>authentication-service patton<br> realm 1 asterisk<br> username patton password Otx2vJCEWP+8Bb6tqoGkwA== encrypted<br>
<br>location-service ASTERISK_SRV<br> domain 1 192.168.100.254 5060<br> domain 2 asterisk 5060<br><br> identity-group default<br> identity patton<br> alias name patton<br><br> authentication outbound<br> authenticate 1 authentication-service patton username patton<br>
<br> registration outbound<br> registrar 192.168.100.254 5060<br> proxy none<br> lifetime 3600<br> register auto<br> retry-timeout on-system-error 10<br> retry-timeout on-client-error 10<br>
retry-timeout on-server-error 10<br><br> call outbound<br> use profile tone-set default<br> use profile voip default<br> use profile sip default<br> preferred-transport-protocol udp<br> invite-transaction-timeout 32<br>
non-invite-transaction-timeout 32<br><br> call inbound<br> use profile tone-set default<br> use profile voip default<br> use profile sip default<br></pre><br>
</blockquote></div><br>