Test the SIP account with a softphone or something. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 20/09/2010 à 23:42, Mario a écrit :
Brand new FS user here. Here is an additional info for my issue of not being able to make outbound calls. I searched and found a "sofia loglevel all 9" command and found that when I make a call I get the 503 service not available message. Inbound works fine. Anyone have an idea? Many thanks!
nua(0x8135520): recv signal r_respond 503 Service Unavailable
nua: nua_stack_set_params: entering soa_set_params(static::0x812ecc8, ...) called nua: nua_invite_server_respond: entering soa_clear_remote_sdp(static::0x812ecc8) called tport_tsend(0x817aa88) tpn = UDP/10.246.1.27:5060 tport_resolve addrinfo = 10.246.1.27:5060 tport_by_addrinfo(0x817aa88): not found by name UDP/10.246.1.27:5060 tport_vsend(0x817aa88): 838 bytes of 838 to udp/10.246.1.27:5060 tport_vsend returned 838 nta: sent 503 Service Unavailable for INVITE (102) nta: timer shortened to 500 ms nua(0x8135520): removing session usage nua(0x8135520): call state changed: received -> terminated nua(0x8135520): event i_state 503 Service Unavailable nua(0x8135520): event i_terminated 503 Service Unavailable soa_destroy(static::0x812ecc8) called nta_leg_destroy(0x81325f0) nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x8135520): sent signal r_destroy nua(0x8135520): recv signal r_destroy nta_leg_destroy((nil)) tport_wakeup_pri(0x817aa88): events IN tport_recv_event(0x817aa88) tport_recv_iovec(0x817aa88) msg 0x81715a8 from (udp/10.246.1.37:5060) has 638 bytes, veclen = 1 tport_deliver(0x817aa88): msg 0x81715a8 (638 bytes) from udp/10.246.1.27:5060/sip next=(nil) nta: received ACK sip:914082558194@10.246.1.37 SIP/2.0 (CSeq 102) nta: ACK (102) is going to INVITE (102) nta: timer set next to 2305 ms nta: timer I fired, terminate 407 response incoming_reclaim_all((nil), (nil), 0xb54550ec) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/3 term, 1/3 free nta: timer set next to 2206 ms nta: timer I fired, terminate 503 response incoming_reclaim_all((nil), (nil), 0xb54550ec) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free
On 09/20/10 14:18, Mario wrote:
For this ITSP, registration is ok and inbound calls work. But when making an outbound call the call is sent to the itsp but then gets a fast busy. From what I see in the trace the itsp uses asterisk. I sent a trace in a previous email but it shows nothing I could see missing or in error. The itsp sent the config info below but I have been using them without a STUN server fine on my old PBX. Is there anything that looks like it is out of sync with defaults (like port?). Also, the SIP and DP are below but since the call is sent the DP must be ok. Thanks for any help/suggestions!
From the ITSP:> Sip Proxy - sip.voipyourphone.com STN Server - stun.voipyourphone.com Registration Port - any one port between 5060 and 5069 DTMF port 5004 RTP ports 16384 to 16482 Audio Codecs G711u (account default), G711a, G729a and iLbc DTMF rfc2833 (default), inband, SIP-INFO
SIP: <gateway name="uid1"> <param name="proxy" value="sip.voipyourphone.com"/> <param name="username" value="uid1"/> <param name="from-user" value="uid1"/> <param name="password" value="pwd1"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/>
DP <extension name="local.uid1"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^9(\d{7})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/uid1/1${default_areacode} $1@sip.voipyourlife.com"/> </condition> </extension>
_______________________________________________ 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