[Freeswitch-users] ITSP outbound call gets 503 unavailable

[David Ponzone]

Test the SIP account with a softphone or something.

David Ponzone  Direction Technique
email: david.ponzone at 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

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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 at 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 at sip.voipyourlife.com"/>
>>    </condition>
>>  </extension>
>>
>
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