[Freeswitch-users] ITSP outbound call gets 503 unavailable

Mario mario_fs at mgtech.com
Mon Sep 20 15:47:14 PDT 2010 

This ITSP account is working for 2 years on a linksys/CICS PBX, SPA962
phones, and any softphone. It works now. One other ITSP works fine in
and out, Only on FS do I have the problem with this ITSP. I have 2 new
items after talking to the ITSP:

1. I noticed I am getting 407 Proxy Authentication Required messages
when I try to make a call. Inbound ok.

2. The ITSP says he sees no return IP address for the call. I am behind
a firewall with NAT and all ports open for testing. The Linksys PBX and
other phones work fine. The other ITSP also works fine with FS. It's
only this one ITSP. I am trying to find keywords on authentication for
the gateway parms if there are any.
Mario

On 09/20/10 15:31, David Ponzone wrote:
> Test the SIP account with a softphone or something.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr <mailto: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 <BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com
> <BLOCKED::http://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 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}$1 at sip.voipyourlife.com
>>> <mailto:sofia/gateway/uid1/1${default_areacode}$1 at sip.voipyourlife.com>"/>
>>>    </condition>
>>>  </extension>
>>>
>>
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> 
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