Problem is definetly on far end. If you look at the siptrace, you have the following sequence: 1. Asterisk calls in 2. FreeSWITCH replies with a Trying(100) to complete call right away and proceeds to dialplan 3. FreeSWITCH invites (calls) 7 times the final destination that never responds. 4. Asterisk sends a CANCEL message In all that, your final endpoint never responds to any message. Are you sure you can reach it? jmesquita On Mon, Aug 31, 2009 at 5:18 PM, e schmidbauer <e.schmidbauer@gmail.com>wrote:
here's the sip trace: http://pastebin.freeswitch.org/10172
2009/8/31 João Mesquita <jmesquita@freeswitch.org>:
2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel sofia/external/Anonymous@anonymous.invalid entering state [terminated][487]
The far end seems to be replying with 487 - Request Terminated...
Nothing wrong on FS, seems to be a problem with your endpoints. Can you enable a sip trace?
jmesquita
On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer <e.schmidbauer@gmail.com> wrote:
thanks...heres the pastebin: http://pastebin.freeswitch.org/10171
2009/8/31 João Mesquita <jmesquita@freeswitch.org>:
Check the password dialog. It will tell you what the username/password is.
post the logs for a call as well, please.
Regards,
jmesquita
On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer <
e.schmidbauer@gmail.com>
wrote:
=================================================================================================
Name external Domain Name N/A DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 192.168.0.125 Ext-RTP-IP 98.118.151.30 SIP-IP 192.168.0.125 Ext-SIP-IP 98.118.151.30 URL sip:mod_sofia@192.168.0.125:5080 BIND-URL sip:mod_sofia@192.168.0.125:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS CELT@48000h,CELT@32000h,speex@32000h@20i,speex@16000h @20i,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 17 FAILED-CALLS-IN 11 CALLS-OUT 9 FAILED-CALLS-OUT 9
Registrations:
=================================================================================================
Call-ID: MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI. User: 1000@server1.altpressonline.com Contact: "1000" <sip:1000@69.204.30.67:16006;rinstance=50cbd6140e1ab991> Agent: X-Lite release 1103k stamp 53621 Status: Registered(UDP)(unknown) EXP(2009-08-31 16:24:11) Host: server1.altpressonline.com IP: 69.204.30.67 Port: 16006 Auth-User: 1000 Auth-Realm: server1.altpressonline.com
=================================================================================================
sorry, i don't know how to login to pastebin
On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<msc@freeswitch.org> wrote:
Could you give a few more details? For example, could you pastebin the output of "sofia status profile external"? -MC
On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer <e.schmidbauer@gmail.com> wrote: > > I am unable to call a user outside of our local area network. the > user > is registered on the external profile but there is no way to call > the > phone. does anyone have any suggestions how to do this? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:
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