Folks, I'm hoping someone has already run into this ... the only other complaint I've seen is here:
http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000640.html and that basically was a problem with the /etc/hosts ... my server is definitely described in my hosts file. I've been using asterisk with IAX and a voicepulse connect number. No problems at all receiving calls. Now, I've just purchased a DID in Canada from another provider, and their proxy only supports SIP. So, following the generic instructions I've found off the web, I set up my SIP.conf to point to voicepulse's server, and set up the other DID to point into this newly defined sip context, i.e., to uid:[EMAIL PROTECTED]/888 The problem? The remote DID, when called, simply gives me a busy signal. Also, on the asterisk console, I'm seeing these messages that don't tell me anything: -------------------------------- May 1 18:37:09 WARNING[12065]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (Critical Request) May 1 18:37:23 NOTICE[12065]: chan_sip.c:4036 sip_reg_timeout: -- Registration for '[EMAIL PROTECTED]' timed out, trying again May 1 18:37:23 DEBUG[12065]: chan_sip.c:4150 transmit_register: Scheduled a registration timeout # 5 May 1 18:37:29 WARNING[12065]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (Critical Request) May 1 18:37:43 NOTICE[12065]: chan_sip.c:4036 sip_reg_timeout: -- Registration for '[EMAIL PROTECTED]' timed out, trying again May 1 18:37:43 DEBUG[12065]: chan_sip.c:4150 transmit_register: Scheduled a registration timeout # 7 -------------------------------- It looks like the remote DID is failing to register with the voicepulse server. Any hints on what could be the problem? If it helps, here is the relevant portion of my sip.conf file. [general] ;context=default ; Default context for incoming calls context=unwelcome-calls ; Default context for incoming calls ; After all, we don't want any random ; incoming calls to have access to outbound ; calling - Maya Kurup, May 1, 2005 ;recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) ;realm=mydomain.tld ; Realm for digest authentication ; defaults to "asterisk" ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) ... register => uid:[EMAIL PROTECTED] ; We need to allow at least incoming calls to ; accept calls via libretel, etc. ; So, let's add a context for that: [888] ; For incoming calls ONLY type=user ; This device takes incoming calls username=uid ; Username on device secret=secret ; Password for device host=srvr.voicepulse.com ; This host will not ; change frequently context=allowed_context ; Inbound calls from ; this host go ; to the normal context ----- and I have allowed_context described in my extensions.conf, it's the same one I'm using for regular IAX incoming calls, and works fine. The context for unwelcome-calls is as follows: [unwelcome-calls] ; ; Take unknown callers that may have found ; our system, and send them to a re-order tone. ; The string "_." matches any dialed sequence, so all ; calls will result in the Congestion tone application ; being called. They'll get bored and hang up eventually. ; exten => _.,1,Congestion --------------- Any help would be appreciated. Thanks, Maya __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
