after registering the phones correctly and receiving a "200 o.k." message i can connect to other registered softphones and pstn endpoints [ via an voicepulse account ], but after making the initial connection, i can't hear any sound and i get disconnected after getting the following error message:
chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 41270
i've compiled the stock asterisk tarball on a redhat 7.3 box with a public ip address. the clients are xten lite softphone's running on ibooks with os 10.3.4. the clients are natted behind a linksys wrt54g wireless router running the sveasoft [2] firmware. i'm perplexed, because i can get things to work fine if i use ser/rtpproxy instead of asterisk. i can also connect directly to my voicepulse connect account with the xten softphone and things work great. so i think i have the xten client configured properly and i know that the sveasoft firmware isn't throwing a monkey wrench into the picture. i suppose i could configure ser to "front" asterisk since it appears to deal with the nat, but i'm wondering if i'm missing something basic.
my channel config files look like the following:
sip.conf
[general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference
[2000]
type=friend ; This device takes and makes calls username=2000 ; Username on device secret=supersecret ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip-internal nat=yes canreinvite=no ; Typically set to NO if behind NAT qualify=500
[2001]
type=friend ; This device takes and makes calls username=2001 ; Username on device secret=supersecret2 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip-internal nat=yes canreinvite=no ; Typically set to NO if behind NAT qualify=500
iax.conf
[general] port=5036 bandwidth=low disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference jitterbuffer=no
[voicepulse] context = voicepulse-in secret=topsecrect auth=md5 type=friend host=gw5.voicepulse.com
[1] http://www.google.com/search?q=retrans_pkt:+Maximum+retries+exceeded+on+call++site:http://lists.digium.com&hl=en&lr=&ie=UTF-8&start=10&sa=N
[2] http://www.sveasoft.com/modules/phpBB2/index.php _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
