[Asterisk-Users] Unable to process inband DTMF on 2 frames and other messages
Hi, I have Asterisk up and running with one FXO, one FXS and 2 SIP clients (X-Lite). One SIP client is running on laptop (wired connection), and another on PocketPC (5550, WLAN connection). I have dialplan which enables all internal (each user with each user internally) and also external calls via FXO (each client can call outside line). Voice mail, music on hold (MP3) etc all are ok. Still, I have several problems: a) During SIP calls I have the following messages scrolling: dsp.c:1452 ast_dsp_process: Unable to process inband DTMF on 2 frames Does anyone know how to deal with this to eliminate this message? b) When there is no activity in the system, I have another message scrolling (every 15sec or every 1min): chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call some long string or hex numbers@10.10.1.101 for seqno 102 (Request) where 10.10.1.101 is my Asterisk box. Again, does anyone know how to eliminate this message? In addition I have some quality issues with PocketPC calls but I suppose that can not be resolved easily with X-Lite version of the soft client and especially over 802.11b integrated connection. Thanks in advance for any help. Goran ___ 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
[Asterisk-Users] X-Lite on PocketPC or WindowsXP
Hi, I just downloaded X-Lite for PocketPC and I am wondering did anyone have it working with Asterisk? I suppose it will be ok, but when I start the application on my PocketPC (iPaq 5550) there is no Manu button to configure the settings for SIP. Can someone help with this issue. Thanks. Also did someone configure it on laptop? Any config files? Goran ___ 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
[Asterisk-Users] Messanger 6.2 with Asterisk
I am trying to configure Messenger 6.2 with * but all the notes how to do it are for older 4.x version of Messenger. Basically, on 6.2 there is no Accounts tab so I cant configure services for account on asterisk. Does anyone have 6.2 working with *. Thanks. Goran ___ 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
[Asterisk-Users] Outgoing call fails on pulse dial line
I have Digium FXO/FXS card and one of my phone lines is with pulse dialing. At first I didnt have dial tone at all, but after upgrading my Asterisk and Zaptel SW to the latest one (1.0.2) I have dial tone. But, when I try to dial outgoing number it fails after first key pressed. Does anyone know how to solve this? I am in Eastern Europe (Belgrade, Serbia) and our phone lines are mixture of old (pulse) and new ones. So I have 2 lines in my house, one with tone dialing and one with pulse dialing. Is this related to some signaling settings in Zapata.conf? Thanks, Goran ___ 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
