Am using the latest * but have an issue with voice. When i call someone on a GSM network, using ulaw, the call goes thru Ok but neither parties can hear each other. The call goes silent.
Is it an issue with codecs? Ive allowed all codecs. Where can i purchase a g723.1 license?
is this only when calling cell phones? can you call other phones at all?
It could be a NAT issue… are you behind a NAT? did you set up externip=, localnet= and forward ports?
I am behind a NAT firewall. Ive only set NAT=No The rest of ive not put anything. Let me try fill in the ips then revert back. Yes the problem is only with GSM network calls.
sounds like it might be a problem with your provider then…
My Asterisk sits behind a NAT firewall and runs on a private ip 192.168.1.*…on my NAT settings ive enbaled the externip n local ip classes.
Am using ulaw n alaw codecs since i havent bought any licenses for g729 n g723.1 This time round ive made some progress.
When i call the GSM network number, the call goes thru’ and i can hear the GSM side but the Mobile guy who i callled from the IP side cannot hear me? What might be the problem?
Below is part of my Debug Log file
[quote]2006-09-11 19:37:02 DEBUG: chan_sip.c:1883 create_addr_from_peer: Setting NAT on RTP to 524288
2006-09-11 19:37:02 DEBUG: acl.c:211 ast_apply_ha: ##### Testing 192.168.1.87 with 192.168.1.0
2006-09-11 19:37:02 DEBUG: chan_sip.c:2077 sip_call: Outgoing Call for 0722636102
2006-09-11 19:37:02 WARNING: res_musiconhold.c:900 local_ast_moh_start: No class: acc_1
2006-09-11 19:37:02 DEBUG: chan_sip.c:1463 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on ‘firstname.lastname@example.org’ Request 102: Found
2006-09-11 19:37:05 DEBUG: chan_sip.c:1332 __sip_autodestruct: Auto destroying call 'c0a80157-13c4-45059122-2d663-ecd’
2006-09-11 19:37:06 DEBUG: chan_sip.c:1463 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on ‘email@example.com’ Request 102: Found
2006-09-11 19:37:11 DEBUG: chan_sip.c:1388 __sip_ack: Acked pending invite 102
2006-09-11 19:37:11 DEBUG: chan_sip.c:1410 __sip_ack: Stopping retransmission on ‘firstname.lastname@example.org’ of Request 102: Match Found
2006-09-11 19:37:11 DEBUG: chan_sip.c:6145 build_route: build_route: Contact hop: sip:email@example.com:5060
2006-09-11 19:37:46 DEBUG: channel.c:3321 ast_generic_bridge: Didn’t get a frame from channel: SIP/192.168.1.14-098deaf0
2006-09-11 19:37:46 DEBUG: channel.c:3604 ast_channel_bridge: Bridge stops bridging channels SIP/192.168.1.14-098deaf0 and SIP/3445-0990ebb8
2006-09-11 19:37:46 DEBUG: chan_sip.c:2432 sip_hangup: update_call_counter(0722636102) - decrement call limit counter
2006-09-11 19:37:46 DEBUG: app_dial.c:1635 dial_exec_full: Exiting with DIALSTATUS=ANSWER.
a2billing.php|1: line:930 - -> dialstatus : ANSWER, answered time is 35