1-way audio problem (X-Lite=Asterisk=VoIP Gateway=POTS)

Asterisk Asterisk Support
1

Dear all,

This is my setup:
[X-Lite] <–SIP/RTP–> [Asterisk] <–SIP/RTP–> [Addpac AP1100F VoIP Gateway] <–FXO—FXS–> [GSM gateway] <–> [Mobile phone]

The problem is that the audio only works in one direction.
I am using X-Lite and can hear the person in the mobile phone, but the person in the mobile phone cannot hear me.

OK: [X-Lite] <—AUDIO— [Mobile phone]
DOESN’T WORK: [X-Lite] —AUDIO—> [Mobile phone]

I have tried to route the call to another X-Lite through the Asterisk, and then the audio works fine both ways.

Any clues?

This is my Asterisk configuration:

sip.conf:

[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes

[xlite]
type=friend
username=xlite
context=ap1100f
secret=XXXXXXXXX
host=dynamic
canreinvite=no

extensions.conf:

[ap1100f]
exten => _467XXXXXXXX,1,Dial(SIP/${EXTEN}@192.168.0.10)

IP-addresses:
[X-Lite 192.168.10.X] <—> [192.168.10.2 : Asterisk : 192.168.0.1] <—> [192.168.0.10 AP1100F]

Verbose debug log:

*CLI> Apr 16 15:41:42 DEBUG[9443]: chan_sip.c:2363 sip_alloc: Allocating new SIP call for 32306588-DC41-471B-8EBE-76271D8046F8@192.168.10.113
Apr 16 15:41:42 DEBUG[9443]: chan_sip.c:5456 check_user_full: Setting NAT on RTP to 0
Apr 16 15:41:42 DEBUG[9443]: chan_sip.c:841 __sip_ack: Stopping retransmission on ‘32306588-DC41-471B-8EBE-76271D8046F8@192.168.10.113’ of Response 39166: Found
Apr 16 15:41:42 DEBUG[9443]: chan_sip.c:5456 check_user_full: Setting NAT on RTP to 0
Apr 16 15:41:42 DEBUG[9443]: chan_sip.c:7354 handle_request: Check for res for xlite
Apr 16 15:41:42 DEBUG[9443]: chan_sip.c:1623 update_user_counter: Call from user ‘xlite’ is 1 out of 0
Apr 16 15:41:42 DEBUG[9443]: chan_sip.c:4643 build_route: build_route: Contact hop: sip:xlite@192.168.10.113:5060
Apr 16 15:41:42 DEBUG[9443]: pbx.c:1274 pbx_extension_helper: Launching ‘Dial’
– Executing Dial(“SIP/xlite-83ab”, “SIP/46701234567@192.168.0.10”) in new stack
Urgent handler
Apr 16 15:41:42 DEBUG[9443]: chan_sip.c:2363 sip_alloc: Allocating new SIP call for (null)
Urgent handler
Apr 16 15:41:43 DEBUG[9443]: chan_sip.c:1490 sip_call: Outgoing Call for 46701234567
Apr 16 15:41:43 DEBUG[9443]: chan_sip.c:1595 update_user_counter: 46701234567 is not a local user
– Called 46701234567@192.168.0.10
Apr 16 15:41:43 DEBUG[9443]: channel.c:1752 ast_set_read_format: Set channel SIP/192.168.0.10-3f5f to read format ulaw
Apr 16 15:41:43 DEBUG[9443]: channel.c:1719 ast_set_write_format: Set channel SIP/xlite-83ab to write format ulaw
Apr 16 15:41:43 DEBUG[9443]: channel.c:1719 ast_set_write_format: Set channel SIP/192.168.0.10-3f5f to write format ulaw
Apr 16 15:41:43 DEBUG[9443]: channel.c:1752 ast_set_read_format: Set channel SIP/xlite-83ab to read format ulaw
Urgent handler
Apr 16 15:41:43 DEBUG[9443]: chan_sip.c:873 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on ‘4f1cc47c310536d530bac9732e8e07e0@192.168.0.1’ Request 102: Found
Apr 16 15:41:43 DEBUG[9443]: chan_sip.c:873 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on ‘4f1cc47c310536d530bac9732e8e07e0@192.168.0.1’ Request 102: Found
– SIP/192.168.0.10-3f5f is making progress passing it to SIP/xlite-83ab
Urgent handler
Apr 16 15:41:43 DEBUG[9443]: chan_sip.c:2230 sip_rtp_read: Oooh, format changed to 2
Apr 16 15:41:43 DEBUG[9443]: channel.c:1752 ast_set_read_format: Set channel SIP/xlite-83ab to read format ulaw
Apr 16 15:41:43 DEBUG[9443]: channel.c:1719 ast_set_write_format: Set channel SIP/xlite-83ab to write format ulaw
Apr 16 15:41:43 DEBUG[9443]: rtp.c:1195 ast_rtp_write: Ooh, format changed from unknown to ulaw
Apr 16 15:41:47 DEBUG[9443]: rtp.c:1195 ast_rtp_write: Ooh, format changed from unknown to gsm
Apr 16 15:41:47 DEBUG[9443]: rtp.c:375 ast_rtcp_read: Got RTCP report of 60 bytes
Apr 16 15:41:53 DEBUG[9443]: rtp.c:375 ast_rtcp_read: Got RTCP report of 60 bytes
Apr 16 15:41:57 DEBUG[9443]: rtp.c:375 ast_rtcp_read: Got RTCP report of 60 bytes
Apr 16 15:41:59 DEBUG[9443]: rtp.c:375 ast_rtcp_read: Got RTCP report of 60 bytes
Apr 16 15:42:06 DEBUG[9443]: rtp.c:375 ast_rtcp_read: Got RTCP report of 60 bytes
Apr 16 15:42:11 DEBUG[9443]: rtp.c:375 ast_rtcp_read: Got RTCP report of 60 bytes
Apr 16 15:42:14 DEBUG[9443]: rtp.c:375 ast_rtcp_read: Got RTCP report of 60 bytes
Apr 16 15:42:16 DEBUG[9443]: channel.c:739 ast_hangup: Hanging up channel 'SIP/192.168.0.10-3f5f’
Apr 16 15:42:16 DEBUG[9443]: chan_sip.c:1711 sip_hangup: sip_hangup(SIP/192.168.0.10-3f5f)
Apr 16 15:42:16 DEBUG[9443]: chan_sip.c:1723 sip_hangup: update_user_counter(46701234567) - decrement outUse counter
Apr 16 15:42:16 DEBUG[9443]: chan_sip.c:1595 update_user_counter: 46701234567 is not a local user
Apr 16 15:42:16 DEBUG[9443]: app_dial.c:1054 dial_exec: Exiting with DIALSTATUS=CANCEL.
Apr 16 15:42:16 DEBUG[9443]: pbx.c:1851 ast_pbx_run: Spawn extension (ap1100f,46701234567,1) exited non-zero on 'SIP/xlite-83ab’
Apr 16 15:42:16 DEBUG[9443]: chan_sip.c:7823 sipsock_read: Failed to grab lock, trying again…
Apr 16 15:42:16 DEBUG[9443]: chan_sip.c:7823 sipsock_read: Failed to grab lock, trying again…
Apr 16 15:42:16 DEBUG[9443]: chan_sip.c:7823 sipsock_read: Failed to grab lock, trying again…
Apr 16 15:42:16 DEBUG[9443]: chan_sip.c:7823 sipsock_read: Failed to grab lock, trying again…
Apr 16 15:42:16 DEBUG[9443]: channel.c:739 ast_hangup: Hanging up channel 'SIP/xlite-83ab’
Apr 16 15:42:16 DEBUG[9443]: chan_sip.c:1711 sip_hangup: sip_hangup(SIP/xlite-83ab)
Apr 16 15:42:16 DEBUG[9443]: chan_sip.c:1726 sip_hangup: update_user_counter(xlite) - decrement inUse counter
Apr 16 15:42:16 DEBUG[9443]: chan_sip.c:841 __sip_ack: Stopping retransmission on ‘32306588-DC41-471B-8EBE-76271D8046F8@192.168.10.113’ of Response 39167: Found
Apr 16 15:42:16 DEBUG[9443]: chan_sip.c:2363 sip_alloc: Allocating new SIP call for 32306588-DC41-471B-8EBE-76271D8046F8@192.168.10.113
Apr 16 15:42:16 DEBUG[9443]: chan_sip.c:823 __sip_ack: Acked pending invite 102
Apr 16 15:42:16 DEBUG[9443]: chan_sip.c:841 __sip_ack: Stopping retransmission on ‘4f1cc47c310536d530bac9732e8e07e0@192.168.0.1’ of Request 102: Found
Apr 16 15:42:16 DEBUG[9443]: chan_sip.c:841 __sip_ack: Stopping retransmission on ‘4f1cc47c310536d530bac9732e8e07e0@192.168.0.1’ of Request 102: Found
Apr 16 15:42:16 DEBUG[9443]: chan_sip.c:1595 update_user_counter: 46701234567 is not a local user

Addpac 1100F:

AP1100F# show call active all
Total calls : 1

Call Number : 16
status = CalleeInitiated
start time = Apr 16 14:42:34
Calling party information :
endpoint type = SIP
address (NAME) = 192.168.0.1
address (IP) = 192.168.0.1
address (NUMBER) = asterisk
session type = voice
codec type = PCMU
local RTP port = 23032
local RTCP port = 23033
remote RTP port = 16732
remote RTCP port = 16733
Media packet information :
VAD = enabled
transmitted RTP packets = 523
transmitted RTCP packets = 3
transmitted T38 packets = 0
received RTP packets = 0
received RTCP packets = 0
received T38 packets = 0
Called party information :
endpoint type = FXO
address = 46T
port = 0/0(0)

2

IP-addresses:
[X-Lite 192.168.10.X] <—> [192.168.10.2 : Asterisk : 192.168.0.1] <—> [192.168.0.10 AP1100F]

Can u explain your setup more detailed ?
How/why has asterisk 2 IP adresses ?
Asterisk is no router.

3

I got it to work.
I upgraded the firmware on the AP1100F and configured it from scratch.

Finally! :smile:

Btw, the Asterisk in my setup is terminating traffic from VoIP to the PSTN via the Addpac AP1100F device. That is why it has two IP-addresses. (The Addpac is connected directly to the Asterisk second Ethernet interface, since it shouldn’t be possible to connect to it directly from the Internet. You can achieve the same thing using firewall settings, but I thought this was an easier approach)