Thanks for posting! Others on the forum may reply with working configurations, however many of these situations are unique. I would first verify that you have no errors on any interfaces on the Adtran unit. Next, verify that the all network devices between the RTP endpoints, including the Adtran unit, have QoS enabled if real-time and regular data traffic share those links.
I would then change your voice user configuration to include "modem-passthrough" and omit "ring-voltage 70" unless you have determined that it is necessary for inbound calls. Lastly, I would specify a codec for this voice user so that it starts outbound calls on G.711 only. Below is an example.
voice codec-list 711
voice user 1001
connect fxs 0/1
We can then check the network performance with "show voice quality-stats". It's important to note that some statistics, particularly the lost packets, can be improperly reported if the RTP sequence number is not consistent from the far end for the duration of the call. Below is an example of a voice call followed by a fax/modem call.
Start Lost Discard Delay
ID Time From To Duration Codec Pkts Pkts Avg Max
1964 8:38 AM 9001 2125 0:30 G729 0 0 80 100
1965 8:38 AM 9001 2126 2:29 G711u 0 0 50 50
For the fax/modem call, you should see the average and maximum delay at 50ms. This is because a call in which fax tones are detected will switch to a fixed 50ms jitter buffer. However, the normal phone calls can give us clues about network performance. In a network with low delay and jitter, they normal phone calls will stay with the 50ms jitter buffer. However, as shown in the example above, increasing delay indicates that the adaptive jitter buffer increased due to excessive jitter. In general, faxing becomes unreliable when network jitter forces the jitter buffer to enlarge to greater than 70ms for non-fax calls. However, values slightly higher than 50ms may help us determine static jitter buffer recommendations. Let us know if you would like us to review the output of "show voice quality-stats" from your unit.
We would then capture the following debug output for a given test call.
debug sip stack message
debug voice verbose
debug int fxs
In particular, we will be looking for what codec is being used, what fax/modem tones, if any, were detected, and if a Re-Invite was sent by either party. If you would like us to review the voice quality statistics or your test call debug output, you may need to create a regular support ticket. Otherwise, just make sure you do a find and replace so that customer IP addresses and phone numbers are removed from the log.
Thanks for the reply. Enabling Modem Pass through worked for me.
Glad to hear that took care of the issue. I'm going to go ahead and mark this post as assumed answered, but let us know if you have any further questions.
I went ahead and flagged the "Correct Answer" on this post to make it more visible and help other members of the community find solutions more easily. If you don't feel like the answer I marked was correct, feel free to come back to this post and unmark it and select another in its place with the applicable buttons. If you still need assistance, we would be more than happy to continue working with you on this - just let us know in a reply.