Audio problems behind Draytek VDSL modem (Draytek 2750)


One of our users has 3 grandstream phones behind a Draytek 2750 VDSL modem and they are experiencing audio difficulties on internal calls.

I’ve searching for the solutions for days now… I hope someone can help me. Other users are not having this problem.

The situation is as follows:

When i recieve a call for this user it is forwarded to all SIP phones, i someone answers there is audio however if they try to transfer the call to one of the other phones. Audio stops…

I’m working on a Asterisk 1.8 cert4 version wich is hosted in a datacenter without nat. The users allready have directmedia=no and canreinvite=no

I tought it would be a NAT issue so i’ve tried nat=yes and nat=force_rport,comedia

Can somebody help me?

Best regards,

Marcel van Dijk

This is the peer info on one of the phones:

* Name : 25131 Realtime peer: Yes, cached Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : 31 Subscr.Cont. : 31 Language : nl Accountcode : 31 AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : 31 Pickupgroup : MOH Suggest : Mailbox : voicemail31 VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 8 Max forwards : 0 Dynamic : Yes Callerid : "Harry" MaxCallBR : 384 kbps Expire : 857 Insecure : no Force rport : Yes ACL : No DirectMedACL : No T.38 support : Yes T.38 EC mode : FEC T.38 MaxDtgrm: 400 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : auto Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : X.X.X.X:5062 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 25131 SIP Options : (none) Codecs : 0xc (ulaw|alaw) Codec Order : (ulaw:20,alaw:20) Auto-Framing : No Status : OK (47 ms) Useragent : Grandstream GXP2100 Reg. Contact : sip:25131@ Qualify Freq : 60000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No

And the general section of sip.conf

[general] port=5060 srvlookup=yes disallow=all allow=alaw allow=ulaw dtmfmode=auto nat=never bindaddr = udpbindaddr = externip=X.X.X.X context=incoming language=nl t38pt_udptl = yes,fec,maxdatagram=400 callcounter=yes limitonpeer=yes directmedia=yes rtcachefriends=yes rtupdate=yes notifyringing=yes

The extentions.conf section for this user looks like this:

exten=>200,hint,SIP/25135 exten=>200,1,Dial(SIP/25135) exten=>200,n,Hangup exten=>201,hint,SIP/25131 exten=>201,1,Dial(SIP/25131) exten=>201,n,Hangup exten=>202,hint,SIP/25132 exten=>202,1,Dial(SIP/25132) exten=>202,n,Hangup exten=>203,hint,SIP/25136 exten=>203,1,Dial(SIP/25136) exten=>203,n,Hangup

How are they doing the transfer:

  • features.conf
  • SIP with true blind transfer
  • SIP with blind transfer implemented as attended transfer

Is there any possibility that the phones are trying to transfer directly, bypassing Asterisk (only applies to SIP transfers)? (Phone makes enquiry phone to phone to the other phone then sends a REFER referencing the IP address of the target phone.)

Otherwise you need to provide details of the SIP and SDP exchanged.

Note, though, that nat= applies vaious hacks that may be needed in some NAT cases, but is not sufficient and may not be necessary for the most common NAT case. You seem to be saying that you don’t have the most common NAT case.

Hi David,

Thanks for your reply, the users transfer by the featers.conf methode for atxfer.

I can’t think of any mechanism that doesn’t involve direct media, as the operation should be entirely internal to the PABX. It would still be worth providing the SIP traces.


make sure they have the sip alg turned off in the router and I assume they have NO port forwarding set.

run rtp debug on their ip to see where the stream is going, alson teh router get them to look at teh nat sessions.

Drayteks of late have proven to be a bit broken.