Phone rings but no audio on either side

Asterisk Asterisk Support
1

hi,

i’m using asterisk with openbts. here’s my code. phone rings but there’s no audio on either side. please, help. thanks.

“extensions.conf”

[globals]
cell1 = SIP/IMSI0010xxxxxxxxxx1
cell2 = SIP/IMSI0010xxxxxxxxxx2

[macro-dialGSM]
exten => s,1,Dial(${ARG1},20)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten => s-NOANSWER,2,Goto(incoming,s,1)
exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten => s-BUSY,2,Goto(incoming,s,1)
exten => _s-.,1,Goto(s-NOANSWER,1)
;exten => s,1,Dial(${ARG1})
;exten => s,2,Goto(s-${DIALSTATUS},1)
;exten => s-CANCEL,1,Hangup
;exten => s-NOANSWER,1,Hangup
;exten => s-BUSY,1,Busy(30)
;exten => s-CONGESTION,1,Congestion(30)
;exten => s-CHANUNAVAIL,1,playback(ss-noservice)
;exten => s-CANCEL,1,Hangup

[sip-external]
;exten => 1000,1,Macro(dialGSM,${cell1})
;exten => 2000,1,Macro(dialGSM,${cell2})

exten => 1000,1,Macro(voicemail,${cell1})
exten => 2000,1,Macro(voicemail,${cell2})

“sip.conf”

[IMSI0011xxxxxxxxx1] ; cell1
canreinvite=no
type=friend
allow=gsm
context=sip-external
host=dynamic

[IMSI001011xxxxxxx2] ; cell2
canreinvite=no
type=friend
allow=gsm
context=sip-external
host=dynamic

should i change the voicemail.conf file as well?
thanks in advance! i have no audio at the moment when either phone asnwers.

2

I’m curious if you ever resolved your no audio issue? I am using OpenBTS with Asterisk 1.6.2 and am having a similar situation. I can place a call from a mobile phone (MS) to a SIP phone attached to Asterisk without any issues; audio works full duplex. However, if I place a call from a SIP phone to a mobile phone (MS), I have no audio in either direction. The call completes and works other than audio.

I did notice one thing on your sip.conf file, but I think you may have just truncated it. Did you declare disallow=all before declaring allow=gsm within each of the sip user profiles? I’m not sure if this would cause a codec issue or not, but since your post is a few months old, I’m sure you either resolved the issue or moved on to a different approach.

3

Hi, after 4 years,
I still faced exact the same this audio issue!
MS to SIP Softphone works fine,
but
SIP Softphone to MS or MS to MS have no audio at all, sent nothing on rtp.
no nat issue( as the log shows)

Log with sip debug on & rtp debug on as follows,
please kindly help with this issue, thank you.

MS to SIP phone : dropbox.com/s/w05kcqtsw2hsv … e.log?dl=0
SIP phone to MS : dropbox.com/s/zzxy74h14dmq5 … S.log?dl=0
MS to MS : dropbox.com/s/x5bzuh34vfz0q … S.log?dl=0

4

Hi,

I faced exactly the same problem you had. After some testing and debugging I noticed that SMS took a long time to be delivered, that was because there was an increasing lag between OpenBTS and the antenna, so it came out that audio was working, but it was delayed. Is your server running in a virtual machine with a bridged ethernet adapter ? If so, I suggest to move it to a physical server with a dedicated Gigabit Interface between the server and the antenna. After migration, audio was working sharp and flawlessly during calls between MS.

Regards.