Sipgate outgoing call no sound problem
Sipgate outgoing call no sound problem
Actually, I can make outgoing call with my Sipgate account directly without any problem. However, I cannot use this Sipgate account via MySIPSwitch. The registration is successful but when making an outgoing call, it can dial a number but after connection, it has no sound.
My Orbtalk account and other accounts (except Sipgate.co.uk) via MySIPSwitch are working well.
I have tried to input several information associated with this Sipgate account in MySIPSwitch profile but the problem still exists. How should I do?
Thank you very much in advance.
My Orbtalk account and other accounts (except Sipgate.co.uk) via MySIPSwitch are working well.
I have tried to input several information associated with this Sipgate account in MySIPSwitch profile but the problem still exists. How should I do?
Thank you very much in advance.
I guess I have an identical problem. When I use following setting for my inbound calls, I have a good connection with sound for both incoming and outgoing:
"user@sip.mysipswitch.com"
but when I use the following I have NO outgoing sound although incoming works fine:
"user@myIPaddress:5060"
The monitor shows that the call is handled fine, no error at all, so I guess the connection is set-up properly in both situations.
I would like to use the last option but I really cannot figure out why there is no sound in the outgoing direction. I have tried to set a STUN server but no success either. Does anyone have any suggestions?
Thanks.
"user@sip.mysipswitch.com"
but when I use the following I have NO outgoing sound although incoming works fine:
"user@myIPaddress:5060"
The monitor shows that the call is handled fine, no error at all, so I guess the connection is set-up properly in both situations.
I would like to use the last option but I really cannot figure out why there is no sound in the outgoing direction. I have tried to set a STUN server but no success either. Does anyone have any suggestions?
Thanks.
Similar Blank outgoing calls issue with SMSDiscount.com
Has anyone figured out the issue? I can receive calls normally. But when I dial out, I don't hear anything but the phone that I dialed rings up and even if someone picks up, they can hear me but I don't hear anything - Not the Ring nor the person speaking.
Any help is greatly appreciated.
Thanks.
Sekhar..
Any help is greatly appreciated.
Thanks.
Sekhar..
hi
did you resolve this issue?
i have the same problem with freephoneline, when i dial i can hear the ring tone, the moment it's picked up there is a dead silence, but i'm told that i could be heard without any problems.
on the other hand, received calls seem to be fine.
i have the 5060 and the other ports required by freephoneline open.
thanks
did you resolve this issue?
i have the same problem with freephoneline, when i dial i can hear the ring tone, the moment it's picked up there is a dead silence, but i'm told that i could be heard without any problems.
on the other hand, received calls seem to be fine.
i have the 5060 and the other ports required by freephoneline open.
thanks
Hi all,
I have the same problem.
My sip provider is wengo.fr.
The tricky thing is I made call to China, I can hear the other side very clear but when I try France, the quality was so pool. Of cause both of them couldn’t hear me.
And I try the test call of wengo.fr service : 333, I can hear myself, but the quality was terribly bad.
By the way, may I ask Carlo, where did you change the setting as you said :
I guess I have an identical problem. When I use following setting for my inbound calls, I have a good connection with sound for both incoming and outgoing:
"user@sip.mysipswitch.com"
but when I use the following I have NO outgoing sound although incoming works fine:
"user@myIPaddress:5060"
Thanks!
I have the same problem.
My sip provider is wengo.fr.
The tricky thing is I made call to China, I can hear the other side very clear but when I try France, the quality was so pool. Of cause both of them couldn’t hear me.
And I try the test call of wengo.fr service : 333, I can hear myself, but the quality was terribly bad.
By the way, may I ask Carlo, where did you change the setting as you said :
I guess I have an identical problem. When I use following setting for my inbound calls, I have a good connection with sound for both incoming and outgoing:
"user@sip.mysipswitch.com"
but when I use the following I have NO outgoing sound although incoming works fine:
"user@myIPaddress:5060"
Thanks!
Carlo wrote:I guess I have an identical problem. When I use following setting for my inbound calls, I have a good connection with sound for both incoming and outgoing:
"user@sip.mysipswitch.com"
but when I use the following I have NO outgoing sound although incoming works fine:
"user@myIPaddress:5060"
The monitor shows that the call is handled fine, no error at all, so I guess the connection is set-up properly in both situations.
I would like to use the last option but I really cannot figure out why there is no sound in the outgoing direction. I have tried to set a STUN server but no success either. Does anyone have any suggestions?
Thanks.
Hi fengfr,
My SIP Switch cannot influence audio quality as we deal with SIP only, it doesn't get into the audio path.
Have you specified a STUN server on your IP phone , softphone or ATA ?
My SIP Switch cannot influence audio quality as we deal with SIP only, it doesn't get into the audio path.
Have you specified a STUN server on your IP phone , softphone or ATA ?
is in the "Providers" settings, under the "Contact" field, if you tick 'Register with this provider'.
Blueface [url=http://www.blueface.ie/]Phone[/url] Service
Thanks a lot for your reply.
Here the stun I tried : stun.wengo.fr:3478
But changed nothing.
And about the “Providers” is it in the dial plan? Here is mine:
#Ruby
# Dial Plan Generated by Rubyzard v0.1
# If you need help, please post in our forum
# http://www.mysipswitch.com
# SIP tracing : true or false
sys.Trace = false
sys.Log("call from #{req.Header.From.FromURI.ToString()} to #{req.URI.User}.")
if sys.In then
# Do your INCOMING call processing customisations here.
if sys.IsAvailable() then
sys.Dial("#{sys.Username}@local",30)
sys.Respond(480, "#{sys.Username} Not available")
else
sys.Dial("Enter Number@Neuftalk",30)
sys.Respond(480, "#{sys.Username} Not available")
end
else
# Do your OUTGOING call processing customisations here.
sys.Dial("Neuftalk")
end
Here the stun I tried : stun.wengo.fr:3478
But changed nothing.
And about the “Providers” is it in the dial plan? Here is mine:
#Ruby
# Dial Plan Generated by Rubyzard v0.1
# If you need help, please post in our forum
# http://www.mysipswitch.com
# SIP tracing : true or false
sys.Trace = false
sys.Log("call from #{req.Header.From.FromURI.ToString()} to #{req.URI.User}.")
if sys.In then
# Do your INCOMING call processing customisations here.
if sys.IsAvailable() then
sys.Dial("#{sys.Username}@local",30)
sys.Respond(480, "#{sys.Username} Not available")
else
sys.Dial("Enter Number@Neuftalk",30)
sys.Respond(480, "#{sys.Username} Not available")
end
else
# Do your OUTGOING call processing customisations here.
sys.Dial("Neuftalk")
end
gbonnet wrote:Hi fengfr,
My SIP Switch cannot influence audio quality as we deal with SIP only, it doesn't get into the audio path.
Have you specified a STUN server on your IP phone , softphone or ATA ?
is in the "Providers" settings, under the "Contact" field, if you tick 'Register with this provider'.