Dear All.

When I calling a number from web, my softphone show me “Answer” and
“Decline” bottoms, and then I have to click Answer to call the number. it seems it is two step to calling the number. If I type the number direct to my client softphone, it calls directlly the number without show me to choose Answer to calling.

My source code is in AMI fsocket open to make call from web. how can I call direct to the number?

Hi All;

How I can acheive the following:

From sip client softphone (from the iPhone for example), if I dialed a number that I need to call it, then a call to be initated to a specific number through DAHDI channel and another call to be initiated for the destination number (the number that I dialed it from the softphone) and these two calls to be linked togethor (to call each other directly). So the call from the softphone just to important in the begining to trigger this scenario.

How this settings to be done?

Regards Bilal

I am having a problem trying to get a particular softphone working on my setup.

The machine it runs on has more than one interface. When the softphone registers, it registers fine, and asterisk is given the correct IP for registration.

Whenever RTP is set-up however, the client gives the wrong IP to connect to and I get the inevitable problem with one-way media.

Is there any way of forcing that SIP account to have the rtp always sent to a particular IP. (I know that this still may not work, because the device is probably listening on the wrong interface as well, but it’s worth a try).

I haven’t been able to get a response from the vendor of the softphone.

All,

We are having issues with one of our customers. They typically are
using remote sip clients on smart phones. For the purpose of allowing
the apps to work properly in the background we have to utilize TCP which
works fine.

The problem comes up when the softphone loses connectivity for some
reason. The session timers are not ending the call as they do on a UDP
session. Basically from the sip debug it sends the re-invite for the
session timer according to the sip debug and it appears all is fine
instead of not getting a response back from the client and disconnecting
the call as it does with udp. There is no way it is getting a response
back from the client however as the client has no network connectivity.

I have run some tcpdump’s on the server and when tracing the call I
actually never see those re-invites going out at all from the server.

We are running asterisk 1.8.7.0 currently.

I can reproduce the issue at will by establishing a call from a
softphone and then putting it into airplane mode to simulate the
connectivity loss.

Are session-timers expected to work with tcp? If so can anyone tell me
where to look to see what might be going on?

Thanks in Advance.

Hi;

In Voicemail.conf 

If am am using

format=h263|gsm ,and i want to store only audio , then it is not storing.In log it shows that video is deposite less then 5 second. If i want to store video and audio both then it will store properly.

If am using  

format=gsm|h263 ,then my Xlite  softphone will go to haung.

I just want to store audio and video both or some time only audio .

1)Plz guide me which combination of codec will be usefull.

2)Is there is any serial number signifance in format,ie one time if i use as format=h263|gsm and second time i am using format=gsm|h263,why  is diffrence  come?

Thanks

Durgesh Mishra

Rancore Technologies.