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tony at softins.clara....
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 Posted: Sun May 18, 2008 3:14 am Post subject: [asterisk-users] One way sound when Using Dial cmd without & |
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In article <28749f210805170447w7e2da378vb11d12bdf8dd4b81 at mail.gmail.com>,
Mohammad A. Navid <manavid at gmail.com> wrote:
| Quote: |
I'm implementing a simple calling card feature for testing purpose. I have a
DID number, when I called my DID number and enter the phone number to call,
Asterisk would dial the number for me but the sound was only one way.
After hours of struggling with the problem, I found out that I need to add
"t" to my dial options, this is the correct way of dialing out:
-> Dial(SIP/carrier/3105555555|20|t)
Now I need to know what was going on? Why with option "t" both parties can
hear each other, but without option "t" in dial cmd only one party could
hear?
Another interesting issue is, if I use Answer() command at the begining the
sound becomes one way even if I use "t" in options.
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Try adding "reinvite=no" to the sip.conf or users.conf definition for your
SIP service provider.
Cheers
Tony
--
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org |
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tony at softins.clara....
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| Posted: Sun May 18, 2008 4:20 pm Post subject: [asterisk-users] One way sound when Using Dial cmd without & |
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In article <28749f210805180219g21ae2e1esbfa8ceaf61d5e223 at mail.gmail.com>,
Moe Navid <manavid at gmail.com> wrote:
| Quote: | Thanks Tony for you reply.
|
Did my suggestion fix the problem?
Ah yes, I just noticed you said it in the subject line.
| Quote: | Do you have any idea why Asterisk require "t" in Dial command?
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Yes, "t" specifies that the called party may transfer the call by pressing #
(or some other sequence defined in features.conf). Likewise "T" says the
same about the calling party.
In either case, Asterisk needs to remain in the media path so that it can
listen for the DTMF.
If neither option is specified, Asterisk may try to optimise itself out of
the media path by getting the two SIP endpoints to talk to each other
directly. I think this is what is happening when you get one-way audio,
since the two endpoints may not know how to reach each other directly
(particularly if NAT is involved).
Putting "reinvite=no" in sip.conf for either endpoint (but best to do it
for the Service Provider endpoint) tells Asterisk never to optimise
itself out of the media path, even if "t" and "T" are not specified.
Cheers
Tony
| Quote: | Cheers,
Moe
On Sun, May 18, 2008 at 1:14 AM, Tony Mountifield <tony at softins.clara.co.uk>
wrote:
| Quote: | In article <28749f210805170447w7e2da378vb11d12bdf8dd4b81 at mail.gmail.com>,
Mohammad A. Navid <manavid at gmail.com> wrote:
| Quote: |
I'm implementing a simple calling card feature for testing purpose. I
|
have a
| Quote: | DID number, when I called my DID number and enter the phone number to
|
call,
| Quote: | Asterisk would dial the number for me but the sound was only one way.
After hours of struggling with the problem, I found out that I need to
|
add
| Quote: | "t" to my dial options, this is the correct way of dialing out:
-> Dial(SIP/carrier/3105555555|20|t)
Now I need to know what was going on? Why with option "t" both parties
|
can
| Quote: | hear each other, but without option "t" in dial cmd only one party could
hear?
Another interesting issue is, if I use Answer() command at the begining
|
the
| Quote: | sound becomes one way even if I use "t" in options.
|
Try adding "reinvite=no" to the sip.conf or users.conf definition for your
SIP service provider.
Cheers
Tony
--
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org
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--
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org |
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