Hello list,
I am trying to solve a problem and after unsucessfully chasing forums
and google for some hours, I turn to you in hope of a solution. I feel
it's just a configuration issue but I just can't get my head wrapped
around it.
The situation is basically this: I have an Asterisk connected
Could anyone help me set up Asterisk in such a way that it makes SIP-SIP
transfers using the REFER / NOTIFY method according to RFC-3515 ?
SCANARIO:
- Asterisk registers with PSTN<->SIP VoIP provider "V" (Vonage) as a friend
- Asterisk is located in Europe, Vonage in located US.
- Asterisk acts a
I was able to get a full debug report (packet dump and asterisk debug) for one of these dropped calls and it does seem to be the provider that is at fault. I can see that they stop sending RTP packets to Asterisk when this happens and after a while they send a BYE. I will keep investigating thoug
In addition to having this with my SIP phones, I have also
experienced it with SCCP.
It started when I updated to the 1.2 release of asterisk. At
the time I updated I also switched VoIP providers and thought it was them.
Did you file this as a bug or find a solution to it? Thanks!
Asterisk 1.2.1 installation. It seems that calls are being dropped for no valid reason, completely random, in the middle of the call. I first thought that it was either the network or the VoIP provider dropping packets and confusing Asterisk into hanging up the call.
However I happened to be ru
so make sure that canreinvite is OFF for both.
- Original Message -
From: "Ian Pattison" <[EMAIL PROTECTED]>
To:
Sent: Thursday, April 07, 2005 4:49 AM
Subject: [Asterisk-Users] SIP - SIP Problems
Hi Everybody...
Continuing the litany of problems I'm experiencing
oth.
- Original Message -
From: "Ian Pattison" <[EMAIL PROTECTED]>
To:
Sent: Thursday, April 07, 2005 4:49 AM
Subject: [Asterisk-Users] SIP - SIP Problems
Hi Everybody...
Continuing the litany of problems I'm experiencing with my new system I'm
=etting iss
the SIP phones? I'd
be inclined to turn them both ON to ensure that symmetrical RTP in being
used. Also make sure that canreinvite is OFF for both.
- Original Message -
From: "Ian Pattison" <[EMAIL PROTECTED]>
To:
Sent: Thursday, April 07, 2005 4:49 AM
Subject: [As
D]>
To:
Sent: Thursday, April 07, 2005 4:49 AM
Subject: [Asterisk-Users] SIP - SIP Problems
Hi Everybody...
Continuing the litany of problems I'm experiencing with my new system I'm
=etting issues connecting between SIP phones.
A bit of background... I have an asterisk server ru
On Wed, 2005-04-06 at 14:49 -0400, Ian Pattison wrote:
> Hi Everybody...
>
> Continuing the litany of problems I'm experiencing with my new system I'm
> getting issues connecting between SIP phones.
>
> A bit of background... I have an asterisk server running in a central
> location where I hav
Hi Everybody...
Continuing the litany of problems I'm experiencing with my new system I'm
getting issues connecting between SIP phones.
A bit of background... I have an asterisk server running in a central location
where I have two incoming analog lines connected to FXO ports, two analog
phone
Good day all
We have a asterisk server running sip for about 20 users
We have a client running a unknown sip server in a different country
I phone the guy there and he gave a a account(username+password)
What I want is if a users calls the number of that country it should be
send to the sip server
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