Hello,
I would like to verify if a specific SIP header exists, and if yes, extract
the partial content from another header.
1. Is there a way to verify if a specific header exists?
2. How do I extract data that is between the first : and the following @?
Specifically, The data looks like
Hello,
I'm trying to send a tif file, using Fax for Asterisk and the call is
executed, but when I get the reINVITE with T.38 data, the local server
doesn't recognize that we have this capability and sends a 488 message.
These are the logs:
--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---
It's set to yes for this peer.
also t38pt_udptl is set to yes.
:(
On Tue, Oct 19, 2010 at 5:12 PM, David Backeberg dbackeb...@gmail.comwrote:
On Tue, Oct 19, 2010 at 10:36 AM, VoIP Question voip.quest...@gmail.com
wrote:
Hello,
I'm trying to send a tif file, using Fax for Asterisk
, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
Thanks.
Michael
On Tue, Oct 19, 2010 at 5:40 PM, David Backeberg dbackeb...@gmail.comwrote:
On Tue, Oct 19, 2010 at 11:21 AM, VoIP Question voip.quest...@gmail.com
wrote:
It's set to yes for this peer.
also
Digium claims that their FFA is the best and most compatible solution and
they give one channel for free, but do not provide support for those that do
not buy more channels, but why buy more channels if the free/test one
doesn't work?
I know they read (and sometimes respond) to this list, so I
Max-Forwards: 70
Content-Length: 0
Thanks,
Michael
On Tue, Oct 19, 2010 at 8:56 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 10/19/2010 12:01 PM, VoIP Question wrote:
Digium claims that their FFA is the best and most compatible solution
and they give one channel for free, but do
Hello all,
We would like to inform the caller of the reason for a failed call.
For example, when we get a 486 Busy Here, the system accepts it and in the
CLI we see Everyone is busy/congested at this time.
Can we use this data to play an announcement to the caller?
Thank you in advance for
Thank you Kevin,
We'll upgrade our server to 1.6.2.12 and try again.
Another question: Is there (expect for the admin guide that we didn't
succeed to understand the example in) an example somewhere for ReceiveFax
full extensions.conf diaplan? We would like to allocate one of the
extensions that
Hello all,
We're trying to build a small IVR application to allow callers to use the
Asterisk for outgoing calls in a 2 steps dialing mode.
The context for outgoing calls is called outgoing (we have there an LCR
and routing mechanism we want to use, depending on the destination).
This is what
Hello again,
If I set a peer to use G.711 only, they try to process a sent fax in G.711,
but Asterisk doesn't like it:
WARNING[4903]: res_fax.c:1709 sendfax_t38_init: Audio FAX not allowed on
channel 'SIP/Main-000a' and T.38 negotiation failed; aborting.
What can I do to enable it?
Thanks,
SendFax rejects T.38 reINVITE (488 Not
acceptable here)
From: Kevin P. Fleming kpflem...@digium.com
To: asterisk-users@lists.digium.com
Date: Thursday, 21 October, 2010 16:11:00
On 10/20/2010 11:35 AM, VoIP Question wrote:
On Wed, Oct 20, 2010 at 4:25 PM, Kevin P. Flemingkpflem...@digium.com
Not
acceptable here)
From: Kevin P. Fleming kpflem...@digium.com
To: asterisk-users@lists.digium.com
Date: Thursday, 21 October, 2010 16:13:02
On 10/20/2010 09:35 AM, VoIP Question wrote:
Thank you Kevin,
We'll upgrade our server to 1.6.2.12 and try again.
Another question: Is there (expect
Hello list members,
We're trying to get MWI notifications on our ATA device and we set it to
send SUBSCRIBE messages to Asterisk, but it gets UNAUTHORIZED messages,
despite the fact that we set the following lines in its settings in
sip.conf:
subscribemwi=yes
mailbox...@from-extensions
We
Hello,
We have a strange situation (asterisk 1.6.2.14), where we get a result for
DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL.
This is the (relevant) test dialplan:
[incoming-private]
exten = _X., n, Dial(SIP/1001,30)
exten = _X., n,
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