Displaying 15 results from an estimated 15 matches for "dalgliesh".
2004 Apr 20
1
Re: SIP re-invite
.../usr/src/asterisk/channels/Makefile
chan_sip2.so: chan_sip2.o
cd /usr/src/asterisk
make
make install
I assume that problem is with what did or didn't add to the Makefile
Thank for any help
----- Original Message -----
From: "Olle E. Johansson" <oej@edvina.net>
To: "Glenn Dalgliesh" <asterisk@techhat.com>
Sent: Tuesday, April 20, 2004 1:29 PM
Subject: SIP re-invite
> Could you please test this with my chan_sip2. I have a hunch it will work
with
> that channel.
>
> /Olle
>
2005 Jan 02
1
pridialplan=unknown ?
...gt; Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0:
0 Location: User (0)
> Ext: 1 Progress Description: Calling
equipment is non-ISDN. (3) ]
> [28 10 b1 44 61 6c 67 6c 69 65 73 68 20 47 6c 65 6e 6e]
> Display (len=16) Charset: 31 [ Dalgliesh Glenn ]
> [6c 0c 21 83 34 31 30 37 33 35 38 35 35 30]
> Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> Presentation: Presentation allowed of network
provided number (3) '4102228550' ]
>...
2004 May 11
2
SDP messages relating to rtpmap Question
SDP question if * recieves "a=rtpmap:103 telephone-event/8000" it shouldn't
it send out the same "a=rtpmap:103 telephone-event/8000" to the other side
of the connection? and not something like "a=rtpmap:101
telephone-event/8000"?
Thanks
2004 Oct 05
2
Problems installing app_valetparking
I download app_valetparking.c from
http://www.loligo.com/asterisk/misc/apps/app_valetparking.c
and followed the directions on
http://www.loligo.com/asterisk/misc/apps/app_valetparking.README
I am using asterisk-1.0.0 any suggestions
[root@localhost asterisk]# astxs -install apps/app_valetparking.c
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat
ions -g -Iinclude
2003 Dec 05
3
GrandStream Budgetone Phone & DHCP & General Observations
Symptom: Phone after about 15mins will stop functioning
Problem: DHCP lease renewed but default route dropped
Fix: Assign a static ip and problem is resolved. Upgrade to new firmware once it is released
It turn's out that these phones have a few issue in 1.0.3.81 firmware. The phone may stop transmitting packets if configured with DHCP, if DHCP is being provided by certain devices. Netopia
2003 Dec 10
1
sip.conf and Codecs
I have been doing some testing and have found issue with certain devices and negotiating codecs in doing this I have noticed something that seems peculiar to me. It seems that including allow=all yields different results than having no disallow or allows in the sip.conf. Could someone please explain why that is true?
Thanks
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2005 Feb 15
1
Queue strategy
Just woundering if the intentend functionality of leastrecent and
fewestcalls it to continually dial only the first chosen ext. in the queue.
In other words if a memeber is logged into the queue but doesn't answer the
call the call never moves on in my configuration from that ext. This could
be really bad!!!!
Thanks
[support]
announce-frequency=45
strategy=leastrecent
music=default
2004 Mar 25
2
Watchguard Firebox 1000 and Asterisk
Has any had any experiences with Watchguard Firebox 1000 and Asterisk. I
have asterisk on public side and phones on the private side. I am able to
get the phones to register and make outbound calls but the inbound calls are
intermittent. I have NAT enable in asterisk and on the Cisco 7960.
Any insight would be appreciated.
Thanks
2004 May 05
3
sip.conf and SIP client host= not recognized in some cases
I am seeing an issue with getting certain sip devices to be recognized as
defined SIP clients host= in the sip.conf and the only deference that I can
find btw sources that work and don't work is that devices that send packets
with an Initial Via header of themselves appears to work and pick the
context correctly but those that don't have the Via just get dropped in the
context of the
2003 Oct 03
4
Iconnect Incomming calls
I have an IconnectHere account with a Inbound number and have setup the sip.conf to register and am recieving the call but When I answer the call it disconnect. I have tried sending the call to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon as I accept the call it disconnects. I believe it may be some type of codec issue but I am not very familiar with that layer.
2003 Oct 17
0
Dial Multiple extension but require input not just off hook to bridge calls
I am try to come up with a way to dial multiple ext and require one or more of the extension to require input before actually bridging the calls.
Example:
A call from PSTN into * a match is made in extensions.conf and * then dials a local(fxs) ext 1 and a Cell Phone
If ext 1 picks up * bridges the call
if Cell picks up require digit to in order to bridge the call
if neither occurs and timeout is
2004 Jan 11
2
Forward call with response required to accept
I am looking for a way to Forward to a external or internal number and require a digit(s) in order to complete forward.
Example:
PSTN1 Calls * dials PSTN2
if PSTN2 presses proper digits bridge the PSTN1 and PSTN2
if no response return to a context
Reasons: 2 actually
1. call is forwarded to cell phone but If cell is out of range, turned off,
2004 Jan 12
0
Fw: Forward call with response required to accept
Sorry, If this is a dual post, was having trouble with email.
I am looking for a way to Forward to a external or internal number and require a digit(s) in order to complete forward.
Example:
PSTN1 Calls * dials PSTN2
if PSTN2 presses proper digits bridge the PSTN1 and PSTN2
if no response return to a context
Reasons: 2 actually
1. call is
2004 Apr 20
0
SIP re-INVITES problem
When a call is place to xxx9931211 from the pstn the call proceeds normally
until asterisk issues the Second INVITE, which is MESSAGE 14, and instead of
call being sent with INVITE sip:xxx9931211@proxy.yyyyy.net SIP/2.0. It gets
sent with INVITE sip:xxx9931211@yyy.33.165.201:5060 SIP/2.0 and this seems
to cause SNOM proxy to return the packet without a Record-Route entry and
then asterisk starts
2004 Apr 26
0
Record-route Issues
Could some please confirm that this behavior is incorrect. I am seeing
issues where it appears that asterisk is not following the Record-route on
it's reply messages. Please let me know if you require any other
information.
Thanks
Example:
xxx.yyy.154.243(PSTN-GW) <--sip--> xxx.yyy.77.23(Asterisk) <--sip-->
xxx.yyy.91.74(SNOM or SER proxy) <--sip---->