Displaying 20 results from an estimated 6000 matches similar to: "RE: still no solution for me, if one"
2006 May 15
3
How to tell if RTP stream is has been reinvited?
Howdy,
How can you tell if RTP traffic has been reinvited/is bypassing an * server?
Sincerely,
Brent A. Torrenga
brent.torrenga@torrenga.com
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
+1 219 836 8918 x325 Voice
+1 219 836 1138 Facsimile
www.torrenga.com
2006 May 26
0
Sip Notify cisco-check-cfg - Does it still workwith 8.2?
It does on my test phone. Is your tftp server available?
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Brent
Torrenga
Sent: Monday, April 17, 2006 11:39 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Sip Notify cisco-check-cfg - Does it still
workwith 8.2?
Has anyone else noticed that
2006 Apr 17
0
Sip Notify cisco-check-cfg - Does it still work with 8.2?
Has anyone else noticed that notifying a 79[46]0 with cisco-check-cfg
doesn't elicit any response from the phone using fw 8.2?
Sincerely,
Brent A. Torrenga
brent.torrenga@torrenga.com
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
+1 219 836 8918 x325 Voice
+1 219 836 1138 Facsimile
www.torrenga.com
2006 Apr 18
0
Re: Cisco 7940/7960 SIP 8.2 Freely
It doesn't seem as much broken as just annoying. I am holding off on
upgrading until this resolves, but it doesn't seem to affect performance,
anyways. BTW, some folks say that the server address only gets appended to
the CID when a redirect or something comes about. Our experience here shows
that the IP always gets appended.
>Alexander Burke wrote:
>> Just in case anyone here
2006 Apr 18
0
Voicemail Issue - Failed to lock path
What would cause this? It happened out of the blue:
-- Executing VoiceMail("Zap/3-1", "u326@default") in new stack
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/3' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/6' (language 'en')
-- Playing
2006 Apr 28
0
DNSMasq - Why the stuff hits the fan when the net connection is down
List,
Per someone's suggestion (thanks, whoever you were) from this list, I
implemented dnsmasq to prevent the issue of resolving DNS when the net
connection goes down.
This morning the net connection was down, and our * server didn't miss a
beat. I recommend looking into:
http://thekelleys.org.uk/dnsmasq/doc.html
And
http://www.enterprisenetworkingplanet.com/netos/article.php/3377351
2006 May 23
0
Sip.conf: domain=huh?
So I saw mention of a way to allow dialing using SIP URI's on Dave McNett's
site at http://slacker.com/~nugget/projects/asterisk/page7
Wow, awesome, I can call anywhere now. However, I think there is a more
elegant way of figuring out whether or not the local * server should handle
a given domain. Specifically, Dave compares a series of domains within
extensions.conf to figure out how to
2006 Apr 11
4
Why is the internet connection important to LAN and PSTN calls?
Out internet connection was out this morning. It seems that the SIP
extensions on our LAN were affected. Behavior like:
Call comes in over POTS to a TDM400P, there is a delay then before the Cisco
79[46]0's start to ring.
If we were lucky enough to get a call through, then we could not transfer
the call, or place the call on hold, or park the call.
Outbound calls seemed to have a delay
2007 Jul 05
2
REGEX expression for NXXNXXXXXX?
Hola,
What would a valid regexp in Asterisk be to identify a NANP number, i.e.,
NXXNXXXXXX?
Sincerely,
Brent A. Torrenga
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
tel:+1 219 836 8918 x325
fax:+1 219 836 1138
email:brent.torrenga at torrenga.com
web:www.torrenga.com
2006 May 31
1
Can you dial with different CID's?
Is it possible to dial more than one extension with a different CID to each
extension? I'm thinking macros might be needed, but I don't have a good
handle on macros. Is it possible? Any hints?
BTW - this would be used for showing an internal extension to one phone and
a PSTN accessible number to another phone.
Sincerely,
Brent A. Torrenga
Torrenga Engineering, Inc.
907 Ridge Road
2006 Dec 18
1
Cisco 7940 - NAT Option
I am thinking of turning on the NAT option in our Cisco phones (and the
corresponding sip.conf modification) to allow the phones to be taken outside
the LAN.
Can anyone think of any reason not to just always turn on the NAT enabled
option? I can't think of a reason not to always operate these phones with
this enabled, since it would likely allow them to be taken outside our LAN
and used.
2006 Jun 14
3
Directory - First Name/Last Name - How to use both? a@h?
I think A@Home allows a user to search a directory by either first OR last
name, right? I don't know for sure since I don't run A@Home.
I would like to offer that functionality in my system - and I'd have done it
by now if there was a prompt where Allison asks "press 1 to search by first
name, press 2 to search by last name". But I don't think that prompt exists.
Can
2006 Jun 06
1
SIP One-way audio: == Forcing Marker bit, because SSRC has changed - trxtel.com
Dear list (and more specifically Bret),
I am getting one-way (inbound only) audio when trying to place a SIP call
via voip.trxtel.com (i.e. 18005558355@voip.trxtel.com). The Cli spits out
"== Forcing Marker bit, because SSRC has changed" 5 times after atempting a
native bridge. I realize this is most certainly a NAT issue, the * server is
behind one. Sip.conf has externip=, and
2007 May 14
0
How is Context Determined when Transferring a Call?
When trasferring a call, how is the context determined?
When using a zap device, and the DTMF code for blind or attended transfer is
entered, does the tranfer originate at the context the zap device is set to
be in, or does it originate from where the outside call being transferred
originated in, or the context the current call is in?
I ask because I am seeing strange behavior when trying to
2007 Aug 01
0
Can you specify a sip UA's codec based on IP?
Does anyone have any tricks to use some logic with SIP UA's codec
negotiation based on the UA's IP? What I would like to do is have Cisco
7960's use g711u when they register with a local IP, and g729 when they
register with a non-local IP. I was thinking about sip.conf and making two
entries for each UA, one where the host=dynamic, disallow=all, then
allow=g729; the other
2006 Apr 10
1
RE: still no solution for me, if one provider
>Our user places a call, the gateway responds with no sound at all, or
>hangs up, or gives busy tone.
>
>How can we get to the next provider?
>
>I have now:
>exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-a)
>;exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-b)
>;exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-c)
>exten =>
2006 Jan 30
1
Need to recompile * after changing zap echo method?
Dearest List,
I guess I missed this point: Is it true that if you change the echo canceler
in zconfig.h, and then recompile/install your zap modules, that for this to
be taken into effect by * you must then recompile/install *?
I would have figured that the zap echo cancellation method was independent
of *, and I don't recall seeing any docs mentioning either way.
Sincerely,
Brent A.
2006 Feb 27
2
Echo on PRI/BRI?
Howdy:
Does echo only occur on analogue PSTN lines, or can it also occur on PRI and
BRI lines? If so, for the same reasons? This is a part of our consideration
to transition to BRI.
Sincerely,
Brent A. Torrenga
brent.torrenga@torrenga.com
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
219.836.8918x325 Voice
219.836.1138 Facsimile
www.torrenga.com
2006 Jan 03
2
Looping Problem With Call Forwards - Do you have comments on my solution?
I use IP Kall to forward my missed cell phone calls to. This way, if my
phone is off, or out of a service area, calls will go to my * box.
Concurrently, all incoming calls to my * box cause it to dial my local
extensions at home, my extension at work, and my cell phone via NuFone.
Problem: A loop can be created if my cell phone is not on. Say a call comes
into my * box, it uses NuFone to call my
2006 Jan 27
6
Lockups since upgrade 1.2.3 - anyone else? Any ideas?
Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had
the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24
hours or so.
Since upgrading to 1.2.3, though, the whole system has locked up twice. Once
on Thursday, and then about a half hour ago. The server would reply to a
ping, but no ssh login, no local console login - just locked up. This ain't
good for