Displaying 20 results from an estimated 1000 matches similar to: "Asterisk 1.6 + Cisco AS5300"
2013 Jun 12
2
Asterisk sends the INTERNAL IP address of my equipments to my SIP friends?!?
Good morning, or Good afternoon! It depends :-)
I have a standard Asterisk configuration:
SIP friends (phones) <-----> Asterisk <-----> SIP gateway to
PSTN converter
80.236.215.61 109.69.217.6 internal IP (
10.4.0.10/255.255.255.0)
When analyzing traffic on a SIP friend/phone I see this:
INVITE sip:xxxx at 80.236.215.61:64946;ob
2011 Jan 10
3
sendrpid does not work!
Hello,
I have Asterisk 1.6.2.9-2, the directive "sendrpid" does not work!
I placed this in my peer: (sip.conf)
sendrpid=yes
trustrpid=yes
or
sendrpid=yes
trustrpid=no
(and restarted Asterisk)
and the line "Remote-Party-ID" does not appear in my sip debug!
Please help me,
Mickael.
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2010 Apr 07
2
AGI + Dial + stream file ?
Hi all,
I am running an AGI script in a command dial, or call a SIP trunk.
I want to execute after 10 minutes a voice message (stream file) on the
channel to warn the person that the call is about to end. How to do that?
Thank you,
Mickael.
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2014 Mar 26
2
Default extension
Hello,
When I get a SIP INVITE as follows:
INVITE sip:s at 10.1.0.191:5060 SIP/2.0
Max-Forwards: 69
From: "0475XXXXXX" <sip:1053212 at sip.domain.com>;tag=as7df9ab18
To: <sip:02XXXXXX at IP:5060>
Contact: <sip:1053212 at IP:5060>
Call-ID: 344d42bd16975a54141d11f635bdfc71 at sip.domain.com
CSeq: 102 INVITE
Date: Wed, 26 Mar 2014 15:06:01 GMT
Allow: INVITE, ACK, CANCEL,
2009 Oct 28
2
Asterisk/Cisco AS5300 => Two problems in incoming (extension not found)
Hi
Now, my Cisco AS5300 sent call to my asterisk, but two problems:
When i call the phone number, i have:
[Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite:
Call from '' to extension '0426000000' rejected because extension not found.
[Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite:
Call from '' to extension
2010 Jul 16
4
chan_local - Asterisk 1.6.2.6
Hello
I just coding a AGI script for billing.
- For external calls, I pass the call directly on a trunk. I do :
Dial(trunk1/extension) -> OK !
- For internal calls (shortcode, others users ...) I am
Dial(Local/extension at context/n)
The problem is that through chan_local.so, I sound as it cut!
Example if I call the voicemail ... "You have No messa ..." or "You have
2004 Jun 08
4
AS5300 and Asterisk
Hey all,
I have an as5300 I use for dial in customers, we have 4 PRIs on it.
We have a few free channels on it. I'm wondering if I setup SIP on the
as5300 I can have asterisk use the free channels for dial out.
I'd still have to use my TDM04B for incoming calls, but at least I can
expand my outgoing.
Anyone done anything like this before? I've never messed with VoIP on
Cisco
2004 Dec 01
3
Asterisk + AS5300
Is it possible to terminate calls via SIP on a Cisco AS5300? Did anyone do it? How? Do i need an special IOS version?
Ive been trying to compile the OpenH323 channel for the last month, but errors still happens.
Thanks in advance.
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2010 Jun 23
1
I look ARI (Asterisk Recording Interface)
Hello,
I look ARI (Asterisk Recording Interface)
the publisher site is closed...
http://www.littlejohnconsulting.com/ari
Thank you,
Mickael
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2003 Sep 22
2
G.729A + Cisco AS5300
Hello,
I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected.
The codec list show on my cisco AS5300 for g.729 are:
g729r8
g729br8
I suspect that
2006 Jan 24
1
need help asterisk and AS5300
hi All
Any body already setup asteriks call routing to Cisco AS5300 with SIP Server ?
i need informations sample config for that, or can show how to route docs .
thanks
Dirgan
---------------------------------
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Yahoo! Asia presents Meetic - where millions of singles gather
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2003 Jul 17
3
Asterisk -> AS5300 SIP Interoperability
Greetings,
I am attempting to configure an AS5300 to provide a SIP based gateway to the PSTN from Asterisk. I have been unable to identify through the docs how specifically this should be configured in Asterisk and have not been able to get things working through trial and error.
I am sure I am missing something fairly obvious here but any guidance (or example cfgs) would be much appreciated.
2007 Jan 04
2
Cisco AS5300
Hi all,
I realize this is OT.
I just got a Cisco AS5300, and I need to configure it like such:
Asterisk -----(H323/SIP)------> Cisco ----- (E1/PRI)------->Telco
So calls originate from the Asterisk side (registered users on SIP or just ZAP phones), and they go
out H323 or SIP to Cisco, where they go out PRI.
I have the Asterisk side sorted :) (either H323 or SIP), I need help in the
2018 Nov 03
2
limit-rate
Hi,
Where is the mount option 'limit-rate' in the current version?
I checked in cfgfile.c and in the documentation, no mention.
Yet this option did exist at one time:
http://lists.xiph.org/pipermail/icecast/2010-October/011703.html
http://lists.xiph.org/pipermail/icecast/2009-January/011391.html
I try to limit the bitrate of a mount-point, is there another solution?
Do you know why this
2005 May 11
2
Asterisk and Cisco AS5300 or 3600
Guys.
I need some advice on some h323 issues. I need to test connectivity from
Asterisk to a Cisco AS5300 that has PSTN lines and to cisco 3600 voip
routers.
H323 needs to be used here but I was wondering if anybody has linked
Asterisk to these Cisco routers before?
Thank you for any pointers.
2018 Nov 03
2
limit-rate
Hello,
Thank you for your response.
It is on the kh version..
https://github.com/karlheyes/icecast-kh
Le sam. 3 nov. 2018 à 21:47, Thomas B. Rücker <thomas at ruecker.fi> a écrit :
> Hi,
>
> On 11/03/2018 07:33 PM, Mickael MONSIEUR wrote:
> > Hi,
> > Where is the mount option 'limit-rate' in the current version?
> > I checked in cfgfile.c and in the
2004 Apr 05
1
Extensions.conf sending calls to Cisco AS5300
I have my server configured to send to send all PSTN traffic to my Cisco
AS5300 gateway via SIP. I use the following line in the extensions.conf file
to accomplish this:
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@10.1.1.1,240,T)
Unfortunately, when I removed the T from the end of the statement, the calls
still complete, but they drop as soon as the called party answers the phone.
I thought
2005 Jun 03
1
oh-323 / Cisco AS5300 problem
Hi i'm trying to connect to the PSTN in the following way
sip ATA -> * -> gnugk -> Cisco AS5300 -> PSTN
I'm using asterisk CVS-HEAD-06/01/05-14:33:15 running over RH EL3
Asterisk-Oh323 0.7.2 pre1
Open H323 v1.13.5
pwlib v1.6.6
and I'm having a lot of trouble, gnugk and * both have public ips and are not behind any type of firewall, the sip ATA is behind a firewall and
2003 Sep 29
2
cisco AS5300 : problem configuration
I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ...
I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly
2010 Jun 11
1
MeetMe
What is the interest to supply binary of Asterisk, under debian for example,
while to use MeetMe it is necessary to COMPILE Asterisk ??? :-))
Mickael.
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