Displaying 20 results from an estimated 3000 matches similar to: "G.729A + Cisco AS5300"
2003 Oct 13
2
Extension Dialing problem with SIP
Hi List..
I m getting this mesg while trying to dial an extension, both SIP UAs are registered with asterisk, m trying to dial extension 1015 from UA 12321@xyz.com to extension 1016 of UA 77777@xyz.com
In extensions.conf I added
exten => 1015,1, Dial(SIP/77777,20,tr)
Any hint?
JF
WARNING[16397]: File pbx.c, Line 1153 (pbx_extension_helper): No application ' Dial' for
2005 Feb 20
1
Re: Ring/Off-hook in strange state 6 on channel...
Hello Eric,
call progress detection is the problem. Asterisk mistakenly recognizes the call to be answered and then still "hears" the ringing that should not be there if the line was really up.
To solve the problem you would have to either implement a progress detection matching your country's indication tones or at least adjust the existing one for US or Costa Rica in dsp.c.
By
2005 Mar 18
3
Asterisk handling of SIP info
We encouter a situation where we need to use SIP info to convey infomation
for one end point to another endpoint. I use asterisk to do the test and
find asterisk does not forward the SIP info to another endpoint, but act as
UAS and returns a 4xx error message. I think asterisk is not right to handle
this SIP info message.
In RFC 3261 Page 70 "This protocol is designed to be extended.
2003 Aug 06
1
chan_oh323 + dtmf
Hello all,
I have a cisco AS5300 which is register with a gatekeeper and a Asterisk server also register with the gatekeeper.
PSTN ---->AS5300 ---->Gatekeeper ---->Asterisk
I set up a conference room on the Asterisk sever (Room No 1234).
I try to call from PSTN to AS5300, The AS5300 will call the Asterisk server through the gatekeeper.
I manage to get to the start of the conference
2003 Jan 24
3
OT: don't send html email - RE: Musicmatch
uggggh,
Friends don't let friends send HTML email.
A friendly request that you be considerate to those that do not want email
in virii susceptible formats.
Myles.
<p>Lorenzo banged on his keyboard and his computer puked the following....
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content="MSHTML
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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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.
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
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.
2013 Mar 07
2
Asterisk 1.6 + Cisco AS5300
Hello,
I have a Cisco AS5300 connected to Asterisk (1.6.2.9)
Between 15-16 minutes, the call is disconnected without reason.
Here is what is displayed in the debug:
Received an SDES from 10.4.0.10:17399
-- Got SIP response 420 "Bad Extension" back from 10.4.0.10
-- Stopped music on hold on SIP/as5300-1-0000004d
== Spawn extension (dialin, 065939191, 2) exited non-zero on
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
---------------------------------
Meet your soulmate!
Yahoo! Asia presents Meetic - where millions of singles gather
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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
2005 Mar 29
3
help w/ basics
Hello, I am new to Asterisk and new to this list. I got Asterisk setup and
running using Asterisk@home, and purchased a PolyCom SoundPoint IP500 phone
to test out.
I cannot get the phone to talk to the Asterisk box. On bootup of the phone,
it tells me that it cannot contact boot server. Why is that? It gets an IP
fine, and I have also tried manually setting the IP of the phone and the
Asterisk
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
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 Aug 21
3
Conference + time limit
Hello
Conference again. Meetme can now limit number of users in a room. Can it also limit how long a conference session? Someone ask the same question (from achive) but doesn't have a solid answer.
Foong
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2005 Feb 24
2
asterisk supports VXML?
Hello,
Does asterisk supports VXML?
Couldn't find much resource on that on google and wiki.
Thanks
Foong
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
2000 Mar 08
1
infinite recursion problem
hello r-users,
sorry for asking a long question that may not be very relevant for the
list but it's upsetting me and I get no other solution...
I get a function HTMLExport.lm that uses another function called
HTMLExport.list.
My problem is that function HTMLExport.list works fine when used alone
but HTMLExport.lm crashes with the following error :
> HTMLExport(iris.lm)
lm(formula =