Displaying 20 results from an estimated 4000 matches similar to: "7940 & AS5300 codec issues/questions G.729 & G.711"
2009 Oct 15
2
Asterisk with a Cisco AS5300 gateway
Hi
i test a new equipment on my backbone: a Cisco AS5300 with voice dsp
ressource
connected at a E1 Voice Link.
I want that all call incoming on the cisco 5300 are sent to Asterisk and
all Asterisk outgoing
call are sent to Cisco AS5300.
Actually, i configure the AS5300:
isdn switch-type primary-net5
!
voice service voip
sip
!
voice class codec 400
codec preference 1 g711alaw
codec
2005 Jun 01
0
Segmentation Fautl / Core Dump with G.729
Hello,
Has anyone experienced a segmentation fault in asterisk for using G729
against an AS5300 in SIP ?
I'm having this problem. Also, any other codec I use gives me incompatible
media (can be read in SIP DEBUG messages).
AS5300 configured for multiple codecs, so is Asterisk.
Tried G711u/A G723 and G.729. Any clues ?
Regards,
Jorge A.
Info:
Asterisk ver 1.0.7 stable
Using AMPortal
2007 Jan 24
0
NewTopic - Asterisk and Cisco AS5300 via E1/PRI
Hi,
I had previously posted about connecting an AS5300 to * via SIP/H323. I got it to work via SIP, but
only 1 call at a time would work, and if a user from the * side hung up, the cisco would'nt catch
the hangup.
I an now trying to hook up to the cisco via E1, with a Sangoma A101 card in my * box. I would like
it such that I call from * via E1/PRI to the cisco, and call out via R2 to
2012 Jan 06
0
no audio using g729A for Cisco AS5300 sip peer
Hi,
We need help in enabling g729a codec for our SIP peer that's using Cisco
AS5300.
Our codec is purchased from Digium.
We are able to dial out the numbers and answer the call, but there's no
audio. This is when only g729a is allowed.
We noticed when they also allow ulaw codec on their side, the codec used
falls back to ulaw and the problem is gone.
-------------- next part
2003 Aug 01
0
Cisco AS5300 -- Not hearing anything
Hi to all!
I have this config,
PSTN <--> AS5300 <--> ASTERISK
I am using the Cisco as5300 to receive incoming calls
and routing them to Asterisk for IVR.
When I ran asterisk this is what I get when calling
the voicemail demo.
*CLI> -- Executing Playback("SIP/-081058b8", "transfer|skip") in new
stack
-- Executing Macro("SIP/-081058b8",
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
2004 Jul 15
3
G.729 codec doesn't seem to work *even* after installing the license
Hi,
I am trying to post this again as I am getting no answers and the
support@digium.com bounces...
(I have searched the whole list and can't find the answer either)
I have installed a 5 user license for G.729 and want to route calls through
Asterisk from my G.729 phone to Cisco AS5300 also using G729.
Both Cisco and the phone connect using this codec if I do not force the call
to go
2004 Mar 30
1
G.729 and h323.conf
What should my allow= line look like in h323.conf for G.729?
I've tried
allow=G729A
but this doesn't seem to be right. These "codec indentifiers" sure are
mysterious. Take g711alaw. To allow it you seem to have to use allow=ALAW.
Even though "ALAW" does not show anywhere as an identifier when you say
show codecs.
2003 Jul 30
5
chan_sip.c problems problems from cvs 1.134
All,
I've found problems in my setup with the latest couple of revisions (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, everything is in the same VLAN and only running SIP.
Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300
But inbound calls fail, I see the initial INVITE from the
2003 Jul 28
8
RTP session traversing Asterisk server ...
I've been reading up on the SIP and related (SDP/RTP) RFC's and as I would expect the RTP session should ideally be between the two end points of the call, in my case the AS5300 and the 7940 which are connected on the same VLAN as the Asterisk server.
When I sniff the packets on the VLAN I find that all RTP packets are being relayed by the Asterisk server causing increased load on the
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
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
2004 Jul 29
0
G.729 between Zap and SIP
Hi,
I have licensed the digium G.729A codec. But for some reason incoming and
outgoing calls will ALWAYS use G.711a. When I force my phone to only accept
G.729 then an incoming call from ZAP goes straight to my voicemailbox as the
phone doesn't accept the codec Asterisk wants, even if I force it in
sip.conf.
Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ?
The
2004 Jul 30
0
G.729 <-> ZAP ?
Hi,
I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card.
Incoming calls and outgoing calls between my cisco and my SIP phone works
fine on G.729. Recording messages in the asterisk voice-mailbox also works
fine from both my SIP phone as well as PSTN -> Cisco -> Asterisk. I have
licensed the digium G.729A codec.
When I connect my ISDN PRI to my Zap card and I call
2003 Nov 24
3
Cisco to asterisk termination with h323 and g729 finally works.
Hello,
I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323.
I tested both oh323 from inaccessnetwork and JerJers chan_h323.
I used 1.12.2 version of oh323 and 1.5.2 version of pwlib.
After latest changes from JerJer chan_h323.c works ok when receiving traffic
from ciscos. I havnt found any audio problems although I didnt send much
traffic.
Latest oh323 has some
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
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
2009 Nov 07
1
Asterisk 1.6.1 + Cisco AS5300 + Fax T38 ?
Hi
I have finished the installation of my VoIP basic configuration ...
Actually:
- All calls from my E1 are received by a Cisco AS5300 and sent to my
Asterisk (in G711 by SIP).
- All user are connected by SIP to the Asterisk
- All calls from User are sent by asterisk to the Cisco AS5300
Now, i want see if i can supply T38 Fax Gateway ....
I am search to:
- Cisco Receive all
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