search for: as5400

i want to use Cisco AS5400 media Gateway as my PSTN Gateway instead of using the E1 PCI cards in asterisk box ,is this practically possible? can i use SIP in the connection between Asterisk and Cisco AS 5400 Gateway? _________________________________________________________________ Express yourself instantly with MSN Messeng...

Hi, Would just like to inquire if anyone here has a setup of asterisk to send traffic to AS5400 connected to an SS7-PRI.? this is more of a AS54 question, as i've been reading and i always stumble upon PGW2200 as a requirement to handle SS7 signaling on the AS54. Has anyone able to send calls from asterisk to an as 54 with SS7-PRI without PGW2200? TIA Regards, Nhadie ----------...

...re shooting for scalability, so the Asterisk server itself does no transcoding or DSP. We have noloaded all codecs except one and moved any of the resource-intensive activities to the gateway and the support servers. Our production setup will replace the Asterisk TDM-VoIP gateway with a Cisco AS5400HPX Universal Gateway. MCI has an AS5400 waiting for us at the D-Lab, and while they are familiar with most aspects of it, they lack any experience configuring it as a SIP peer for Asterisk. If anyone has experience with this, please share it with me. Copies of your configuration files from t...

I'm trying to get Asterisk to pass thru calls using the G729 codec. I've got a 7960 phone and my gateway is an AS5400. I got the following messages when debugging SIP (7778881000 is the 7960): WARNING[1872]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/7778881000-2874(4) to SIP/as5400-35c1(256) WARNING[1872]: app_dial.c:1002 dial_exec: Had to drop call because I couldn't make SIP...

We've got this configuration : Cisco as5400 --- asterisk main server ---- asterisk for cells ---- gsm gateway cisco and the gsm gateway are connected to asterisk via sip, the two asterisk servers are connected via iax. On a succesful call the cisco (not always, 60% of the times) will keep sending a ringtone to the connected phone, even if...

...i upgraded asterisk from 1.0 to 1.2.4, in 1.0 i used to configure the incoming peers like this: register => @prepago-in [prepago-in] type=friend host=192.168.10.102 ; this is the cisco's ip context = from-external dtmfmode=rfc2833 insecure=very ; required for incoming FWD calls in cisco as5400 the dial-peer is configured like this: dial-peer voice 2662 voip tone ringback alert-no-PI description OUTPUT_TO_ASTERISK translation-profile outgoing remove_# destination-pattern 22662[0,1,8]T voice-class codec 5 session protocol sipv2 session target ipv4:192.168.10.103 <--- this is the...

Hello, I heard that since asterisk 1.6.0.6, now you can send faxes with t.38 through asterisk to a PST gateway that supports t.38 too. Is that true ? If so, what elements you need to make it work beside asterisk and the PSTN trunk ? Thanks all.- -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have the following setup: DS3 -> Cisco AS5400 -> H.323 (ooh323) -> Asterisk Inbound calls work great but outbound calls fail. So to check and make sure we have outbound calling ability on the DS3 we setup a Cisco Call Manager Express and it can make outbound calls both local and long distance with no problems. The failure code is Caus...

I've been searching to a solution to this for a while and can't figure it out, perhaps someone has done something similar. I have a Cisco AS5400 sending SIP traffic via PCMU / ulaw directly to my Asterisk (1.4.19-rc2) box. Jitter and latency are incredibly low on my lightly loaded switched gigabit ethernet network. One Asterisk uses Zaptel and a Digium card, and DTMF recognition works great. However, the same software can't seem...

I've encountered an odd situation with Asterisk 1.4.19 that I can't figure out. If I dial an extension via a Cisco AS5400 with the "g" option to come back, when I then Dial another extension after that, we don't get audio from the caller. There are no firewalls, no routers, no anything but a network switch between. The calls come in as SIP from the Cisco and terminate on a SIP soft client. I sea...

Has anyone got a Premisys Slimline channel bank working with a Cisco AS5400 or similar? I'm not sure if my unit is bad, or what. I'm using FXS Loop Start. Calling the port connects immediately without ringing the attached phone. If I pick up the phone, it's connected and I can talk to the caller. Hanging up has no effect. I can see the bit transitions (0...

Hi All, i have 3 asterisk server in a cluster using a cluster of mysql server via realtime, users can register via DNS SRV. I send/receive calls to an AS5400 via a SIP trunk defined on the realtime sip table, the trunk has call-limit=5. Problem i encountered is each of the 3 asterisk servers will 5 channels each to them instead of 5 for all 3 servers. Is there any solution to this? Regards, Nhadie

...give me SIP that I can point to my Asterisk servers. Currently doing DS3 to Adtran but I want to get away from having PRI cards in all my Asterisk boxes. From looking around I've found some people using: Lucent Max TNT Dialogic IMG 1010 Cisco (Not sure which model would be best for this, the AS5400?) Any real world experience/advice using something like this would be appreciated, thanks. -- Kyle Sexton

Andrew Thompson wrote: > You should be able to set the inbound callerid from the switch/gateway > to a specific unknown in sip.conf file with just a callerid= line. > > The place I looked on the wiki didn't show a specific description for > the callerid= line, but that's what I thought I read for it somewhere. > >

...From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of mmiranda@americatel.com.sv Sent: Friday, January 21, 2005 1:41 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] three way call using sip I connect to the PSTN using cisco as5400 gateways, this cisco devices have E1's to a DMS300 switch. I mean, i configured sip channels (in and oout) to these gateways, i dont have any special hardware in the asterisk server. thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@...

Hi, i cant make a three way call using grandstream phones (BT-100) and asterisk using sip, is this supported or i need a zap interface? thanks

I recently moved my Asterisk configuration to a new server and re-built Asterisk from CVS. Now, I'm experiencing the following issue with SIP: Executing Dial("Zap/1-1", "SIP/100|20") in new stack NOTICE[-1232077904]: File app_dial.c, Line 518 (dial_exec): Unable to create channel of type 'SIP' == Everyone is busy at this time Has anyone seen this issue before?

I am working with implementing Asterisk between four different AS5400's located in multiple sites with different PSTN gateways. I can get two of them to work without a problem, but I am getting the following on the others when I make a SIP call to the other two sites. Got SIP response 500 "Internal Server Error" back from 10.1.3.28 SIP/alma-1b77 is ci...

Hi When I register a SIP or IAX client to asterisk and I dial to it from another UA then there is no problem at all But, when I register two or more clients to the SAME peer (with the same user/pass) and I call to this peer.. Then only the UA which registered the last will ring.. Others don't ring... What can I do about this?? I would like to register for example 10 UA's to the same

Any know what Digium hasn't released the DS3 card? It was supposed to be out a while ago. -Matt