similar to: Asterisk -> AS5300 SIP Interoperability

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

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.

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

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

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. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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

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 -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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

Hello, I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in turn is talking to an Asterisk server via SIP for call origination and termination. Seems simple enough, and it works for the most part, but: 1) Caller ID name data comes in on the PRI, but doesn't appear to get handed off to the Asterisk server via SIP, at least not in any format that Asterisk

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

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",

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

Hello all, I have data: Genes time rep vart dye y trt 130911 sa1-d07 030min 1 col g 9.636244 o 145771 sa1-d07 030min 1 col r 8.107577 c 93335 sa1-d07 030min 1 ler g 7.409566 o 94821 sa1-d07 030min 1 ler r 5.107160 c 10119101 sa1-d07 030min 2 col g 8.336862 o 11605101 sa1-d07 030min 2 col r 7.824530 c 725313 sa1-d07

I don't think it will be easy to identify the problem remotely. You can try logging in with password (if not disabled), or sshing with some other key, or logging into some other user. If you are able to get access to the machine, post the server's error log here. On Mon, Oct 24, 2016 at 9:55 PM, Jin Li <lijin.abc at gmail.com> wrote: > Hi Tanmay, > > The server is not

Hi All, I use Ubuntu server 10.04 LTS as virtualization platform. Actually running kernel 2.6.32-31-server #61-Ubuntu S root at jupiter:~# uname -a Linux jupiter 2.6.32-31-server #61-Ubuntu SMP Fri Apr 8 19:44:42 UTC 2011 x86_64 GNU/Linux We have on running virtual root at jupiter:~# virsh list --all Id Name State ---------------------------------- 1 kvmtik.4safety.cz

Hello folks, I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no dialtone & can't get it to ring. My mgcp.conf says: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [172.16.2.25] host = 172.16.2.25 context = default line => aaln/1 And here's the interesting bits of extensions.conf: [globals] ... TRUNK=H323/BYEXTENSION@pstn_gw ...

OK, here goes my next problem. I have puchased a DID which I can connect to via SIP I have been given the following details: Username: uka1xxxxxx Password: 1000xxxxxx Server: brxxxx.net:5160 My equipment is Asterisk CVS HEAD on Red Hat EL 3.0 (NO NAT) The other end is a Cisco AS5300 (NO NAT) I can register with the Cisco with no problem. When I dial the DID it sends the call to my asterisk

Thanks for both the mapply and array approaches! However, although intended to generate the same result, they don't: # mapply approach n = 3 sa <- rnorm(n,0.8,0.1) so <- rnorm(n,0.5,0.1) m <- rnorm(n,1.2,0.1) mats = mapply(function(sa1, so1, m1) matrix(c(0,sa1*m1,so1,sa1),2,2,byrow=T), sa, so, m, SIMPLIFY = FALSE) print(mats) [[1]] ????????? [,1]????? [,2] [1,] 0.0000000