similar to: need help asterisk and AS5300

I do part time consulting work. I need to setup an asterisk system to allow me to record both inbound and outbound calls to clients. I have one client that is just a PITA. The client has changed their mind three times so far and we are at step one. I have a spare slackware box and a seperate phone line for the consulting work. I have MCI Neighorhood as my carrier. What I need to know is: 1.

Hi, I am currently replacing my old Hylafax server to a new hardware with CentOS 4.2 (x86_64). Since CentOS 4.2 (and probably RHEL4) is shipping with libtiff-3.6.1-8, I'm wondering if this version of libtiff has the patch mentioned in the hylafax bugzilla http://bugs.hylafax.org/bugzilla/show_bug.cgi?id=500 and/or is it compatible with the latest Hylafax 4.2.5 (i read libtiff-3.6.1 is

Hello, i've recently switched over from 1.0.9 to 1.2.3. I've experienced some (to me) weird behaviour. This is the config for an example queue.conf: [654] wrapuptime=30 timeout=20 strategy=ringall retry=5 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=queue-thankyou queue-callswaiting=queue-callswaiting music=default monitor-join=yes monitor-format=

Hi: I have a Windows 2003 Server Enterprise Ed. as Domain controller, an its current domain functional level is 'Windows Server 2003'. Also, I have a RedHat Linux 7.3 server with SaMBa (tested with rpm samba-3.0.0-2, and compiling the samba source code). I'd joined the linux server to the AD tree without problems, access from it to the Win2003 shared resources too, but I have

hi All I need your help , for used Digium Card TE405P, for setting this Board AS E1 ISDN PRI. 1 .Current for make sure my config its rights or no I inform my configurations in Board Jumper T1/E1 is Closed is that rights or no ? for E1 i closed the Jumper. 2. I Want To seeting E1 in ASterisk/PC Back To Back To Cisco E1 AS5300 Use ISDN Signaling, my configutration :

Hi, After following the suggestions on the mailing lists and the wiki I'm still experiencing choppy moh. The song plays but with frequent noise parts. - I'm using asterisk 1.2.4 on our production server and 1.2.7 on the test server. - native moh with .gsm and .pcm formats (according to http://astrecipes.net/?n=152) - compiled ztdummy as a timing source any pointers on how to dig deeper

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

I've previously been using G711alaw on both the AS5300 and the phones but feel the need for a less bandwidth hungry codec for those users that are connected behind ADSL and so was investigating G.729 but .. Firstly I found that on my AS5300 I have either G.729r8 or G.729br8 and on the 7940 phones I have G.729a, I'm not sure which interoperate the best with each other and so was wondering

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

I've got asterisk working great with cisco 7905 sip phones. I've just got one issue that I can't figure out. I have a 5300 connected to the PSTN via PRI. When I send a call from the 5300 to asterisk it will ring the 7905 phone for 4 seconds then drop. This is because the only message asterisk sends to the cisco is the 100 trying message. The 5300 receives that and sends a call

I am trying to send calls from an AS5300 to Asterisk via e1 and I get this bit of information in place of routing information Going to extension s|1 because of Complete received Accepting call from '' to 's' on channel 1, span 1 Here are the relevant zaptel and zapata pieces. span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 signalling=pri_cpe switchtype=national

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

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

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:

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 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

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

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 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