similar to: cisco as5400, sip, asterisk. cisco won't detect that the call is answered

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 Messenger! Download today it's FREE!

List users, It's been a while since I've posted here, but I've been hard at work pushing toward our large scale Asterisk goal and keeping up with this list can be a full time job by itself (I have19,543 unread list messages!!). This Friday, September 16th 2005, my team will be at the MCI Development Lab in Richardson, Texas testing our setup. We have a three server system

Hi, this is the second time that i post this, may be a wasnt clear the first time. Im having problems with an incoming peer after 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

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 --------------

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 Cause i = 0x8381 - Unallocated/unassigned number. We

I've the following installation : |asterisk client| --- > |asterisk server| --- > |other asterisk server| all the connections are made in IAX, the client and first server allows 711 and 729 the other server only allows 729 since it has low bandwidth at disposal all the numbers but a few are routed to a digium card in the first server, the others are routed to the other server, this

I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ? There's one configuration working : lynksys pap -sip-> asterisk server -sip-> quescom this way both sides can hear voice but with : lynksys pap connected to a switch -sip->

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

I'm trying to find a way to generate an 'internal extensions' tonelist but I can't seem to find anything on how to do this. My idea was to start a Playtones(intercom) tonelist and not indicate ringing to the line (dead air). But then, somehow StopPlayTones needs to be run once the ringing telephone picks up. This seems like a dirty way to do this. I envision an option to the

CentOS Errata and Bugfix Advisory 2016:0110 Upstream details at : https://rhn.redhat.com/errata/RHBA-2016-0110.html The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) i386: 10407eeb7b75e7d5d7c18fff8a7553be96f96ca65a470ef1589d6c780413fbac paps-0.6.8-13.el6.3.i686.rpm ef013597297b221d5e774a6358fb7770866991a8f53fa93cc3e81fa11eb423fe

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 (0101 to 1111 when I go

CentOS Errata and Bugfix Advisory 2015:1131 Upstream details at : https://rhn.redhat.com/errata/RHBA-2015-1131.html The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) x86_64: 24b3bf897337e5c2f78711db4df73cc5d9265db50d8d84480b418cf381269835 paps-0.6.8-28.el7.1.x86_64.rpm

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-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This is slowly driving me nuts! I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk 1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing calls I get a double ring tone (UK style + US style). I also have a DECT phone on a Sipura SPA-3000 configured with UK tones. This gives me a double ring of UK + UK, so this

Hi, I have the following topology: PSTN/H323 gateway->GNUGK->chan_h323/chan_sip->SIP EP Mostly everything works fine except chan_h323 is not passing audio from PSTN before the call is answered and as a result users can't hear PSTN announcements (like "the number is not in service") that's played on unanswered call. All they hear is just continuous ringtone as though

Hello I recently get a Cisco 7940G IP Phone and I try to make several things with it and I en counted many difficulties: 1.) I tried to unlock the phone and to set manually IP Address, Netmask, Gateway etc. I don't get any luck. 2.) I tried to upgrade firmware like they said with tftp server... I downloaded: P0S3-08-11-00.zip and I uncompressed the files in tftpboot directory. I don't get

I've setup this simple configuration to test the new mediaonly iax feature in 1.4 : Input (client) -> Server (routing) -> Termination transfer=no transfer=mediaonly transfer=no all the machines are in the same 192.168.0.x net the routing Server in the middle has iaxusers realtime backend on mysql the call is originated with a sip phone registered on the Input client

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HI, I am facing problem in connecting ftp from the windows client. CENTOS5U1 Running vsftp daemon. From linux client i am able to upload and download. When i do an ftp upload of any files or folders from windows command line i am getting below error . *vsftp 553 Could not create file* *NOTE: SELINUX IS DISABLED AND THERE IS NO FIREWALL RUNNING* See my user permission and ownership details