similar to: problem with incoming peer (cisco as5400)

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

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!

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

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 the call is answered, actually if the user behind

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

Hi, this was my fault, you are right, i tried with a X-lite Professional and the conference (3-way call) is working now, i guess the phone BT-100 doesnt support it, i dont have a BT102D, so i can tell if it works too, bye -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of mmiranda@americatel.com.sv Sent: Friday,

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'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

Hi, Digging on this case : 2009/5/26 Olivier <oza-4h07 at myamail.com> > Hi, > > In my sip.conf, I've got : > [general](+) > ; register=>tcp://trunk4ipbx:password at 192.168.100.129<trunk4ipbx%3Apassword at 192.168.100.129> > register=>trunk4ipbx:password at 192.168.100.129<trunk4ipbx%3Apassword at 192.168.100.129> > > When

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?

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

Hi, In my sip.conf, I've got : [general](+) ; register=>tcp://trunk4ipbx:password at 192.168.100.129<trunk4ipbx%3Apassword at 192.168.100.129> register=>trunk4ipbx:password at 192.168.100.129<trunk4ipbx%3Apassword at 192.168.100.129> When I'm using the TCP line instead of the other, I've got : [May 26 17:58:42] NOTICE[2859]: chan_sip.c:20169

I finally decided to get a little source code dirt under my fingernails tonight and dig through chan_sip.c to understand how registrations are currently implemented. The hope is to perhaps at least seed some ideas about how to make registrations to a server name, which resolves to multiple IPs, either attempt each IP in the order they're returned by dns, or, simply attempt to register with

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?

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

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

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

Does anyone have any hardware recommendations for a device to take an incoming DS3 circuit and 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

Hi all, I'm trying to convers some Cisco SSCP phones to the SIP formware. The phone boots, I see it tries to fetch a bunch of files on my TFTP: Jun 17 09:37:45 firewall dnsmasq-dhcp[21202]: DHCPACK(eth2) 192.168.10.103 6c:50:4d:da:f0:67 SEP6C504DDAF067 Jun 17 09:38:10 firewall in.tftpd[22666]: RRQ from 192.168.10.103 filename CTLSEP6C504DDAF067.tlv Jun 17 09:38:10 firewall in.tftpd[22666]:

Hi, I got a weird behaviour in asterisk (original found in 1.8 but it is still the same in 11.15.0). I have three phones communicating via OpenSIPs with asterisk. Phone A dials 100 and asterisk calls SIP/phone-b. Phone B accepts the call. The User on Phone B places the call on hold, dials 200 and, while hearing the dial tone of ringing Phone C, places the handset on hook. Phone B sends a REFER,