similar to: MGCP : chan_mgcp.c:1509 find_subchannel

Where does the date/time stamp from Caller ID come from? On my extensions ATA188 and IAX2 soft phone the caller id date / time is 12/30 12:00AM. The Linux time is correct. SayUnixTime return the correct time. Any Ideas? Does this work? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. Thanks dave -- Dave Weis "I believe there are more instances of the abridgment djweis@sjdjweis.com of the freedom of the

Hi, Is there is some form of VAD on * for SIP channels, cause I have a problem with MOH. I made an extension which simply plays MOH, when I dial that extension with my ATA188 MOH sounds choppy if I talk on the phone the MOH keeps playing. I saw the sip channel (show channel SIP/*) and I see no packets going in/out when I talk then packets shows going in/out. I don?t have this kind of problem

I have a DG-104S (which I reset to factory settings, it's DHCP'ing off my network, plugged into the WAN port). The system comes up, and I through the web browser set under Call Agent IP Address to: Notify Entry: dlinkgw@[192.168.1.1]:2427 (192.168.1.1 is the * server) I have RGW Name: and DNS IP addres the DNS IP of the MGCP box and DNS State disabled (not sure what to set it to) --

Hi, This is a message I already posted on the chan_sccp mailing list, but since this list has a lot of active members, I'm hoping someone might be able to help (And my problem is * related, so I guess it's ok if I post it here also ;) ). I'm trying to get SCCP ATA188s to run with Asterisk. The Asterisk box uses the latest Asterisk@Home image (Version 2.6). I have compiled and

Does anybody has the expirience configuring Asterisk with Cisco ATA 186 MGCP firmware ? I have Cisco software v3.1.1 atamgcp (Build 040629A) Asterisk 1.0-RC1 On ATA i only put domain test. mgcp.conf looks like this [test] host = 192.168.195.55 context = default line => aaln/2 line => aaln/1 Asterisk CLI shows this: Jul 31 16:05:40 WARNING[135449600]: chan_mgcp.c:485

Hi, there is a bug in chan_mgcp.c which shows up if you have more than one MGCP gateway configured with host=dynamic. The problem is in the routine find_subchannel when a MGCP response is received. When the response is handled find_subchannel is called with name = NULL and sin = address. This cause the find_subchannel routine to alter the address of all gateways up to the one the response is

I'm not sure that the problem lies in the NAT because the phone is talking to Asterisk. I'm hoping this is a simple config thing I've overlooked but I've tried all kinds of combos inside the [] in my mgcp.cfg file. The phone's IP is 192.168.1.116 (my comp is .110). The router to which the phone and my comp is plugged into has a WAN IP of 10.0.0.28. All the other comps (and SIP

Hello: The hostname is 192.168.1.27 The server_name is: server_name www.lpbn.org Anyone know of how to get rid of the space? <p>When people click on my links they get a space between the url and the port. The link on the page is: http://www.lpbn.org:8000/playlist.pls?mount=/file/Fri-16-20.mp3&file=dummy.pls The returned link to the client's player is http://www.lpbn.org

CentOS 6.4, clean install. Zimbra 8.0.3 I am behind a PfSense box using a virtual IP. So the IP of the box is 192.168.1.27 I entered this in /etc/hosts: 127.0.0.1 localhost localhost.localdomain localhost4 localhost4.localdomain4 ::1 localhost localhost.localdomain localhost6 localhost6.localdomain6 192.168.1.27 mail mail.meowbox.me meowbox.me but `hostname -f` says: $ hostname -f

Hi All, I have 2 NICS in this system. CentOS 6.4 eth0 is the virtual IP from PFSense mapping connected to the router, works fine. eth1 is a second NIC that I have assigned a private IP to and connected it to a switch on the private network. I have many other private devices, so I know this setup works. When I connect the cable to the switch and bring up eth1 the system basically stops taking

I manage a bunch of workstations at the Wendell Free Library. They are all diskless, boot via PXE and mount all of their file systems via NFS from a server. All of the machines are 32-bit and run CentOS (fully up-to-date running 5.10). There are two printers with queues managed on the server. The server 'shares' these printers on the local LAN (eg with all of the workstations).

I am trying to dial a mgcp extention from my sip phone and i am getting this error message. anyone got any idea? error I> -- Executing Dial("SIP/2204-5dc2", "MGCP/aaln/1@10.0.1.150") in new stack May 19 22:30:01 NOTICE[1251156800]: chan_mgcp.c:1104 find_subchannel: Gateway '10.0.1.150' (and thus its endpoint 'aaln/1') does not exist May 19 22:30:01

Hi. I'm new in Asterisk with MGCP. I set up a MGCP user agent and start asterisk with the next configuration files. '--------------- extensions.conf ---------------------------------------------------- [general] static=yes writeprotect=yes [globals] ap1 => mgcp/aaln/ap200@64.76.148.186 [macro-apl1] exten => s,1,Dial(${ARG1},30,Ttmr) ;exten => s,2,Voicemail(u${MACRO_EXTEN})

Okay, so I'm a noob. Asterisk looks very promising, so I say "thanks" and "good job" to all who contribute. My * test box is up and running with soft phones using IAX and SIP, so now I'm on to testing hard phones. I borrowed a couple ShoreTel 210 phones from somebody who had them on hand but they only support MGCP. I see that there's an mgcp.conf in

Hi all, I have VOIP network built with MGCP endpoints.The manufacturer of endpoints is ASKEY. I downloaded latest Asterisk software and found it very useful for me. I configured it and it seems taht everything works OK when I am testing it with one or two endpoints. After that I tried to move Asterisk to working network and replace existing call manager. It starts working and calls are

Hi folks I seted up the asterisk with an active ISDN B1 AVM Card (german vendor) and it works fine, various SIP clients (IP fon snom, xlite under MacOSX) and also incoming and outgoing connectins. Ok. No problem. After that I configured a CP7940G with a MGCP IOS. It's connected to the asterisk too via switch. No NAT deivce between. And of course, I can call out to PSTN and also to one of the

On Friday, 08 February 2002 at 13:12, Samuel Hathaway wrote: > On Fri, 8 Feb 2002, Akos Maroy wrote: > > > Jack Moffitt wrote: > > > > > Why are you writing clients that don't use libshout? > > > > darkice doesn't use libshout either. > > > > I guess the main reason to have a protocol is that anyone can connect > > using it? >

I repost a message I put a week ago: I have a Cisco IAD 2431 which has MGCP protocol; I cannot make it to work againts Asterisk; at least there is some MGCP conversation between them but when I offhook a phone attached to IAD I get no tone at all. As anybody managed to get working Asterisk against an MGCP Cisco gateway ? Which MGCP version should I use ? Also I recently

Hello. Trying to get shorewall to do what I want it to do and also not to do what I don''t, I have the following example which I''d like to ask about. I get this log message: Nov 9 11:06:36 fw kernel: Shorewall:all2all:REJECT:IN=eth1 OUT= MAC=00:c0:4f:60:b3:e5:08:00:20:b0:92:c1:08:00 SRC=192.168.1.59 DST=192.168.1.1 LEN=84 TOS=0x00 PREC=0x00 TTL=64 ID=0 DF PROTO=UDP SPT=942