similar to: Asterisk to CCM4 SIP Trunk one-way audio problem.

Hello We have integrated cisco callmanager 4.1 with asterisk and we can dial from cisco to asterisk but we're getting an error if we call from asterisk to callmanager. This is the error I'm getting anybody can help me? Verbosity is at least 3 -- Executing Dial("SIP/2234-e084", "SIP/cme-pbx/4455") in new stack -- Called cme-pbx/4455 -- SIP/cme-pbx-25ae is

Hello list. No intention to start a flamewar here but I would really like opinions from those who know both the Cisco and Asterisk system. I'm working for a company with 15 offices in 11 countries, offices are relatively small (3-20 people each) and most of them have a Cisco 1760 Router installed with Call manager express (CME) and 1-3 ISDN lines (2-6 simultaneous calls). We

I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) --> Internet --> PIX (NAT) --> Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the

Can anyone point me in a good direction for configuring SIP through a PIX using 1:1 NAT. I have read anything I could get my hands on and tried them all with very little success. I can get it to work through the cheap little cable modem routers, but not this PIX. I -can- make a direct SIP call using the IP address of the * server (ie.exten@ipaddr), but when I do that * still doesn't

Consider the following directory structure /foo/aaa/*/* /foo/bbb/*/* /foo/ccc/*/* I want to sync all of /foo, but exclude /foo/aaa but not exclude any other occurances of "aaa" or "foo/aaa" (be they files or dirs) that might occur within the other dirs /foo/bbb/* /foo/ccc/* etc I don't want to exclude /foo/bbb/aaa or /foo/ccc/111/aaa or /foo/ccc/111/foo/aaa etc...

Hi, I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall. After some period of time, asterisk says that some of them are unreachable, and the phones lose their registration. The only way to make the phones recover is to clear the NAT translation tables for the phones on the PIX (clear xlate...) Does anyone know how to fix this? As you can imagine, it is quite annoying. And it does not

does anyone have a sip working through a cisco pix firewall? i can get the phone to register and the call to be negotiated, but as soon as the call is answered there is no sound and the call ends immediately. im sure this is due to the RTP negotiation being rejected by the pix. any helpful ways around this? right now my only solution is to put a small box outside the firewall and IAX the

Hi, I asked this last week but i didn't get any answer So i will elaborate on my question. I need to setup a pix 515 firewall (running 7.2.2 OS) to allow sip traffic thru it from a sip phone wherever i may be. The pix is where all my servers are colocated and i will need to connect thru it from softphones / hardphones wherever i happen to be traveling. I need help setting up the pix for

Is there anything special that anyone here has had to do to get an Aastra phone (on the Internet) to talk to Asterisk behind a PIX firewall? Ports 10000-20000 UDP are open on the PIX and forwarding to the Asterisk server. The Asterisk server's RTP.CONF is set to use 10000-20000. The phone registers, and will place AND receive calls, however, no audio is passed. The phone is an Aastra

I have Asterisk server providing phone service for my company. The server is behind a PIX-515 FW and is assigned a private address 192.168.11.X/24. With that said what is best to provide remote SIP phones (home offices) securely. If the solution is to put up another Asterisk server with a public IP address I am opposed to that. I am looking for the a secure reliable solution to set up remote SIP

This is kind of an odd request so I''ve got the asbestos undies on. I have a client who currently has a layer 3 switch plugged into a cisco pix. Routing is handled via RIP and now the client wants to insert a linux box running shorewall behind the PIX and in front of the switch to act as a content filter+backup firewall. The immediate problem I forsee happening is that RIP broadcasts to

I''m trying to get picture from ethloop and gnuplot. But the output seems weird. I still do not know why and want some examples. Thanks in advance. --------------------------------- Do You Yahoo!? 注册世界一流品质的雅虎免费电邮 --0-1760319804-1107141989=:62961 Content-Type: text/html; charset=gb2312 Content-Transfer-Encoding: 8bit <DIV>I''m trying to get picture from ethloop and

I normally use ssh to log into a remote server, change directory and then use scp from there to copy files from another remote server to the first one. Now the first server has been hit by continuous error correction messages from the ECC controller, all of which are corrected, and I am unable to get a command line to issue the required commands to change directory and then run scp from the other

Hi, We have a shorewall firewall running on SUSE 10. We have three nic''s, Lan, DMZ and Internet. We also have a Cisco Pix 506e. We moved from sending all our traffic through the pix to using the Suse box yesterday. The PIX is in the DMZ, with a connection to the LAN switch, the idea being that VPN users can connect to the pix to the lan. The Pix is on the 10.0.1.x subnet, the lan is

Hi, I have the Cisco PIX 506 firewall right in front of the asterisk and I am getting a one-way audio. I need your help/guidance to resolve this problem. I have the "fixups" disabled for SIP in the Cisco PIX 506. Any help rendered by you in this subject is greatly appreciated. I have been breaking my head trying to resolve this problem for more than one month. I have included the

Hello, I am not too familiar with the settings in our PIX (learning though). Here is the only access-list setting that we have in place for Asterisk: access-list acl-prod permit udp any host EXTERNAL_*_IP_HERE eq 5060 In rtp.conf we are allowing ports 10000 - 20000. We are not using SIP Fixup in our PIX due to firmware version. How do I go about adding the ability for udp ports 10000 - 20000

hey friends, I am looking for a free open source software (web based or application) through which I can monitor the Pix Firewall. What it should show Interface status or traffic , VPN Connectivity status, CPU Status, Memory Status etc. I am also running DHCP server on Pix Firewall (due to some reasons) If it can monitor that also means showing how many IPAddresses has been assigned, to whom,

Hi everyone, This is my first post, so apologies if I have not included all details about the issue. I am using a Nokia e71 to connect to a corporate asterisk server and am having issue with dialing. I can dial all extensions and receive all types of incoming calls. I cannot however, dial local phone numbers. When putting the service into debug, it appears that the device does not enter

Hello, long time lurker, first time writer.... We have the following set up ITSP | | Internet | | Cisco 2600 | | Switch----Asterisk Server running 1.0.9(has public ip) | | Cisco 515e Pix Firewall running Pix OS 5.3(run's a class c 1-to-1 nat and pat) | | Grandstream GXP-2000(run latest fw from grandstream site 1.0.1.9) The grandstream registers with the public asterisk server fine. I even

Here is a newbie question. Hope, you will not mind. I am trying to backup a file from my NT machine onto AIX machine. Here is the setup: NT machine (testor): rsync is in c:\Rsync\rsync.exe and is running as a daemon. (I created a service for Rsync first). AIX machine (admx): rsync -v rsync version 2.3.1 Copyright Andrew Tridgell and Paul Mackerras admx::nkapoor> telnet testor 873 Trying...