similar to: Cisco PIX and Asterisk

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

Hi all, So I have been reading through the docs available online and the different threads on this list, but I cannot seem to get this phone to work. I have configured the OS79XX.TXT and SIP/SEP*.cnf files (see attached), when I configure the phone to point to my tftp server and reboot it I get this message: Connection received from 10.6.0.224 on port 50608 [09/02 12:16:11.750] Read request

Hi, I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go up to 7.5 However in my first attempt to go from V.5.1 to 6.0 this is hat happens: - The phone reboots - The phone then reads the file OS79XX.TXT from the TFP server. In the file I added the version "P0S3-06-0-00" - It starts upgrading firmware - Then I get the following message: (Upgrade Failed -

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

Anyone have experience with the 3-08-2 release of Cisco's SIP firmware? Are there any new features in the SIPDefault.cnf? Thanks, Ron

Hi All, i dont what changes i made recently but i am unable to hold the call for more 10 secs between MSN and GS. PSTN to MSN and PSTN to GS and vice versa works fine. i am not behind NAT.Also MSN to MSN works fine too. my SIP details [general] port = 5060 bindaddr = 0.0.0.0 context = bogon-calls ;context = default disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm ;My SIP phone - GS

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

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, 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

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

Hi List Just a little bit OT, but then again perhaps an information that could be of great value for a lot of administrators !! Does anyone have experience with how to setup VoIP QoS for outgoing data through a Cisco PIX (515) ? I believe that it should be possible to give higher priority to outgoing VoIP packets. This is due to the problem of ADSL as the UpStream data rate is 1/4 of the

Hi everyone, I'm very new to all this, so please forgive me if I have the terminology mixed-up. We are preparing to install an Asterisk IP PBX over the weekend and I have an issue with the Polycom SP 500 phones we are trying to use. My problem is regarding DHCP. Our DHCP server is our Cisco PIX 501 firewall. I've specified option 66 and the phones connect to the FTP server

Does Kid Pix Studio Deluxe run under Wine? An elementary school I visited recently was running that program on all their Windows machines, and wouldn't consider switching to Linux unless it was supported. winehq.com doesn't resolve in DNS for me tonight :-( so I can't check the database. I don't have a copy myself, so I can't test it personally. (Hmm, maybe I should get

Anyone got a resource on setting up a CentOS box to connect to a Cisco PIX 5xx VPN using IPSec? I don't know if it matters but an important part of functionality would be split tunneling, I need to maintain that from the windows side of functionality. Thanks! jlc

Hi All Total noob on the list so all help appreciated.... I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones.... The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......

Hi everyone, I've been toying with * for quite some time now. I've got two Cisco 7940's with the SIP firmware playing nice with *. I can also make outbound calls via IAXTel (toll-free calls only) and all other calls I have routed out my X100P-clone adapter. Here's my question... Is there a way to capture the inbound callerid from my phone line (coming in on the X100P) and have

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HI All, I am new to asterisk. i want to make outbound calls from asterisk. I tried with many times with the given settings but in vain This is my scenario: I have a *user A* who has registered with sip server(ONDO), I made asterisk to register as a sip client with ONDO, I want to make a call to user A from an extension. My configurations sip.config [general] context=default