similar to: PIX!!!!!

HI all, I have the following setup running: EP<--->Calling Asterisk<--->Relaying Asterisk<--->Softswitch<---> PSTN The Endpoint EP is registered with the Calling Asterisk. Calls are forwarded from this machine to Relaying Asterisk which in turn forwards it to the Softswitch. In addition, this machine also relays back responses from the Softswitch to the Calling

> -----Original Message----- > From: Christopher [mailto:chris.robinson@voipsupply.com] > I see that there is a lot of discussion on the web about a > common error > after installing and modprobe'ing the zaptel driver. However I don't > see any resolution, anyone found a solution? > > Here's the output I get after modprobe: > > Uhhuh. NMI received.

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

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

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

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

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

Hello list, I have a customer who is interested in standardizing on dell servers for asterisk deployments. Has anyone had success with a particular configuration? Anything specifically to watch out for? Thank you for your time, Art Arthur Miller Sr. Sales Associate VoIP Supply, LLC. 454 Sonwil Drive Buffalo, NY 14225 716-250-3871 OFFICE 716-630-1548 FAX arthur at

I am looking at some lower cost phone to use with Asterisk. What is the ArtDio IPF-2000 or the Sipura SPA-841 like? Also, I see voipsupply.com has an ArtDio IPF-1000 listed, is this a new or an old model? I cannot find any information on it. Adi

Anyone using Nokia "E" Series handsets with Asterisk? I'm trying to deploy some e71's and am having an issue. I can get a single handset working, but when I try to create a SIP profile on the second phone, it won't allow me to save the profile, saying that devices in the same "realm" must have identical username and password. Anyone have a workaround for this

Hello, , I have some of asterisk to CCM4 SIP trunk oneway audio problems. I have setup a asterisk server. It's work great and have no any problems connect to local ITSP (using SIP protocol) . But we need to build a sip trunk to another CCM4 server. The network typology like this. SIP Phone (192.168.1.100) <-----------> 192.168.1.254 (Asterisk Server) 59.124. xx.xx

I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk. It registers fine and I can call between the MP-114 and other extensions, but I'm not having much luck with the FXO ports. syslog shows the problem to be in the MP-114 configuration. Can anyone help?

I used to use IDEFISK, but since it was taken over/renamed into Zoiper it's been really hard work - now I'm told that they won't support my chosen distribution - Debian Etch - the current stable version of Debian I prefer. So, looking to dump Zoiper and go with something else - I want something light-weigh (So that rules out Ekiga - and Zoiper was going down the bloatware route

I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david at safedatausa.com

Are their any good chooses for IAX Adapters? -Thanks

We are currently moving away from a wide-spread Cisco CallManager deployment to Asterisk. For many of our small sites we have the routers configured for what Cisco calls SRST so if we have a WAN failure, the router acts as a SCCP registrar. We are converting to SIP, and from what I can tell Cisco wants a license for each router to run SRST over SIP... So my question to the group is: What are

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

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

I have a customer with a Fortinet Firewall that is having stability issues with Asterisk and SIP endpoints (PAP2T) outside his network. The first issue I see is that Asterisk sees all phones as the IP address of the Fortinet. Since the parameter "localnet" defines the local network and that address falls in that range, how will Asterisk treat the endpoints? I have

I am going to buy some IP phones from them but I sent them an email couple of weeks ago and got no reply. Has anyone ordered anything from them? Any other places that I can buy from? Sorry if it's a wrong post. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050517/ba256d25/attachment.htm