similar to: Sound through NAT issue

Hello, all. I'm afraid I've been dropped into the deep end even though I am an Asterisk novice. I've set up a few tiny, tiny systems in the past and have now been asked to pull together Asterisk, FreePBX, Kamailio, RTPProxy, and Fedora Directory Server into a VoIP service. After googling and reading for most of the last 24 hours, I finally have my head around the components and how

Hello, all. I see there is an "o" option for the Dial() command which reverts to the previous behavior of using the original callerid throughout the call - I suppose more specifically, using the callerid from leg 1 for leg 2 in B2BUA if I understand it correctly. That seems to be highly desirable behavior; I know we are seeing some problems with call history and call forwarding because

Hello, all. After reading the README, UPGRADE.txt, and a quick tour through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2, one simply compiles and installs over the old installation being careful to NOT install the sample files? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsullivan at opensourcedevel.com

Hello, all. After many pages of googling and testing in the lab, I'm still a bit perplexed about how to implement tls protection for the asterisk manager. manager.conf allows one to specify the cert file but one normally must also specify the private key file. If I simply enter the cert file: sslenable=yes sslbindport=5038 sslbindaddr=172.x.x.8 sslcert=/etc/pki/tls/certs/pbxc.pem ; path

Hello, all. In our multi-tenant environment, we would like to be able to use the reinvite media redirection within Asterisk for calls within a tenant but not between tenants. We would like inter-tenant calls to be fully proxied by the Asterisk server. I think the answer is, "we can't," but I thought I'd ask anyway. I'd dearly like to remove the substantial traffic

Hello, all. My apologies up front but I must be brain cramping on something very simple. I've tried to pare down my configuration to the absolute minimum for SIP traffic just to understand how it works. My incoming calls are not finding the "s" extension in my dial-plan. I am assuming SIP calls can do this. I am using Asterisk 1.6.1.1 sip.conf has nothing but: [general]

Hello, all. Since implementing an iptables firewall between the Asterisk PBX and several SIP phones, the Asterisk PBX ability to "reinvite" has been broken even when the phones are on the same network (i.e., no firewall between the phones). We've been beating our heads against the wall thinking it was the complex rule set but it appears the issue is ip_conntrack_sip. Before I drop

I recently did a set up where I replaced a simple D-link home router that was having trouble processing a T1's worth of bandwidth with a linux machine running iptables. the kernel was 2.6.29-r5 and I chose the SIP connection tracking modules from the menuconfig. Router worked fine for normal traffic, but I was unable to get the SIP phones to work. Using ngrep it was plain to see

Any vitelity customers with pbxinaflash boxes? I'm able to call in-house, but failing to make outbound calls. My assigned server at vitelity is not reachable. I can ping to my ISP OK. Any help appreciated. Such as actually how to make email contact with support at vitelity. They're not responding. Thanks, Tom

Hi all, I'm having trouble with voicemail on my *NOW 1.5/FreePBX box. I have enabled voicemail in the extensions area, and set the default password. However, every time I try to log in with a mailbox and password, I get the message "login incorrect". I've tried changing the voicemail password, and also disabling and re-enabling the voicemail feature. What else can I do to set up

Hi, Did anyone have any experience with CyberData SIP-enabled VoIP Intercom units please? Are they any good? Can you recommend anything better? Thanks, Finku -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090720/b5d2d785/attachment.html

I want to set sflphone as extension on asterisk. I have a sip account/DID with vitelity.net. Not sure what to put in the wizard: alias ??? hostname ??? is this the asterisk server hostname, or the hostname where my sflphone is sitting on the lan (it's a home network) username ??? is this the assigned extension number? password ??? is this the assigned extension number password? Any

hello is it possible to lock a conference IF no admin are connected ? or how to do to have a conference offline? thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I put a post on here about my issues with outbound calls not ringing but i haven't resolved it. so i am trying again. When i dial any outside number i dont get a ring tone at all. when the person picks up and starts to talk i can hear them fine. it sounds great. How do I start to troubleshot this? _________________________________________________________________ With Windows Live, you can

Hi All, I have a senario where we have multiple locations and all have the ability to call using 1NXXXXXXXXX pattern, so we have created multiple contexts so the outbound goes fine, but while transfer occurs (after picking the inbound call and transfer), it uses the first 1Nxxxxxxxxx priority patterned context, like if the 3rd location is making a transfer, but 1st location have the priority

I have just established a call between 2 sip phones and I have noticed that all RTP traffic goes through Asterisk Server. I was expecting RTP traffic went to one phone to another phone directly. I set canreinvite=yes in sip.conf in both sip peers. I also tested it with 2 mgcp phones and same result, all rtp traffic goes through Asterisk. Is there any way to force traffic to go from one phone

Is there a way to detect if a call is a transfer in the dialplan? Here is my issue: I have an office with 2 extensions. Under normal circumstances any call that comes in should ring both extensions. I accomplish this through a queue. The problem is that if the call is answered on say extension 11 and the answerer wants to transfer the call to the other phone, extension 10, transferring

Hi, How can I match an extension "ending with 3" (just an example but applicable to any other digit, including * or #)? exten => _ZX.3,n,... exten => _ZX.#,n,... (the above does not work) Can regular expressions be used in the standard dialplan (end with: "$")? Thanks, Vieri

Hi Someone may give me an example of followme() application using in a dialplan (including what to configure in followme.conf) ? I use asterisk 1.6.1 so if your example can match to that release it's will be wonderfull. Thank in advance. Harry. -------------- next part -------------- An HTML attachment was scrubbed... URL:

i have posted this before but was unable to resolve it. i have some new info so i figured i would try again. the trace from bandwidth.com are below. they are telling me that the ip that is bold should be our ip not bandwidth.com. i have changed every setting that i can see and nothing fixes this. Where would i change this at? they cannot tell me. INVITE sip:+185993133333 at 216.82.224.202