similar to: Recommended Inbound/Outbound/DID Provider which supports TLS

Hi all, I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. Thanks for any

Hi, According to IETF draft draft-ietf-secsh-transport-14.txt, different ciphers(encryption), MAC and compression can be used for one direction say server-to-client and a completely different cipher, MAC and compression for the other direction client-to-server of the same connection. Is this supported today in OpenSSH, and if not, are there plans to support it in any future releases of the code?

Last night Mark and I made some changes to the IAXTEL tollfree outbound, and inbound access. The inbound access number has changed to: 248-724-0700. (This number is in Pontiac, MI Ratecenter, and is supplied by Telesthetic LLC, a next gen phone compnay) This number will say "Please dial your number now" at that point you can dial your 1-700-XXX-XXXX IAXTEL number assigned. In the

silly me, I thought the inbound would be the hard part. how little I knew... can someone please give me any insight into why outbound is not working, in fact why trying to enable outbound fouls up everything? I'm using asterisk, most recent from cvs, I'm behind a nat, and I'm trying to use iconnecthere.com for outbound and inbound. Inbound is working fine, no problems. But for

Hello All, I would like to record inbound and outbound calls to and from one number. I tried to add lines to my extensions.conf: DAY=`date "+%m-%d-%y_%H:%m"` ;outbound exten => 5555551212,1,Record(${DAY}:gsm) exten => 5555551212,2,Dial(${TRUNKL3}/${EXTEN}) ;Inbound [line2] exten => 5555551212,1,Record(${DAY}:gsm) exten => 5555551212,2,Dial(SIP/101,20) exten =>

Hi to all, I need some help, I have an Asterisk Server in a small call center, for inbound calls I setup a Queue in queues.conf and their respective Agents in agents.conf, but when an Agent is calling out and a call is coming from PSTN the call is send to that agents which have a call in progress. How I can fix this in order to have only one call at a time. I think in limitonpeer and call-limit

I'm trying to set up a 2nd DC.  The primary is a Server 2003 (ads1).  But DNS is being done on a Linux proxy server.  It appears that I have joined correctly (wbinfo -t: checking the trust secret for domain SAMDOM via RPC calls succeeded).  But when I check inbound and outbound (samba-tool drs showrepl) I get the following: > Default-First-Site-Name\ATHENA > DSA Options: 0x00000001

Greetings list- I have a box with a single FXO card in it. I'm able to dial out ZAP/1 with no problems and as expected. However, I would like inbound calls on that POTS line to go unanswered by Asterisk since I have other equipment on the line. I've setup zapata.conf for the channel without a context but the line is still answered. I've also setup a blank context with the same result.

Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this

I need to monitor the states of my T1/PRI Zap channels. Specifically, I need to be able to programmatically determine whether a channel is unused, carrying an inbound call, or carrying an outbound call. Using the manager interface, I can easily tell whether a Zap channel is used or not by looking at the results of: Action: Command Command: zap show channel <x> Or: Action:

Hi, I am using the following script to limit my outbound traffic. This scipt runs on a box behind my firewall. It limits my outbound passive ftp traffic to 39K perfectly....just like i want. However, i just noticed that it is also limiting uploads coming to my server. Is there something I can change to make it not limit uploads to my server? #!/bin/bash #shaping passive ftp traffic #

Dear all, I think this is common, or at least how it is supposed to be, but whening dialing over a ZAP channel, it's taking around 5~ seconds to ring on the over end, likewise inbound. This is just with a normal Dial command. Are there any ways to tweak this? Thanks, Gavin.

I have a strange problem in that when I receive a call through the X100P which is forwarded to my budgetone 100 then the voice quality is perfect both directions. However, if I make a call out from the budgetone to the same caller via the X100P the sound level is a lot lower and the quality a lot poorer. I've had to set the rx tx gain to 1.5 or I can hardly hear at all. Any ideas what is

Hello I get this warning all the time when I am using iax2 for inbound calls or outbound. Aug 24 13:48:41 WARNING[-1105474640]: chan_iax2.c:4873 socket_read: Error: Resource temporarily unavailable I get the calls and the sound is fine. But the screen on the cli is full of these warnings and Error: What can I do to fix this. I get it when using calls to iaxtel, FWD, VoicePulse, Nufone and

I can't find this in the wiki/email history..but I'm sure it's based asked before. The port range define in rtp.conf - is that for connections initiated by asterisk? Or the port range asterisk listens on? Or both? Thanks! MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a way to specify the outbound leg Call-ID? -- Eric Chamberlain

I wrote a note earlier about this problem but have done quite a bit more debugging. Now I'm stuck at what to do next. I have inbound calls being answered by our Asterisk box, which then dials our answering service and bridges those calls. The inbound and outbound are both PRIs. The answering service takes our calls on a PRI. If I don't use the Monitor() application, things work find

Apologies for what might seem a lame question but I recently had a set-up where both sides of an agent conversation were combined into a single file using SOX. I've spent hours trying to find this and simply can't, can anyone point me in the right direction, I've tried all sorts of searches at voip-info and here too. I'm running *@home 0.6 if it helps. Steve

If I have a call coming in on one line and then I am using Dial to send it out to another outside line, is there a good way to get separate CDR for both call legs? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060317/ac95b2fc/attachment.htm

Hi all, In my case I using realtime, here is how it looks in plant [10001] type=registration transport=upd_static outbound_auth=10001 server_uri=sip:600 at 192.168.1.1:5060 client_uri=sip:600 at 192.168.1.4:5060 [10001] type=auth auth_type=userpass password=600 username=600 [10001] type=aor contact=sip:192.168.1.4:5060 [10001] type=endpoint transport=upd_static context=dialmap disallow=all