similar to: Hardware vs. Network Inputs

Hi all, I've been trying to use BroadVoice as a SIP service provider. They don't officially support * but are helpful when it comes to answering questions for setup parameters. They claim they have no firewalls or access lists that need to be set up so I can get access to their servers. However, something's still not quite right when I use the parameters. It looks like our Asterisk

I have a SIP trunk to Broadvoice. My Asterisk box (1.4.13) is on public addresses behind a firewall. Originally it was behind a Linksys WRT54G running sveasoft. Sveasoft really can't NOT do NAT even when you turn it off. My Asterisk box is defined as the DMZ box to Sveasoft and it seemed like it was leaving all packets alone. Now I switch to a Centos-based firewall configured with

Hello, I'm planning to convert my phone system to Asterisk, as I've outgrown my TalkSwitch system. I have a few questions for experienced * users, most of which can be answered yes/no. Current Setup: - Talkswitch 48NLS (4CO/8Ext) phone system. - One CO line, two Vonage lines, one Voicepulse line connected to phone system - A third Vonage line directly connected to a fax machine - A

Hi, I found E400P quad PRI card quite cheap (749USD): http://www.govarion.com/product_info.php?cPath=1&products_id=2&osCsid=68cdd6e3d08754 in comparison to te410p (approx 1500 USD ) http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE410P Now newer generation with HW echo canceling emerged (te411p). I'm not sure in what things those two cards

I had my asterisk configuration working very well with broadvoice, but it stopped working this afternoon. I plugged the Cisco 7960 phone I used for my origional signup (just a few days before they offered generic BYOD) and it works fine. I did notice it seems to do all of its comunication through proxy.broadvoice.com (I used tcpdump). I have never contacted broadvoice about using asterisk

I am thinking of dumping broadvoice so I need some other VoIP providers that have a las vegas DID and a service better than broadvoice. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050131/bf4b1c31/attachment.htm

I have an idea, tell me if this wouldn't work... I know it's really ugly, but it might help some people until we can get round robin DNS checks for peers... Since * does not do GetHostByName() again until you reload your config, and BroadVoice and I'm sure other sip providers are using round robin DNS, why not create 2 [<your server here>-out] contexts in sip.conf, and then in

Hey there, I don't know who else has suffered broadvoices terrible service, but I am about to my end with them. The lack of a LBR codec, the outages, the changing of servers without notifying subscribers haspushed me to my end. Now most incoming calls are abbruptly cut off within a minute of the call starting. Anyone know of any other * friendly providers that have DID, besides Voicepulse,

hello, i installed an asterisk as a pri gateway. Everything is okay. /etc/init.d/zaptel starts successfull with wct4xxp module. /etc/init.d/asterisk starts also successfully. I tested my pri cable and it works. But still my span isn't up. I don't see any error. Do you have any idea? What else i should check? Thanks. My card is 4 span Wildcard TE410P

Which would you recommend as far and quality and pricing to connect to asterisk (including DTMF issues)/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050125/bf7bf081/attachment.htm

I run an automated information retrieval system, using Asterisk. Fairly often the system misses a dialed digit. Our codes are all 4 digits, see lots of logs with: 4199 - OK 530 - Invalid code 330 - Invalid code 5330 - OK As callers experience skipped codes. We're using Broadvoice SIP with inband DTMF (and we've tried every possible setting or option

Hello, I cannot accept any inboud calls from any provders in my asterisk which tries to authenticate the provider and at the end rejects the call with tthese message 407 Proxy Authentication Required How do I turn off this message. Thanks. Ehsanul Karim

Greetings. I did some digging with Google, the wiki, and on the archives, but didn't find a recent conclusive answer. If this is answered in the wiki or archives somewhere, please point me to it. I'm in the process of setting up an Asterisk box for home use. I've got a X100P card on the way. I've not decided what analog adapter(s) to get yet. The only phone service to hook up

I've been looking for a good iax or sip <==> ptsn provider. Someone with very low cost usa calling and can offer incoming ptsn connections in most markets. The only decent providers I could find were iconnecthere and nufone. Has anyone found someone that really stood out? Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office:

Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John

Hi, I have two accounts with broadvoice. Now, I want to be able to distinguish between them. I though that this would be simple by adding "/EXTEN" at the end of the register statement. For example: register => num1:pass@sip.broadvoice.com/1000 Unfortunately, this is not working. When I call into my box I hear busy tone. My config looks like this: [root@voip asterisk]# cat sip.conf

Is the Broadvoice service up? I just signed up with them and started receiving calls in no time but could not make calls. And after a few minutes I cannot even place calls. register => [number]:[password]@sip.broadvoice.com [broadvoice] type=peer fromuser=[number] host=proxy.lax.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice dtmfmode=inband any help would be

Until recently, I was using Broadvoice for my in/out calling thru Asterisk. I was extremely pleased to see that Broadvoice was actually passing the callerid info (number and text) that I had set up on each device in my SIP.CONF file. I had PSTN users tell me that they were actually seeing name and extension info when I called them from the Asterisk box. Last week, due to numerous user quality

I have added the three lines to the sip.conf file based on the latest changes from broadvoice. I can receive incoming calls but cannot place any outgoing calls. The error I get is: *CLI> -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569 -- Attempting call on SIP/Broadvoice/5068012 for application Playback(demo-congrats) (Retry 1) Mar 8 08:35:21 NOTICE[29290]:

Hello All, I have 4 Broadvoice lines. If I call any of the lines it shows that is coming from the first line. exaple register=XXXXXXXXX1@sip.broadvoice.com:passwd:XXXXXXXXX1@sip.broadvoice.com register=XXXXXXXXX2@sip.broadvoice.com:passwd:XXXXXXXXX2@sip.broadvoice.com register=XXXXXXXXX3@sip.broadvoice.com:passwd:XXXXXXXXX3@sip.broadvoice.com