similar to: Testing a sip call through Asterisk?

Hi, all. The asterisk setup is working fine, receiving calls via broadvoice "initially". ? When call comes in via broadvoice number, asterisk picks it up and routes correctly, as long as the call came in within ~2 min from the previous one. In other words, as long as a call comes in within ~2 min since the previous one, asterisk will answer the call. However, if the call comes in

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

Hi list, I have for a year I have an account to call with broadvoice from about 3 days beginning a not registered problem of, asterisk shows to a message of error with the DNS, and my dns this working fine WARNING[5770]: chan_sip.c:7595 transmit_register: Probably a DNS error for registration to 908XXXXXXX at sip.broadvoice.com@sip.broadvoice.com, trying REGISTER again (after 20 seconds) [Dec 16

I just signed up and configured a SIP connection from BroadVoice. It works great. This issue I have is that it seems after a couple calls (or a certain amount of time) Asterisk doesn't seem to be receiving these calls anymore. It seems as if BroadVoice is not redirecting the call to my Asterisk. Asterisk still seems to be ready for the call: *CLI> sip show registry Host

Anyone know how many simultaneous calls you can receive on a NuFone DID? -Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050310/e20d2eb8/attachment.htm

Apologies if this is a repeat: I trawled through the archives and couldn't find a reasonable answer, so I'm asking here. I have an Asterisk install connecting from behind a NAT device (DSL modem) to a SIP proxy (in my case, Broadvoice). I have an sjphone softphone on a Windows PC also behind the NAT device that connects to the Asterisk install, and using this setup I've been pretty

I am running RedHat 7.1 on it's own linux drive, and have my win98 drive mounted in fstab as vfat. I have installed wine, and can run programs on my windows drive in old 8.3 named directoies. However, whenever I try to run anything in Program Files, for example (with the space in the file name), I get the following error: /opt/wine/bin/wine.bin: cannot find 'mnt/c/Program\

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

I've tried everything with the * box after this weekend. I have read every document on the problems people are having with them after this weekend as well, but none of them address my problem. I checked my settings in my sips which I have below as well, I have changed the host file a few times, but this was new to me and I never had modified it before. I have and the same results

Hello, Have a weird problem when using asterisk (1.0.6). There are certain numbers I cannot dial when using asterisk with my broadvoice account. No problems with inbound. With outbound calls, I can call some numbers (for example broadvoice customer support number) and unsuccessfully with some. However, when I configure my account directly on x-lite, I dont see these outbound problems. Here is a

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

Hello All, I'm just getting into *, and trying to use a Broadvoice account. It works inbound, but Outbound fails no matter what sip.conf parameters I try. From the recent posts here I think it could be: A bad CVS release - I will try to download and build from a new one Broadvoice not challenging and/or Asterisk not responding with an Authorization: in the INVITE header. I am

Hello, Please forgive me if I'm repeating this post. I have searched and looked for similar problem with a solution but have not see a similar one. My outgoing SIP and other channels work fine but the incoming/inbound SIP call goes straight to Broadvoice voicemail. I see that Broadvoice is registered when I look at the SIP registry. I have turned on SIP Debug and it is below. Anyone know

Hello all, When I call the Broadvoice number all is good. When I try to call out through DISA on my broadvoice line i get the following: Executing Dial("SIP/147.135.0.129-0815bc60", "SIP/16037862111@proxy.bos.broadvoice.com|30") in new stack -- Called 16037862111@proxy.bos.broadvoice.com -- Got SIP response 480 "Temporarily Not Available" back from

Egad, not again with Broadvoice! Anyhow, I recently installed AAH and configured my TDM11B and got that and some SIP phones working. I still have some issues to work out, etc, but my current problem is Broadvoice. I have checked out all of the online resources, including the recent list exchange about the recent changes made by Broadvoice. However, the one thing I have found to be consitent in

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

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

Hi all, I just upgraded to Asterisk 1.4.18 a few days ago and I don't use Broadvoice TOO often, however I have a Vermont number with them and so my mother in law calls it to talk to my wife once in a while, so that's why it took me so long to notice it wasn't working. Anyway, when she calls she gets a busy signal (as I've tested when calling it from my cell). When I enable

All, I have been fighting with my dialplan for hours now, and google searches talk lots but offer nothing in terms of explication for this. I have my SIP peer set up and working with Broadvoice: [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=5555551234 secret=password defaultuser=5555551234 insecure=port,invite context=broadvoice

I just started service with Broadvoice.com and everything seems to work. However, apparently my understanding of incoming sip contexts is less then what I thought it was. Could someone point me in the right direction? (* on a public address, CVS-HEAD-07/12/04, C7960 phones) In my sip.conf I have: [general] port = 5060 bindaddr = 0.0.0.0 allow=ulaw tos=0x18 ;sets ip tos bits (=lowdelay and