similar to: SIP Transfers (Possibly reinvite)

Like everyone else who used asterisk with broadvoice, my outgoing calls died last week. I made the appropriate changes, and now basic incoming and outgoing calls are working. However, I have a few call-forwarding rules that are no longer working. It's certainly no coincidence. I can dial to all these number directly, but the problem only appears when there is an incoming broadvoice call, and I

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

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

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

I have just installed * from the latest CVS and I can make calls via X-Lite to outside numbers but for only 30 to 35 Seconds at a time then Broadvoice will hang up on me but X-Lite will not know it. Once I hang up X-Lite then I will get the following error message: Spawn extension (default, Number-I-Dialed, 1) exited non-zero on 'SIP/200-6a22' -- Got SIP response 404 "Not Found"

Hello everyone, Well here is my initial posting to the list, and I will admit Asterisk is new to me. I just got everything running here a couple days ago, so still learning the ropes for sure. OK, here is my problem. Currently I have it setup talking to a couple Cisco IP phones, and some Xten softphones, this works great. I also got an account with FreeWorld Dialup using IAX2 and that

HELP! Ok, so I have the following SIP.CONF: [general] context=default port=5060 bindaddr=10.1.1.200 externip = XX.XXX.XX.XX localnet=10.0.0.0/255.0.0.0 disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw register => ##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234 [sip.broadvoice.com] type=peer host=sip.broadvoice.com fromdomain=sip.broadvoice.com

I've been wondering for a long time why my reinvite option is not working with Broadvoice anymore. It happend during the time Broadvoice was having a lot of issues, so I decided to wait. Recently I decided to test the same sip.conf with another VSP (SIPphone) and it worked fine! No issues on the reinvite. Note: clients and server using ULAW (only), no NAT or firewalls, public ip address and

I did a quick google search of the lists site and couldn't find a definitive answer, so if it's there, I apologize for asking again. Starting about noon yesterday, I am no longer able to send/receive calls via Broadvoice. When calling in, I get a fast busy, and when calling out I get the following error: -- Executing Dial("SIP/112-572a",

Hi List, I hope this setup must be done by our astersik users.. I am using Sipura 3000 to receive PSTN calls and forward those calls to asterisk for voice processing and after that, I am transferring call to extension through FXS port on SPA 3000. Currently, media of call is trombone through asterisk. i.e achieving blind transfers on asterisk with SPA 3000. Is it possible to stop trombone

Hello everyone, Having recently gotten Broadvoice inbound DTMF to work (thanks Greg)...I am now running into a frustrating problem...when a call comes in to the BV number via a cell phone (tested with 3 different cell phones; albeit all on T-Mobile) the beginning of the IVR welcome audio is cut off. A call placed via a landline phone over the PSTN to the BV number does not exhibit the problem.

I have been going crazy with this also since Sat. Our server was working perfectly with BV but will now not place calls to BV. Incoming from BV works fine. I felt sad rebooting it today, it had been running for almost 200 days! Here is my error message from the console... Notice I am running today's CVS Connected to Asterisk CVS-03/08/05-14:32:39 currently running on com (pid = 1624)

I just signed up for the BroadVoice service a few hours ago, but for the life of me I can't get any incoming voice. The incoming connection is fine as it rings my extension from outside, but I can't hear anyone talking. Outgoing voice is working fine though. I've been looking through the archives, but I haven't found a solution to the problem yet. I even tried another router

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,

Hi All First time poster, long time reader. I just ran into something really bizarre. I've enabled videosupport and have been testing sip calls with Xten Eyebeam software, it works (mostly) However, when I have Videosupport=yes In the [general] section of my sip.conf, broadvoice calls fail w/ "We're sorry your call cannot be completed at this time" So... I've

Hello Everyone. I usually find my own solutions for problems but this time, after several months, I've given up. My asterisk is set up so that incoming calls from my voip provider ring on both my sip extension and my cellphone at the same time. When the system receives an incoming call, ringtones indicating that the call is being connected play normally for the first 5 seconds to the

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

Is there a way to do the following scenario? 1) my asterisk box receives an incoming call from a toll free number provider such as nufone, voicepulse, etc. 2) It then dials a number via SIP and outputs a DTMF sequence. ok, that part we do every day. 3) After DTMF though, is it possible to get the two SIP channels (original SIP caller plus SIP called) hooked together and have my pbx no longer

I've got a problem with DTMF, again. My asterisk box is connected with the outside world (PSTN) via a sip proxy. The problem is that for some reason, I need to use rfc2833 for signaling digits to the gateway and inband to accept digits from outside (eg. when someone dials one of our DIDs). It's possible to do this? I've ever tried splitting 'peer' and 'user' part in

Hello, I have been researching Asterisk and have a few questions. Is it possible to allow users to only call certain area codes? Reason for this is that the plan from BroadVoice allows unlimited calling to certain countries. Any country outside of this plan would be charged on a per minute basis and I am trying to avoid that. Another question is if there is a way to distinguish a land line