similar to: Broadvoice problems again Attn: James

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

How does one do translation for calls that come in from other pbx's where the incoming caller ID is an internal extension number on their pbx? Eg. when I get a call from Free-World-Dial the CID shows up as "429102" which is essentially their internal extension number sans any routing prefix. To dial the number back I need to dial the extension with FWD's routing prefix

Hi everybody, Is the only way to use asterisk _not_ as root to change the permission of all the directories where asterisk need to create a file? ("/var/run/", "/var/log/asterisk/messages") any help will be appreciated, Cyprien

Hi, I compiled asterisk and chan_h323 on an Opteron in 64 bit mode. In the h323's Makefile I replaced in line 24 CFLAGS += -march=$(shell uname -m) by CFLAGS += -march=k8 and also tried CFLAGS += -m64 -march=k8 Both solutions do compile, but when starting asterisk, a load error occurs: undefined symbol: _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi When I grep

Hi All, Last week I started hearing a huge amount of local end echo on incomming calls. I am using a Sipura SPA-3000 as my FXO connected to an SBC POTS line. Echo cancellation is enabled in the SPA firmware. As a test I switched to a Digium X100 card the still lives in my server but the echo was about the same. I have both Polycom IP600 and SNOM 200 phone, which both hear the echo. I'm

http://yro.slashdot.org/article.pl?sid=04/08/04/2212251&tid=158&tid=95&tid=103 Probably some of you already saw this. Now, beyond discussions regarding the legitimacy of such a ruling (whether they have the legal, moral or whatever right to enforce it), there's the technical aspect. Suppose i provide VoIP services using Asterisk, and i fall under the incidence of the FCC ruling

Hello, I have a problem with connecting a Digium X100P card to a Brazilian analog line. Can somebody help me out with this problem? My /etc/zaptel.conf is loadzone=br defaultzone=br fxsks=1 My /etc/asterisk/indications.conf [general] country=br [br] description = Brazil ringcadance = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 congestion =

Ok we have found a better solution. Put everthing back the way it was and make sure that you have this line in your general section of you sip.conf file: srvlookup=yes We have added a SRV entry in the correct place now. So everyrthing should go the correct servers. -james --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 /

---------- In?cio da mensagem original ----------- De: asterisk-users-admin@lists.digium.com Para: asterisk-users@lists.digium.com Cc: Data: Mon, 12 Jul 2004 11:48:05 -0500 Assunto: Asterisk-Users digest, Vol 1 #4502 - 11 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide

Since Thursday evening my asterisk box has been failing to register with broadvoice. I haven't changed any of my config files in the last week. Can anyone suggest anything? Asterisk is reporting: *CLI> Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout: Registration for '703XXXXXXX@147.135.8.129' timed out, trying again -- Got SIP response 404 "Not found"

... I just tried again after removing my hosts file entry (again) and outbound is now working! I had taken it out before, but I think I was getting a different error at the time. Sometimes it seems like asking for help is itself a cure! Thanks anyway! JDC

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

I was hoping someone can help with a problem with my GrandStream Budgetone "hanging" after a while. Problem seems related to the SIP registration - I am using dynamic registration (host=dynamic in sip.conf). Static IP is not an option in my case. I start Asterisk and all is groovy, phone works fine and can dial around. Then, at the time specified on the phone in the "Register

There is now a peering arrangement between e164.org and FreeWorldDialup which means any and all subscribers on FWD are now easily able to make enum calls by prefixing their call with **164, like wise it's almost as simple to make a call to FWD by hitting 8829990<fwd number> This means that for those of you wanting to send/receive calls to/from FWD subscribers you can now do so, easily

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

Suddenly my broadvoice will no longer register. It was working fine for over 1 month without a single problem, now I get a SIP registration timed out message. I called them, and I was told that they are experiencing problems, and they hoped to have it resolved ASAP. I called them at around 10 AM EST this morning. It's now 8:30 EST PM, and I still have not heard back, and the problem is not

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

Jordan, I have sent the patch via Phabricator: https://reviews.llvm.org/D69188 Let me know if I got it right. -- Tobias On Thu, Oct 17, 2019 at 7:12 PM Jordan Rupprecht <rupprecht at google.com> wrote: > > Tobias, > I don't have much experience with ARM, but from your report and Peter's explanation of why LLD does it, I agree we should be consistent with LLD and keep the

On Tue, 15 Jan 2019, Hans Wennborg wrote: > On Sat, Jan 12, 2019 at 12:49 AM Jordan Rupprecht <rupprecht at google.com> wrote: >> >> >> >> On Fri, Jan 11, 2019 at 6:26 AM Martin Storsjö <martin at martin.st> wrote: >>> >>> Hi, >>> >>> The COFF support in llvm-objcopy is in a pretty half-finished state at the >>>

I believe a good amount of them (if not most of them) have already been ported! Off the top of my head, I remember that asan, tsan, msan, hwasan, the kernel santizers, and sancov have been ported. I don't think ubsan has been ported yet though. You can also check if other passes you need run under the new PM by checking PassRegistry.def. On Fri, Aug 2, 2019, 14:20 Jordan Rupprecht