similar to: SIP: Failed to authenticate

Calling all SIP gurus-- I'm trying to register my asterisk to an ISP's SIP gateway. I'm getting authentification errors. Here's the results of SIP DEBUG against it's IP. [I've tweaked all confidential fields so as to protect the innocent (namely, me).] --- (9 headers 0 lines)--- Responding to challenge, registration to domain/host name myfavoriteisp 12 headers, 0

Hello-- I've been playing with the privacy options on my home/home-office system since August last year, and have some results, gleaned from my CDR records, which over the last 13 months, number a total of 8672, which includes incoming, as well as outgoing calls. Before I start spitting out numbers, let me note that with the current setup, I haven't had to tell a single telemarketer

Hello, I have defined 8 registers in registerinfo.td file in the following order: R_0, R_1, R_2, R_3, R_4, R_5, R_6, R_7 But the generated assembly code only uses 2 registers. How to enable it to use all 8? Also can i control the ordering like after R_0 can i use R_5 without changes in registerinfo.td? What changes are required here? either in scheduling or register allocation phases?

I'm attempting to connect to a Metaswitch, inbound only (at this time). The Metaswitch is the only "connection" (at this time). All I'm getting so far is a bunch of "OPTION" messages which my Asterisk box replies to but I don't get inbound calls. Here's my sip.conf. As you can see I've been trying a bunch of different options without success :(

I'm (still) trying to get my Asterisk box talking to a Metaswitch. All I'm getting is a "heartbeat" of OPTIONS messages coming from the Metaswitch which my Asterisk box replies to. The exchange looks like: <-- SIP read from 172.b.c.d:5060: OPTIONS sip:metaswitch@206.b.c.d:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP

I am setting up VM off Metaswitch due to a problem with Metaswitch VM. I have a couple days to prove this works and I need a little assist please. I am using TRIXBOX 2.6.2.5 and have the Meta SIP trunk up. I have extensions built that can talk to each other. I took a trace on the TRIXBOX that shows when I dial my test phone on Metaswitch it goes to VM after a couple rings and the call goes to my

Hi folks, I am having a small problem with asterisk-1.6.2.7 + app_fax on OpenBSD 4.7 -release. Everything seems to work fine. I have a macro which answers, receives the fax to a tiff, and then runs a script (mailfax) to convert that to pdf and email it. It all works perfectly except for some errors I am seeing in the console. After it hangs up I get a dozen or so messages in the cli

Hello, I''m trying to port one of our existing appliances (running on 64-bit Red Hat Enterprise Linux 6.4) to a Xen HVM guest . We''re seeing some very odd behaviour that doesn''t manifest on other platforms. The guest experiences intermittent lockups of a few seconds- shell sessions become unresponsive, and various software healthchecking in our application triggers,

Im trying to get my asterisk box to register to a sip provider without much success. here is my console output in asterisk Dec 13 12:57:17 NOTICE[213005]: chan_sip.c:3982 sip_reg_timeout: Registration for 'voicemail.nexband.com@metaswitch.nexband.com' timed out, trying again -- Got SIP response 403 "From: URI not recognized" back from 208.149.73.5 Urgent handler in my

I've been racking my brain for the last two days to try to figure out what I could possibly be doing wrong in my configuration for a SIP trunk that's setup through my local ISPs Metaswitch. I've setup a very simple SIP Peer, which I've played around with a lot in the past two days but still comes back to the following basic setup: [provider-fireball] type=friend

Hi All, I'm stumped on a weird problem. I have an * server working fine for local SIP phones and IAX2 connections. We just provisioned a second Ethernet port to attach to a local SIP provider. PSTN calls incoming work fine: PSTN -> SIP Provider -> SIP -> * but outgoing calls are not. Call setup takes place and the caller can hear about 1-2 seconds of audio before the SIP provider

I have tried everything to get BV working outbound. All worked fine until the BV change last week. I called BV and they changed me to sip gen with a new password. I stripped my Asterisk server to one phone on Zap/1 until I get this working. The same BV account works fine with a SPA-3000 so I don't suspect a firewall problem. Symptoms: Asterisk registers with BV Ok Incoming calls work

I recently configured a SIP peer which i must specify my fromuser as my ten digit "DID". I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is "asterisk". Is this a bug? Or is there some other config I must make ? register = 2155551941:123456 at 10.0.138.226/2155551941~600 [peer](!) type=peer context=inbound qualify=yes

I'm trying to send CallerID info to a MetaSwitch system over a PRI. The MetaSwitch gets the info exactly as it is sent by Asterisk, but I think it might be having trouble with the Hexadecimal b1 that is being sent just before the first character of the CallerID Name. Does anyone know what the significance is of the b1 being sent here? Or, is there a way to make Asterisk not send the b1

Greetings, I have a problem getting Asterisk registered as a UAC against the MetaSwitch call agent, because the customer insists on running it on a NAT'd box. Thus, the Contact: field in the REGISTER request betrays the private IP address of the Asterisk box, but the source IP of the message is a public one. Most registrars don't have a problem with this, including Asterisk. However,

i want to emit binary code for the following implemented vector assembly instructions. P_256B_LOAD_DWORD R_0_R2048b_0, pword ptr [rip + b] P_256B_LOAD_DWORD R_0_R2048b_1, pword ptr [rip + c] P_256B_VADD R_0_R2048b_0, R_0_R2048b_1, R_0_R2048b_0 P_256B_STORE_DWORD pword ptr [rip + a], R_0_R2048b_0 I added the following lines in X86MCInstLower.cpp; unsigned NewOpc; switch (OutMI.getOpcode())

Hi, i am trying to connect to a DB2 server using the DBI library. getData <- function() { driver <- dbDriver("DB2") conn <- dbConnect(driver,"server","uname","pword") data <- dbSendquery(conn, "select etc.") } When I run the function, i get the error > data <- getData() Error in

I am having difficulties with customer-bound DTMF being very short & clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE Asterisk is running Asterisk 10.4.0 on a

In 0.9.7 the private key encryption was switched from 3DES to AES, (https://bugzilla.mindrot.org/show_bug.cgi?id=1550) the motivation for this being that 128-bits of security is better than the 112 or so you get from 3DES these days. Interestingly that bug is about upgrading to AES-256, but we ended up with AES-128. Presumably due to the Solaris crippling? However ssh-keygen still uses a

I have been setting up Windows servers to serve windows clients for many years and am now looking at samba as an alternative. I am currently having a problem setting up samba as a PDC for windows XP clients. I can see the samba server through My Network Places. If I try to access it I can after putting in one of the usernames and passwords I have set up using smbpasswd. I have added a machine