similar to: I have to register asterisk/sip with a sipproxy that does not support authentication?

Distro: Archlinux 2.6.38 Wine version: 1.3.17 (The system is fully updated) Code: Affected folder: ~/.wine/drive_c/users/Public/Application Data/Adobe/Photoshop Elements/5.0/Photo Creations/special effects/photo effects/CoffeShop Actions Code: Unaffected folders: ~/.wine/drive_c/users/Public/Application Data/Adobe/Photoshop Elements/5.0/Photo Creations/special effects/photo

hi im trying to install dovecot and use ldap as the backend and i get the following error srv:/root# telnet localhost 143 Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. * OK [CAPABILITY IMAP4rev1 SASL-IR SORT THREAD=REFERENCES MULTIAPPEND UNSELECT LITERAL+ IDLE CHILDREN NAMESPACE LOGIN-REFERRALS AUTH=PLAIN] Dovecot ready. 1 login b.tapratzis passwordhere 1 NO

Dear all, Given such vector of array. tags <- c("aaa", "ttt", "ccc", "gcc", "atn") How can I obtain a matrix corresponding to it [,1] [,2] [,3] [1,] 0 0 0 [2,] 3 3 3 [3,] 1 1 1 [4,] 2 1 1 [5,] 0 3 0 In principle: 1. Number of Column in matrix = length of string (= 3) 2. Number of Row in

On Jul 15, 2019, at 11:15 AM, David White wrote: > > 2. When I check the results from "netstat -t -n" I am NOT finding anything on 3493. Hmmm. I then tried "netstat -l" since there should be a server socket listening on 3494, right? There is nothing of 3493. But I do see an entry with local address = localhost:nut. When I "cat /etc/services" I find nut listed

Still no luck using asterisk as a proxy. 48 hours solid working on this. I'm beginning to think asterisk isn't going to be compatible with the provider I'm using :( Has anyone got *any* clues as to what can cause this message? It's definately provider specific (voiptalk works, pipecall doesn't) but confusingly seems to be caused by something in the client phone app. I

Hello. I'm trying to use Asterisk in combination with SER, to make the routing proccess to my PSTN-Gateways. I made a simple test defining some extension in my extension.conf, when i made a call my SER (SIP) Server forward the call to Asterisk, this proccess is ok, but when the call is answered i see an INVITE going out from Asterisk to my SER Server, this invite is then passed to my

I've been looking through the examples and docs, but haven't yet quite figured out how to do what I want. We've got a T1 coming in carrying a block of telephone numbers, terminating on an Asterisk box. Any call that comes in needs to get sent to a SIP proxy, with a particular extension format: *ANI*DNIS*@sipproxy.address The closest I can see to do this with the Dial() command is:

Hello list. My phone rings, I pick up, and the conversation is terminated. Every time. The setup : Grandstream GXP2010 --> SIPproxy (Endian Firewall) --> Asterisk Server --> ITSP Could it be the SIP proxy of my Endian firewall ?? I have 4 accounts on the Grandstream which listen on port 5060 --> 5063. They have a proxy defined namely my Endian firewall. On this SIPproxy I have a

I have been a reseller & subscriber of pipecall since they started, however I am really struggling to get pipecall to work for outbound or inbound calls. I get errors that the registration has timed out. I have tried many variations of the register command register => 0845xxxxxxx@sipproxy.pipecall.com/1000 register => sipxxxxxxxxx:xxxxxxxxxx@sipproxy.pipecall.com/1000

The Wine development release 1.7.25 is now available. What's new in this release (see below for details): - Implementation of the packet capture library. - A few more DirectWrite functions. - Improvements in HTML table support. - More VBScript math functions. - Various bug fixes. The source is available from the following locations:

Hello, I am using OpenSIPS to register all the users and planning to use asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge. I have a scenario where the signaling does not happen properly: 1) A user from Opensips dials an extension 7000 which is an auto-attendant extension. The call is routed to asterisk to play the auto attendant messages like Welcome and Dial the

How do I get the bit like "IAX2/white_phone" in extensions.conf eg from pre-defined variables or variants thereof ? What I *do* get is strings like this "IAX2/white_phone@white_phone-4" from ${CHANNEL}, but that's the full channel name. What am I missing here ? Thanks, Thomas

Hi Alok, Joining this a bit late as this was just brought to my attention. It would have been nice to CC the virtualization mailing list. Please do in future submissions. Alok Kataria wrote: > VMware PVSCSI driver - v4. > / > > diff --git a/drivers/scsi/pvscsi.h b/drivers/scsi/pvscsi.h > new file mode 100644 > index 0000000..96bb655 > --- /dev/null > +++

Hi Alok, Joining this a bit late as this was just brought to my attention. It would have been nice to CC the virtualization mailing list. Please do in future submissions. Alok Kataria wrote: > VMware PVSCSI driver - v4. > / > > diff --git a/drivers/scsi/pvscsi.h b/drivers/scsi/pvscsi.h > new file mode 100644 > index 0000000..96bb655 > --- /dev/null > +++

Hi, I have access to two providers. On one of them the authuser is the same as the username, so outgoing works. On the other one I can only get incoming - what ever combination I try for outgoing I get an error. The register command has the ability to specify both usernames (which is why incoming works) but outgoing doesn't seem to, and without that I'm stuck. They are defined as:

Hi Tony, Try adding "fromuser=xxxxx", maybe "username=xxxx" isn't enough... Just a guess, it already solved a few problems for me. -Manuel -----Messaggio originale----- Da: Tony Hoyle [mailto:tmh@nodomain.org] Inviato: martedì, 18. maggio 2004 13:03 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Configure asterisk for outgoing.. need authuser parameter?

I think I have got * worked out as far as getting users on a small private network talking with each other, but when it comes to the bigger picture about talking between private networks connected by the Internet then I am getting confused about STUN, SER, SIPPROXY, RTPPROXY, etc. Before I start let me make it clear that I am not looking to drop out onto the public telco network anywhere, not at

Hi, ALL: When I use astcc to do the prepaid function, but if I want to enable "call forward". The result of CDR seems not correct. UA 1011 make a call to UA 9999, and UA 9999 forwards this call to a PSTN number. I think we shall charge the credit from UA 9999 not UA 1011 because UA 1011 don't know where UA 9999 forwards to. But in CDR, I can only find the from(1011) and

Guys, I've downloaded AsteriskNOW few days ago so I'm new to this product. The first issue is on service provider area. I've already used a VoIP account already configured with my ISP, it works fine! This configuration has been used until now with the client SJphone, Now I would use this profile as main VoIP service provider to setup in AsteriskNOW. Here are the profile detail as

Can someone explain what this error is? -- Got SIP response 500 "Server Internal Error - Invalid CSEQ number" back from 209.xxx.xxx.xxx How do I fix this? .o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office