similar to: Anonymous Connection form IP to use specific Context

All, I'm hoping someone can direct me as to why when someone calls my DID Asterisk tries to authenticate the incoming call on my outbound context. If I remove the GoTalk context I can receive incoming calls. Outbound calls work fine while I have the GoTalk context in place. The error I am getting when someone calls the DID is WARNING[16072]: chan_sip.c:8272 check_auth: username mismatch,

Hi All, I'm having problems trying to get a TE120P operational in Canada. I keep getting a congestion error when I try to make a call. I'm not sure if my switching, parity, etc is correct. I'm hoping that someone will be able to verify my config. The Telco is SaskTel, with a 10 channel 50 DDI service. Zap show channels show and ztcfg -vv looks ok and the zttool show

Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I'm not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David

Hi All, I'm hoping someone can recommend a method to integrate Microsoft CRM with Asterisk. Preferably an open source product otherwise a commercial product. Regards David Klaverstyn -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130716/f931d763/attachment.htm>

Hi All, I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. Regards David. -------------- next part

Hi, is there a way to anonymously connect a Windows client to a Samba PDC? I have some random clients which I currently don't want to add to the domain, but would still like them to have access to one share. For example, I have: [skener] path = /var/spool/scanner/scans browseable = yes writable = yes force user = scanner force group = scanner

Hi All, Can someone please tell me if it is possible and if so how do I go about streaming a live conference to the internet for internet users to listen to? I'd hope to be able to do thus dynamically as conferences are created and internet users can tune in via a browser or streaming through media player. Regards David Klaverstyn -------------- next part -------------- An HTML attachment

I would like to be able to use anonymous connections in R and have them close themselves when they go out of scope. Here is an example of what I think should work, but does not at present: ## create test file x <- 1:10 fn <- "anon-con-test-x.rda" save(x, file=fn) testUrl <- paste("file:/", getwd(), fn, sep="/") ## use an anonymous connection to load data

Hi All, This may be an off topic but I'm not sure who else would know the answer. I'm playing around with Asterisk and Polycom phones. I see Polycom supports quite a few codec. The usual ones and these: G722 Siren14.24kbps Siren22.32kbps Siren14.32kbps Siren22.48kbps Siren14.48kbps Siren22.64kbps G7221.16kbps

Hi all Maybe somebody has an idea. I'm tracing a very strange phenomena... I've a connection from Asterisk to a SIP PBX. Most calls have a caller ID. Some International calls don't have any. Now it looks like those calls without caller ID never get to the context where incomming calls from this SIP PBX should get to.... Examples: Call with Caller ID: (slightly anonymized)

Good afternoon; I need to set up a custom form (paper size) on a printer that I share out with anonymous access from v. 3.0.1 on Solaris 8 to Windows 98 & XP Pro clients. Should I change the server to require a password and then set this (according to pages 236 ff. of the new Samba book) and change it back, or is there another way? Thanks in advance for any suggestions. -wde -- Will

Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am running 2.2.5 and I would like to know if the "restrict anonymous" as been implemented correctly, as it was supposed to behave from the start, in order to deny ALL anonymous connections as stated in the man : "When restrict anonymous is yes, all anonymous connections are denied no matter what they are for." Ive been reading some dev mailing lists and someone said

Hi All, I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium cards. The problem I have is that MOH will not play. It starts and then stops. asterisk*CLI> zap show status Description Alarms IRQ bpviol CRC4 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 I'm not sure if the above is correct.

With pjsip (asterisk 13.14.1) I see the problem that an anonymous from header gets user=phone appendend to the URI if user_eq_phone=yes is specified: On the incoming leg: From: anonymous <sip:anonymous at anonymous.invalid:5060>;tag=Q5zBj7BMnvI6Fe6O2866fox3ZHmn-smt Get transformed to From: "Anonymous" <sip:anonymous at

Hi All, I can not install the asterisk-addons as it thinks there is no mysqlclient installed. I have installed mysql, mysql-server and mysql-devel and I am still unable to install the addons. I am running CentOS 5.2 i386. Please somebody help. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, just started using dovecot on FreeBSD. Nice and easy to configure :-) I then wanted to browse the dovecot mailing list, so, according to http://dovecot.procontrol.fi/mailinglists.html: IMAP archives available from dovecot.procontrol.fi, either use ANONYMOUS authentication or give anonymous as username and empty password. I was using a patched sylpheed that doesn't support

Does anyone have experience setting up * to accept anonymous sip UAs and the dumping the call into IVR? I'm thinking this would be a good way to have customers call us without creating an extension. So for my tests have been focused on providing internal functionality. Thanks, Chad -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have tried to disable anonymous in smb.conf, by include the global: restrict anonymous = true password server = * encrypt passwords = yes null passwords = no Buy when I ran the : ./smbclient -L myserver -N I have this message: added interface ip=111.xx.xx.xx bcast=111.xx.xx.255 nmask=255.255.252.0 Got a positive name query response from 111.xx.xx.xx( 111.xx.xxx.xx)

Hi guys, I am trying to set the caller ID to 'Anonymous <anonymous>' if the caller is not registered to the asterisk server. But I can't find a solution. Any ideas? Regards Philipp -- Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a