similar to: SIP Video Camera

Hi to all, I was looking in google and also in this mailing list, but I dont find the solution to my problem, so I subscribe me to the list in order to post this e-mail and find the solution. This is the scenario: GSM Phone ----- GSM Network ---- TDM2406E --- ASterisk 1.4.0 (*) -------- VoIP Provider ------- Sip Phone or H323 Phone The problem is that I am generating calls from SIP and also

Does Asterisk supports 302 SIP message - "Moved temporarely"? I have found mail from Olle E. Johansson (April 2006) that Asterisk doesn't support redirects of registrations. Does it support now? -- Tomislav Parcina firstname.lastname@email.t-com.hr

How to execute some system command from AMI? -- Tomislav Parcina firstname.lastname@email.t-com.hr

Hi the new asterisk 1.4 supports to store queue log information directly into a database? (like CDR) ? thanks

The Asterisk and Zaptel development teams have released Zaptel version 1.2.16. In addition to minor bug fixes, this release fixes a build-time problem on systems where the default language is not English, and also corrects a regression in the driver for the Digium dual- and quad-span cards with hardware echo cancelers that could result in kernel panics. Thanks for your support of Asterisk and

The Asterisk and Zaptel development teams have released Zaptel version 1.2.16. In addition to minor bug fixes, this release fixes a build-time problem on systems where the default language is not English, and also corrects a regression in the driver for the Digium dual- and quad-span cards with hardware echo cancelers that could result in kernel panics. Thanks for your support of Asterisk and

Hi all, I've just been told from an ex workmate that my VSP (who I used to work for) has put an anti flooding limit of 80 SIP messages per IP per minute in place. I run the phone system for a facility that has a lot of extensions, but would rarely have more than 4 or 5 simultaneous external calls. Am I in danger of tripping over this limit? It sounds dangerously low to me.

I am use Fedora 3, and run into a 1.4 compile issue. When 'make install' I got this message. [root@asterix asterisk-1.4.1]# make install make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list->next != 0' failed. make: *** [utils] Aborted [root@asterix asterisk-1.4.1]#

Greetings to everybody. My question is that it?s impossible to pick up a call from ZAP, IAX or mISDN with my Ext Key of my GrandStream. It always give me a Spawn Message on CLI and a ?603? error on my LCD GrandStream. Exactly from my CLI screen i get this message -- Executing NoOp("SIP/11-096c2ac0", "Probando 1 ") in new stack -- Executing

Hello Im having strange asterisk ring. I'm dialing PSTN network, then I get my call answered and I hear a person talking but the same time remote person can't hear me. They get a ring tone. What can be the problem? Where do I need to look for it? have no clue. Running Asterisk 1.2.14 svn rev 48468 Voip gateway is Cisco5300 with IOS 12.3(9) Scheme: 192.168.1.201

please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compile&run clients with srtp (linksys,gxp-2000, minisip, twikle, ...) --------------------------------------- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA - http://lcna.slu.cz =======================================

Hi, Your colleague has forwarded his incoming calls to his secretary. How do you call the feature allowing you to circumvent your colleague call forward to make your colleague's phone ringing ? Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070326/20c37b72/attachment.htm

Hello, is there someone who knows if I can use AOC for billing in Asterisk? I mean: let's say I have an external SIP device that produces AOC data. This device connects me to the telco network. Can Asterisk, if connected via SIP with this device, collect AOC data and put it in the CDR records? If not, which is the right way to use AOC for billing? Thanks a lot Stefano Corsi -- Stefano

Hi, I know this question doesn't exactly relate to the core of this list but I thought it does relate to its "hacker spirit". Is it possible to install a Cisco Call Manager 5.X on a non-Cisco appliance ? A friend of mine working for a Cisco VAR told me his colleagues couldn't make it, even for testing purpose. Do you agree ? Regards -------------- next part -------------- An

hello I want to install Asterisk just to use in my LAN, without a analog or digital devices. I need to install all this packages??? Asterisk 1.2.17 Zaptel 1.2.16 Libpri 1.2.4 Addons 1.2.5 Sounds 1.2.1 thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070328/86e4ad95/attachment.htm

I was wondering if anyone knows how accurate the values are when you do a "sip show peers" from the CLI. My configuration is: Asterisk box (192.168.1.102) -> gigabit switch <- PC running x-lite (192.168.1.100) the CLI reports 101 ms delay however, ping is showing <1ms delay Where is the extra 100ms coming from? The softphone response? Here is a dump of some data: CLI>

Hi, I have declared my sip users call-limit=2 and type=friend. When any user recieves a waiting call while already in a conversation, the peer call counter is set to 2.The problem is that, the counter is not reset to zero after hangup and becoz of this the user is not able to recieve any call anymore even if s/he has hungup. the asterisk cli displays the following error. [Mar 17 16:15:10]

Hey everyone, I came across a situation where I needed to use CDP to advertise a voice vlan to Polycom/Cisco (and other CDP capable phones) without a Cisco switch. Sure I have Cisco switches in places but I like my Polycoms to work out of the box and it isn't always practical to purchase a Cisco switch for every location. cdp-tools homepage: http://gpl.internetconnection.net/ So I

There is hotel application weary popular in Croatia - Micros-Fidelio. Now I need to connect Asterisk with this application for purpose of billing. Thing is that hotel would like to give customer one bill for every service that he used while he was in hotel. Has anybody connected Asterisk with Micros-Fidelio? As I understand this isn't some local developed application, it's something

Hi, I'm setting up a Postfic and Dovecot with LDAP email server. My users in LDAP is like this: dn: uid=firstname,ou=People,dc=domain,dc=com uid: firstname uidNumber: 4025 gidNumber: 4025 givenName: firstname objectClass: top objectClass: person objectClass: posixAccount objectClass: shadowAccount objectClass: organizationalPerson objectClass: