similar to: Call Transfers in SER + Asterisk architecture

Hi, I write the CDR of my Asterisk 1.2.17 server in MySQL database using cdr_addon_mysql.so. Now I'm trying to write some custom values to userfield column by the SET(CDR(USERFILED)=SOME_TEXT) sintax, but nothing gets writeen in MySQL cdr table!! Why? I'm I skeeping something or what? Taking a look at the URL:

Hi, Can Asterisk write to multiple MySQL databases in different machines, at the same time, as a backup scheme? If it does, where can that be configured? In res_mysql.conf file? Does anyone ever made it? Regards, Ricardo. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is it possible to force the Dial function to skip to the next priority if it doesn't find the server of the called contact within a few seconds? I know I can use: Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]) where I can use some short timeout in the "timeout" option, but if I do so, when some call is well succeeded, it will only ring for that

Dear all, I'm trying to enable Asterisk to work with FAX using T38. I've tried Asterisk 1.2.4 with the available patch found at URL http://bugs.digium.com/view.php?id=5090 and also with the new 1.4 Beta3 that is announced to support it too. With both Asterisk versions, I've sent with success FAXes between two FAX machines each one attached to an ATA interface, both registered in

Hi, I use Asterisk 1.2.17 in my site, and now I'm trying to configure ENUM lookup in my server. When someone calls a number that has multiple ENUM entries, randomly Asterisk seems to fail to return a correct answer, and dial by ENUM fails. I've goggled a bit on this, but didn't get any good conclusion. There is some RFC Compliant ENUM Macro that can be used that is announced

Hi all, Does ENUMLOOKUP can query multiple DNS servers without having to replicate the same code in which the only thing replaced is the server? If I use ENUMLOOKUP(${exten}), Asterisk will parse enum.conf file to find the list of DNS servers in order of preference to be queried, but, I pretend to use something like this: ${ENUMLOOKUP(+${ARG1:2:},sip,c) which does not seam to care about

Is it possible to load phone registration information stored in sipfriends MySQL DB, so that Asterisk "thinks" those phones are already registered? This would be very usefull for a redundant server... Regards, Ricardo Carvalho. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Thanks a lot for the help. But if Asterisk has private IP address and the only way to access it from remote sites is to have vpn connection to the site that asterisk existed (the site has vpn), then how that will happen from the Mobile to be able to run the softphone from the mobile? Any help? Regards Bilal ----------------- I installed out of curiosity today, and guess what? You can do SIP over

Hi, How can I configure my asterisk server so that I can receive incomming calls comming from the same IP from where my server also receives phone registrations? The problem is that since the moment any extension registers at that IP (actually I have a registration proxy running at that IP), asterisk no more accepts calls coming from a SIP trunk I also have at that IP, replying back with

We're going to be setting up Asterisk at our data center, as well as our call center locations via an optical fiber point to point connection. Is it best to have the servers communicate to eachother via SIP using SER, or just use the Asterisk functions? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I manage an SBC which stands between my company server farm and some SIP telco trunks. The system works fine, for inbound and outbound calls. Now I've configured the SBC to also act as a registration proxy, forwarding SIP registrations coming from the Internet to my asterisk servers. It all seems fine, but it doesn't work well, because by the time at least one phone registers through

Dear all, I'm trying to configure Asterisk to be able to ask the caller to enter a given password in order to continue dialplan execution. I've tested this feature using the Authenticate application like this: exten => _X./5219,1,Answer exten => _X./5219,2,Authenticate(1234,a) exten => _X./5219,3,Playback(pin-number-accepted) exten => _X./5219,4,Dial(SIP/${EXTEN},120)

Hi all, I've 2 * tryning to connect each other Server A is already registred on server B But server B never registers in server A I always get this: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ Timestamp: 00018ms SCall: 00004 DCall: 00003 [XXX.XXX.XXX.XX:4569] CAUSE : Registration Refused CAUSE CODE : 29 Any tip? Best regards, Marco Mouta

Hi, I've just read the postings regarding the interworking between * and SER. As these persons seem quite knowledgeable on this, I would like to have their advise on my planned installation: - I have broadband cable access - I plan to install a SIP-aware router - I plan to install a Linux server with Digium analog IF card(s) for connection to my analog line (incoming and outgoing) - I plan

Hi everybody, I am trying to make up call flow diagrams for for a setup which include ser as a sip proxy/registrar and asteriks as a voicemail server. Is my sequence correct?: UA 1 send an invite to SER. SER forwards this invite to UA2. UA2 sends back a sends back a 100 trying and 180 ringing message. SER forwards these. However UA2 doesnt answer the phone,so what happens then?...is there a

Hello, Currenly I've ASterik@Home 1.5 running on DMZ. I can register SJphone there, good audio on 8200 (webmeet me calls) and i also can dial Zapata extensions. When I dial sip phone extensions nothing happens if the client that i'm calling is registred, if the client has voicemail it goes to voicemail. IMPORTANT: I get this error message on my Check Point Firewall: "sip

Hi, We are testing a SIP solution * + ser solution for a large implementation. All the clients are nated. When a client is dialing outside the domain (to a FWD sip account for example) all is perfect ! ;-) But ,when a call is done to a sip account, the client is ringing, then the caller can hear the nated client very well, but the nated client does'nt hear anything. RTP issue no ? I've

Hi all, I've the current scenario: User "A" - Zaptel call incoming in my Asterisk to my SIP user "B". "B" gets the Call. "A" says : "B" i would like to call PSTN user "C" "B" places a call to user "C" and asks if "C" wants the call from "A". "C" says yes i want, then B needs to

Boa tarde, Ap?s alguma experi?ncia com o Asterisk, e com muito ainda para aprender, gostaria de saber se h? algu?m nesta mailing list que pretenda criar um Asterisk Users Group para Portugal. Visto que acaba sempre por ser uma enorme aprendizagem ( valor acrescentado) a partilha de experi?ncias/problemas e solu??es nas implementa??es Asterisk. H? spre detalhes que variam entre os Telco's de

Waldo, How do you let your customers manage 'their' PBX. I too have a setup like you. However, I installed a * server for each customer, via vserver. I'd like to now what kind of software/webbased package you use for this. I also have SER installed as a front-end server for the * servers. But, as I'm still not very into SER, don't know exactly how this fits in. Should I use