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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:

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

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 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:

...ody) Just two disatvantages - half-second lag even over 3G (maybe my provider is too slow), and that the battery of my N70 got drained over half-day.. guess i just have to buy second charger for work, but this really rocks :) Regards, Atis > > On Nov 21, 2007 3:28 PM, Ricardo Carvalho <rjcarvalho.lists at gmail.com> wrote: > > Here's one sip softphone for mobiles you can give a try: > > http://www.minisip.org/ > > > > Regards, > > Ricardo Carvalho. > > > > > > _______________________________________________ > > --Bandwidth and Col...

Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem

> From: Ricardo Carvalho <rjcarvalho@reit.up.pt> > Subject: [asterisk-users] How to match wild card inside a GoToIf? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > > How can I match wildcards inside a GoToIf? > > I have something like this, but it does...

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

...time Asterisk picked up to ask the pin. I'm I skipping something in my syntax, or is this some kind of BUG? (I'm using Asterisk version 1.2.17) Regards, ---------------------------------------------- Ricardo Carvalho ITEC / IRICUP / Reitoria UP tel: +351220408108 (Ext: 5219) e-mail/sip: rjcarvalho[at]reit.up.pt ----------------------------------------------

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

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 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

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 there anything special that anyone here has had to do to get an Aastra phone (on the Internet) to talk to Asterisk behind a PIX firewall? Ports 10000-20000 UDP are open on the PIX and forwarding to the Asterisk server. The Asterisk server's RTP.CONF is set to use 10000-20000. The phone registers, and will place AND receive calls, however, no audio is passed. The phone is an Aastra

Dear all, My architecture is having some problems with redirects. In the following diagram is shown a simple erroneous test. When someone dials from the PSTN, signalling of the incoming call is passed to Asterisk which routes to SIP Express Route (Ser), and then Ser routes to the phone. The user has configured the phone to forward all calls to another PSTN number, and then, a "302 Moved

Is there a way to make Asterisk don't send "482 Loop Detected" error messages and continue with the transaction that is taking place? Thanks, Ricardo.

Hi all, Is there a way to keep track in Asterisk of which phones are online in realtime using some MySQL DB table for exemple, much like "sip show peers" does in the CLI? Regards, Ricardo.

As seen in the following URL: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions and as I also tested some time ago with an old release of Asterisk, RealTime Extensions didn't support the Ex-Girlfriend syntax. Is it already working in recent 1.4 or 1.2.15 releases? Is there any other way that I can use to do the same thing but only using contexts, for example? If yes, please

Dear all, I've implemented BLF for use with some Grandstream GXP-2000 phones and it works fine in 1.2.x versions of Asterisk, although I tested it with version 1.4.0 and it doesn't work! Has the needed syntax changed for configure BLF for this version of Asterisk? It it a bug of this version? Or should it be misconfiguration that I'm doing? Thanks, Ricardo.

Dear all, In my Asterisk 1.2.17 architecture different levels of permissions are established using different contexts that hierarchically include more permissive contexts until default context is reached. In default context there are only local extensions, only in more restricted contexts there are the PSTN access. So, if some user dials some number, Asterisk looks which context that user