similar to: Access sip.conf's mailbox from dialplan ? [SOLVED]

Hello, In sip.conf, each peer/friend/user entry gathers several parameters such as type, canreinvite or mailbox. How can you specifically access to mailbox value from dialplan ? I know how to access custom parameters (ie setvar=FOO=value) but I don't know to access standard parameters. I'm specifically concerned to access to mailbox's value (from a given entry) but would be

2009/12/4 Olivier <oza-4h07 at myamail.com> > Hello, > > Has someone successfully used this QUEUE_VARIABLES() function (in > 1.6.2-rc7) ? > I tried to use it as I'm using SIPPEER() but without success. > > A previous question about it remainded unanswered ( > http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466). > > Regards > How can

Hi! Is it somehow possible to evaluate the SIP response code inside the dialplan? I have an Asterisk server which forwards requests to various PSTN gateways with SIP. If the Dial() attempt is not successful I want to differ at least these 3 options: - called destination is busy (486): e.g. activate auto-redial - called destination does not exist, unassigned number (404) - gateway is broken,

Hi! What are the typical ways to work around the 64 groups limit? thanks klaus

While SVCB/HTTPS provides a better solution for the browsing use case, I see other use cases where ALIAS/ANAME would be ideal, notably in apex RRs. So while fostering SVCB/HTTPS deployment is a good thing, I wouldn?t mind name server software implementing ALIAS. Including NSD, but I reckon it?s much more challenging to do due to NSD architecture than it was to implement it in PowerDNS. But if

Hi! I've went through the READMEs and could not answer this question: During installation, the Setup program asks: Would you like update/upgrade wanpipe drivers? (y/n) For a pure Asterisk TDM installation - is it required to patch the kernel or is this only when using the sangoma cards as WAN router? regards klaus

Klaus Darilion <klaus.darilion at nic.at> wrote: > Yes. 4.8 will only use zone files, the "database" option will be ignored. Pre 4.8 versions can disable the "database" as Anand wrote bevor. Thanks for the clarification. It will be nice to stop the duplication! Cheers, Jamie

Hi Christof! AFAIK, PowerDNS is the only open source name server that supports ALIAS. There was an idea to standardize ALIAS as "ANAME" (https://datatracker.ietf.org/doc/draft-ietf-dnsop-aname/), but the idea was dropped in favor of SVCB/HTTPS record https://datatracker.ietf.org/doc/rfc9460/. So now we have to wait until all Browser vendors implement SVCB/HTTPS. Regards Klaus PS: If

Hi! I am looking for a tool (application or webinterface) which shows me the current status of an Asterisk server, e.g.: - Status of the SIP peers (registered/offline) - current incoming and outgoing calls - start-time, numbers, some history - history (calls stopped in the last 15 minutes, who hang up?) - should be possible to link those calls to the relevant SIP peers -

Hi (Armin?) ! Today I had a problem with Diva Server 4BRI-8M 2.0. Asterisk 1.2.12.1 chan_capi-cm-0.6.5 divas4linux-melware-3.0.f-106.622-1 Asterisk could not receive and make calls on the BRI ports, although the ports looked fine within Asterisk. I usually use "/usr/lib/divas/divactrl dchannel -c 1" to test line activity. This time there was no activity (cryptic log

Hi! I have the following problem with Asterisk 1.4.23: ATA w/ T.38 Asterisk ATA w/o T.38 --------INVITE--------> --------INVITE--------> <-------200OK---------- <-------200OK---------- --------ACK-----------> --------ACK-----------> --------INVITE

Hi! All the AEL examples have a semicolon after the closing curly bracket, e.g: context test { 1 => Hangup(); }; but without ; it works fine too, e.g: context test { 1 => Hangup(); } So - what is the reason for the ; after the closing curly bracket? thanks klaus

Hi! Meanwhile also the location is fixed: it is happening at metalab (http://metalab.at/) - a place for geeks. Thus, we meet there at Thursday, 20.9.2007, 19:00 CEST (=local Vienna time). Metalab is located next to the city hall: http://metalab.at/wiki/Lage Metalab is no pub/restaurant. Thus, don't come hungry! Nevertheless liquid food (drinks) is available. We meet in the library (in

Hi! Since some time the download of the newest Asterisk does not contains the version number anymore, but is just called "asterisk-1.4-current.tar.gz" This gives me a tarball where I do not know the version without looking into the tarball. Thus, IMO it would be very useful to switch back to old schema war the download contained the version number. Thanks Klaus

Hi! I have a setup with Asterisk in front of a PBX connected with ISDN to the PSTN and to the PBX. This Asterisk (a old 1.2 instance) is doing ENUM for outgoing calls and allows incoming calls per SIP. Recently the IP connectivity for this location was down the whole telephony was down too - not even incoming calls did work. This is really strange as incoming calls from PSTN are routed

Hi! I wonder why users.conf generates a SIP user and a SIP peer? Why is it not possible to set type=... in users.conf? (Asterisk 1.4.22) thanks klaus

Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. thanks klaus

I'm having the same "random" problem. I have "canreinvite=no" on all extensions. I have "qualify => yes" on all non-NAT extensions. I do have several NAT extensions, but I've not had reports of problems from those. 95% of my extensions (all polycom 501/601) are on a brand-new network comprised of 2 48-port Cisco 3560 1GB switches. In all cases, the

Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Rgds, ango

hi, i made page for Asterisk T.38 Gateway code testing http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway Final T.38 Gateway API for asterisk 1.8 will be posted by Kevin Fleming later BUT Asterisk 1.8 is too far and we need t.38 gw now if you would like help/test current code(last patch from https://issues.asterisk.org/view.php?id=13405), please contact me i have 2 public testing