similar to: Transfer fails

i noticed that when i active the voicemail in the IP-phone where the number 0033149xxxxxx is configured i can call this number without issue Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording SIP/101-0000010d -- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d > 0x2b393cfc2610 -- Probation passed

Hi list, I'm stuck for the moment @ the following : My Query (in a macro) : exten => s,n,MYSQL(Query resultid ${connid} SELECT\ vakantie_set\ vakantie_data1\ vakantie_data2\ FROM\ AstDB\ where\ SIPACCOUNT="${ARG1}") Asterisk CLI : [Aug 30 14:07:42] -- Executing [s at macro-vakantie:2] MYSQL("IAX2/zoiper-9238", "Query resultid 1 SELECT vakantie_set

Hello list, this works : exten => _*XXX*,n,SIPAddHeader("Call-Info:\; answer-after=0") exten => _*XXX*,n,Dial(SIP/${SIPACCOUNT}) The phone auto-answers the call... this does not work : exten => _*XXX*,n,SIPAddHeader("Call-Info:\; answer-after=0") exten => _*XXX*,n,Page(SIP/${SIPACCOUNT}) The phone rings and does not auto-answer the call... Can you tell me

Hello list, I try to pick up a ringing extension but nothing works. To be clear, I'm trying to pick up extension 10. [Jun 14 17:37:34] -- Executing [**10 at from-TESTCORP:4] Pickup("SIP/testcorp3-00000041", "10 at 123456") in new stack [Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No target channel found for 10. [Jun 14 17:37:34] --

Hello list, using asterisk 1.4.30 and trying GROUP() and GROUP_COUNT() for the first time... Having some troubles. This the dialplan (using a sub) : exten => s,n,Set(_custID=${custID}) exten => s,n,GROUP(${custID}) exten => s,n,NoOp(grouppcount = GROUP_COUNT(${custID})) exten => s,n,GoToIf($[ ${GROUP_COUNT(${custID})} > 2 ]?maxreached) The CLI shows : [Jun 5 16:06:26] --

Hello list, Can I do something like this for BLF functionality : [test-blf] exten => _XX,hint,Macro(GetSIPaccount,${EXTEN}) exten => _XX,hint,SIP/${SIPACCOUNT} GetSIPaccount is a macro that looks at a MySQL-database for the realtime table sip_buddies where the SIPusername is taken from. It works great for internal calls... but how about BLF-functionality ?? Feedback is appreciated !

Hello list, what musiconhold class has priority : - field "musiconhold" of the SIPaccount and field "musiconhold" of a queue or - Set(CHANNEL(musicclass)=) ?? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110201/b27a0534/attachment.htm>

---------------------------------------- From: "Joshua Colp" <jcolp at digium.com> Sent: Sunday, October 4, 2015 12:12 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC On 15-10-04 01:09 PM, Bryant Zimmerman wrote: > -- > Joshua > Thanks for your reply. It thought the same thing, but when I

So you are saying that it resolved the issue to activate voicemail on the device that sits past your trunk provider? That confuses me a little, but if your calls are working, that's great news. On Fri, Mar 20, 2015 at 1:44 PM Salaheddine Elharit < salah.elharit200 at gmail.com> wrote: > i noticed that when i active the voicemail in the IP-phone where the > number 0033149xxxxxx is

thanks for your response i noticed that when i active the voicemail in the IP-phone where the number 0033149xxxxxx is configured i can call this number without issue the server asterisk and the ip-phone where the number is configured are in the same network 192.168.1.X Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording

hello list i have an issue related to outbound calls i can contact all the number except on number given by our provider in trunk the issue just when i configure my trunk in our server but when i configure the trunk directly in x-lite i can contact this number without issue below the cli == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0149xxxxxx at

I am having problems transferring calls from one sip extension to another - the extensions use various phones hardware/software. From what I can tell I should just be able to press # and then dial an extension to blind xfer a call right? How do I do attended xfer? Either the phones (for this test I have tried xlite and budgetone102) are not sending DTMF correctly or something else is amiss...

supposed to be able to give you much help with such little info anyway), I can only guess that since you are using the 's' extension, you are in a macro ? If so, try scrolling down the wiki page to the example using '[macro-inbound]'.<br> <br> Rob<br> <br> Jonas Kellens wrote: <blockquote cite="mid:4C17C4A1.8020404 at telenet.be"

On 15-10-04 09:54 AM, Bryant Zimmerman wrote: > I have a pjsip install that is not pulling it's realtime registrations > from an ODBC database. > When I have them directly pull from a MySQL database everything seems to > work. > I am having trouble finding a requirements document for the pjsip > realtime registrations. > Is there some kind of special config that has to be

Hi! > How to do this ?? > To proceed with your answer on PICKUPMARK, where do I put this ??? Look at the example for Asterisk 1.4 on this page: http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup Philipp

The commit 607d432d referred a static int array defined in ctree.h, and a static inline function (btrfs_super_csum_size) using this array, the obvious problem is every c file using that function would have a local copy of that int array, multiple c files calling would result multiple copies of that array: $ nm fs/btrfs/btrfs.ko | grep btrfs_csum_sizes 0000010c r btrfs_csum_sizes 00000114 r

Hi, I am trying to configure busy detect on sip channel but somehow its not working may be this is my mistake could you please help me to figure out. I have added following options in my sip.conf [7527] type=friend context=from-sip host=dynamic dtmfmode=rfc2833 callerid="Guest" <7527> mailbox=7527 at default nat=no qualify=yes cc_agent_policy=generic cc_monitor_policy=generic

Op 20100702 om 11:21 schreef syslinux-bot for H. Peter Anvin: > > lua: Modernize the Makefile <bigsnip/> > -liblua.a: $(LIBLUA_OBJS) > +$(LIBLUA) : $(LIBLUA_OBJS) > + rm -f $@ > $(AR) cq $@ $^ > $(RANLIB) $@ FWIW: Today I also found that missing `rm -f $@` Cheers Geert Stappers

Hi all, I am currently implementing the features I announced in May and I encounter a strange behavior: Assuming we got three nodes, A, B and C. Nodes B and C both connect to node A. A notifies both nodes about each other's UDP endpoints and B & C start communicating directly (not via A). So far so good. If node A goes down for some reason, node B and C cannot send meta

Hello everybody! For my hobby, I have a tabletop mill driven by a linux box (Ubuntu+EMC2). I bought a CAD/CAM package (Vector http://www.vectorcadcam.com ) that works very well; but which runs on windows. Recently, I installed Wine on Ubuntu and tried Vector by copying the whole install directory. To my great surprise :-), it worked OK, except when starting the code generation window, which