search for: a0305292

how does a GotoIf-challenge look like to match e.g. only the first two digits? i want to strip the first two digits from an incoming pstn-call and add a zero instead so when i forward a call to a mobile the called party gets the correct number of the caller. at the moment, incoming calls from the austrian pstn are recognized as e.g. 43650123123 by asterisk, when i forward the call e.g. to a

hello! i'm looking for a feature to play a sound-file containing a text until the called party picks up the phone. i've already tried with the 'special' musiconhold-feature by adding the m-option at the end of DIAL but it is not exactly what i want. the problem with the m-option is that the file is played to a second caller at the same position as it was played to the first

hello! i'm looking for a way to prolonge a pstn-call for 5 seconds before it enters the extensions.conf. this is for testing purposes, all numbers of a ddi should be received by asterisk before the call is walking through the extensions. how can i achive this? i've not seen a feature like this for zapata or zaptel, does anyone have an idea how this could be done? thx christian

Hello! i'm trying to set up transfer without using the respective asterisk-function but with the built-in phone functions. my goal is to have the first callleg billed to the caller and the second callleg to the callee, who is responsible for the forward(and i can't bill a unknown caller anyways) so far it's working without problems, but my cdr's are messed. with the help of the

Hello Folks! in my sip-logs i see that asterisk uses the User-Agent ID "Asterisk PBX": SipClient: Received: 16:34:03.023 --------------------------------- BYE sip:102141@131.130.XXX.XXX:44343;transport=udp SIP/2.0 Max-Forwards: 10 Record-Route: <sip:213.2XX.XXX.XX8;ftag=as2eb3c466;lr=on> Via: SIP/2.0/UDP 213.2XX.XXX.XX8;branch=z9hG4bK539a.47e6e8a7.0 #this is SER Via:

Hi all, I have an asterisk-server (cvs-head from august) connected to a carrier's switch (DMS/Euroisdn) via a te410p, and I am having problems with DDI (standard 'official pstn' number plus extra digits for 'internal' use) Basically, when the entire number (including the extra digits) is dialled via a redial or a programmed key, I see the entire called party number (including

Hello! After upgrading my production machine to 1.2.1(used to be 1.2.0) on friday i experienced strange behaviour yesterday, i received deadlock-avoided-messages and channels refusing to hangup on span1(used for inbound calls), both messages in all cases paired: Jan 9 17:40:01 WARNING[30003] chan_zap.c: Ring requested on channel 0/17 already in use on span 1. Hanging up owner. Jan 9 17:40:01

i'm having this error message when trying to run the agents-feature Sep 2 17:37:40 WARNING[10445]: pbx.c:1645 pbx_extension_helper: No application 'AgentsLogin' for extension (from-internal, 28, 1) while chan_agent.so is beeing loaded i still don't seem to have access to the commands like agentlogin or agentcallbacklogin. my agents.conf and queues.conf are configured correctly

Hello! In my agi-debug i get the following error-message: AGI Rx << Jan 16 14:45:52 WARNING[18299]: asterisk.c:788 ast_set_priority: Unable to set normal priority AGI Tx >> 510 Invalid or unknown command AGI Rx << SET VARIABLE MODCLI 00434345452 the agi i call is a very simple shellscript that simply removes wrong charakters: #!/bin/bash modcli=`echo $1 | sed -e

hello! has this made it into 1.2.3 already: http://bugs.digium.com/view.php?id=6128 ? i'm trying to set a variable that should be used as a dialstring in the dial-command, including parameters seperated with the respective delimiter, e.g. like: exten => 907,1,Set(DESTINATION1=Zap/G1/4989123456789|10|gh) exten => 907,n,Set(DIALSTRING=${DESTINATION1}) exten =>

i would like to carry some oversea pstn-destinations via sip to providers like stanaphone, however, in case of a network-failure or if the provider is not available, i want to fallback to the zap-channels so the call is carried out to the pstn directly. the usual approach would be to check the dialstatus(e.g.NOANSWER). however, asterisk tries >60seconds to reach that peer(even when the ip