similar to: Wrong call information on B leg

Hi, We encounter a problem with the variable CALLERID(dnid) We use E1 lines where we can make an inbound call or an outbound call on the same channel (not at the same time) If the CALLERID(dnid) is not used, than the CALLERID(dnid) will be the CALLERID(dnid) of the previous call For example: - First we get a inbound call on channel DAHDI/11-1 with CALLERID(dnid) = '655871460' We read

Hi all, I?m using * with a Suse 8.1 (kernel 2.4.19-4GB). My hardware are a Eicon Diva 2.01 PCI and a Eicon Diva Server 4BRI; I have installed the drivers wich came with the distribution using the distro installer (yast2). I?m experiencing problems with outgoing calls so I?ll try with chan_capi instead of chan_modem. The point is that I?m having problems with the compilation:

Hi Guys I have just installed a call center onto Suse 10. I have managed to do a DBget (astdb) and extract the DNID numbers to play a DNID specific greeting. We have installed Snom 320 and the customer would like us to Send the DNID(nam) to the phone screens so that the agent will be able to answer in the correct language and with the specific customer company name (ie. Agent says

Hi, Since Carl has kindly provided us with fax support for CAPI based cards, we have been using it with much success. Today I have modified app_capiFax so that it now supports a dynamic CSID. The following example uses the DNID created by chan_capi on an AVM Fritz! card. * Receive a fax with CAPI API. * Usage : capiAnswerFax2(path_output_file.SFF|stationID) * * This function can be

hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only

Hi all, In order to use the "open url" function of zoiper (it opens an url based on the asterisk $callerid(dnid)), I need rewriting of the dnid. In my dialplan I have: exten => 1000,3,Set(CALLERID(dnid)=newdnid) exten => 1000,4,Noop(${CALLERID(dnid)}) exten => 1000,5,Queue(test-queue) but the callerid(dnid) shows the extension called (the member of the test-queue) and not

Hi. I've discovered a bug in the Dial() string processing (for Asterisk 13.14.1 at least). According to the documentation in channels/chan_sip.c the Dial() string syntax is: * SIP/devicename * or SIP/username at domain (SIP uri) * or SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port] * or SIP/devicename/extension * or SIP/devicename/extension/IPorHost * or

Question: Naturally there are times when need to I reformat an extension in a context as such: ;Reformat add CC1 exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1) -or- ;Reformat 011 with with +CC exten => _011X. ,1,Goto(+${EXTEN:3},1) It's a helpful trick, BUT there are times when I want to send the call to another context in its original un-reformatted state. Naturally the ${EXTEN}

Hi, I'm using videocalling on asterisk 1.4.10. When I setup the videocall with exten = n,1,h324m_gw(s at video2webanswer), I loose the variable DNID (${CALLERID(dnid)}) Before the videocall is set up, this variable is filled and after this videocall this variable is empty. Also all local variables are empty. If al look at the A-number (${CALLERID(num)} this variable is not empty

Hi, Yet another question, now that I have callerid working correctly, I'm trying to work out how to utilise the different numbers I have. I have a 100 number range allocated to my E1/PRI/OnRamp service. My incoming calls are handled like this: Advertised/published number is an analogue line terminating on a X101P. If the analog line is busy, it has a call diversion to the PRI on a TE405P

Hi i have an asterisk pbx with E1 port connected to another PBX. Im trying to send the DNID/DNIS to the PBX here's my dialplan exten => 8881111111,1,Dial(ZAP/g2) exten => 8881111111,n,Hangup() The PBX just get the number 2 as it's DNIS when i change it to ZAP/1 or ZAP/g1 the PBX get the number 1. What should i add to send the extension number as DNID/DNIS? Thanks! --------------

Dear Asterisk-Users: Question: ======== How do I get asterisk to pass DNID/RDNIS information between asterisk machines using iax2, in a Dial(IAX2...) command ? Setup: ===== I have two asterisk boxes, MASTER and SLAVE. MASTER is running 1.2.0 and SLAVE is running 1.2.1. The main box handles incoming calls on a multiple lines (both via hardware connection to our internal PBX and calls

Hi all! I'm in the process of configuring an Asterisk server here that, based on which number was called, should send calls to different extensions: 913 - 11111 -> ext. 1 913 - 22222 -> ext. 2 913-11111 & 913-22222 being 2 (of the) numbers we have coming in to our system via our VoIP hosting provider. The config used here is based on Asterisk at home, so it includes also the

Hi! my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides. I compiled pjsip with no resample in pjsip. ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the

Hi, I have an Asterisk 1.2.9.1 with chan_misdn 0.3.1-rc23 and a beronet octoBRI on a Debian box I have to set up instead of an old legacy PBX. My problem is I get only the base DNID and not the extensions (the last two digits) in Asterisk but the old PBX got all the DNID number so I think it is the card. Is there anybody experiencing a problem like this? How can I solve it? Any ideas? TIA

Hi all! I'm in the process of configuring an Asterisk server here that, based on which number was called, should send calls to different extensions: 913 - 11111 -> ext. 1 913 - 22222 -> ext. 2 913-11111 & 913-22222 being 2 (of the) numbers we have coming in to our system via our VoIP hosting provider. The config used here is based on Asterisk at home, so it includes also the

I am configuring a test Asterisk server (1.8.9.2) to practice setting a single codec globally, to avoid transcoding as much as possible. Since all of my recordings are in gsm format, I am trying to make the SIP clients use gsm everywhere. I am using Ekiga on Fedora 16 x86_64 for my tests. [root at elx2 asterisk]# cat /etc/asterisk/sip_general_additional.conf

I am trying to set add a SIP Header to a call before adding it to the Queue. The dial plan sends the call to my macro to perform the work. When I use chan_sip, everything works as expected. When I use PSJIP support, it's not adding the SIP header. Looking at the output, I see the macro is called in both cases. In the PJSIP case, the added sip header never is showing up in the asterisk logs

Hello, I have some problems in placing calls through IAX... It does not work :) in the asterisk console I can't see nothing about dialplan enter or so, IAX debbugging seems to be unuseful... this is my configuration: [612] type=friend secret=123456 notransfer=yes disallow=all allow=gsm allow=ulaw allow=alaw context=from-internal host=dynamic requirecalltoken=no I enabled IAX debugging, but

Hi, Can someone help me formulate MySQL Query(s) which will help me extract the following details for a given DID (date range can be excluded for simplicity). Date-Time DNID (I am recording this is `userfield`) CLID time-1 (when call was received) time-2 (when call was answered by agent) time-3 (when call was hung-up) My Call flow is as follows: - Caller dials a DNID - Call enters queue - Call