similar to: php agi configuration issue

Displaying 20 results from an estimated 2000 matches similar to: "php agi configuration issue"

2006 Mar 01
3
about operator
I would like to know which kind of solutions are available, both software and hardware, for human operator in an asterisk environment. The operator should be able to provide the basic standard operation, like to transfer calls and to see if the extensions are busy or not and so on. Thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.
2005 Sep 23
2
Problems with queue and remote agents
I all. I have configured a pair of * servers, sip connected each other Mi problem is the following If on the first * i configure a queue containing phone number of the second * (i.e with a round robin strategy) I have non problem as far as all phones are online. If one of the remote phone number is unavailable, when the round-robin strategy touch that phone the call is answered by the voicemail
2005 Oct 12
2
asterisk log
Is there a way to 1) disable asterisk from writing in the "full" log ? ( /var/log/asterisk/full ) or 2) implement a log rotation or similar of the full log ? I see the full log grows a lot (about 100 MB per Month) thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it
2006 Oct 18
2
gotoiftime and Macro question
Is there a way to run a macro in a GotoIfTime statement ?? from the wiki documentation it seems not, but...... I would like to do something like this: ......... 554,3,GotoIfTime(08:30-14:30|mon-wed|*|*?Macro(exten-vm,novm,567)) it does not work, as expected from documentation any workaround to call an extension WITHOUT vm (also if vm for that extension is present...) as a consequence of a Time
2006 Mar 13
1
misdn
Hi all, I just arrived in Italy from Cebit, qhere I spoke with digium and Beronet people. They told me to try to use the mISDN stack to drive beronet and the new upcoming digium ISDN Cards. SO I searched, find http://www.beronet.com/download/card_installation_guide.pdf, and I immediately got the error: asterisk01:~ # cd /usr/src/install-misdn/ asterisk01:/usr/src/install-misdn # make install
2006 Jun 09
2
who is the mantainer ....
....of chan_misdn ? I found a bug, and I don't know where to report it. Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it
2006 Dec 07
2
oh323.conf question
Hi all, I would like to know if it exists the possibility to send to different context according to the caller IP Addres I receive H323 calls, and I have to route this to different devices according to the caller ip. I tried to use the context=first-context alias=999999 context=second-context alias=888888 but I was not able to succed in this; Moreover, I think the keyword alias is related to
2007 Jan 17
1
dtmf problem -- second part
I realize I cannot use inband audio for phones (voicemail and internal ivr, password for external trunks and other thing not working) So I put everywhere rfc2833. Doing this, anyway, make any EXTERNAL IVR NOT working. I see a lot of posts about this, but no solution, becouse using inband audio (which works for outside...) breaks inside IVR Is it possible to define to use inband audio ONLY on
2006 Jan 09
2
dual IP connections
Hi all, I would like to know if there is a solution to this question. Scenario: Two asterisk servers connected across the Intenet ( in SIP or IAX mode, no matter) with both of them having static ip addresses Then I add a second link (with another provider), with another NIC at both side, and again both of them having static ip addresses. Is there a way to tell asterisk to use both of these
2005 Aug 16
1
problems with eyebeam - video phone
I am trying to connect two Xten eyeBeam Video Phone No problems in voice connecting. I tryed to modify my sip.conf [general] language=it videosupport=yes ; enable Asterisk video support port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=h263 allow=gsm allow=ulaw allow=alaw ; H.263 is our video codec ;
2006 Oct 17
1
how to activate recording (automon)
Hi all, If I activate recording for an extension everything is OK. but If I activate call recording on demand i am non able to start recording In principle I should have to press *1, as indictaed in features.conf (I am using almost last asterisk code, updated 2 days ago from svn, version SVN-branch-1.2-r39379M ) Actually it produce no effect at all I am using FreePBX interface, and I saw
2007 Jul 30
1
iax2 trunk registration with auth rsa
hi all, I am trunking via iax2 2 asterisk serverses if both of them have static ip addresses, I can connect them using no password, password or auth rsa with a pair of keys. If one of them has dynamic ip address and need to register on the other server, I can connect them with no password, but I am not able to do that using keys. The question is: which is the right register syntax to use when
2005 Oct 17
1
fax - conversion problem
I am having a strange problem. On one * box I setup the fax recive, via spandsp -app_rxfax I have no problem here. On a second box I did the same. The resulting PDF appear "corrupt". If I transmit the same fax to both * box, the tiff files received are the same. A deeper analysis shows the only problem is the width and heigth of the document In the first PDF, I see
2005 Jun 06
1
AMP and custom application
Hi, I am trying to define DID Routes via AMP (last version 1.10.008) I succeded in defining single DID route, one per extension, let's say i.e. DID number 0101234567 set destination to extension 567 DID number 0101234555 set destination to extension 555 and so on Now I was trying to define only one route to a custom application DID number 0101234XXX routes to Custom-App
2006 Feb 13
1
Bug in AMP 1.10.010 in sip outbound callerid
If you define a sip peer, wheather or not you put an entry in the field OUTBOUND CID, if you dial an external extension (let's say an extension on another asterisk server, connected via IAX2 connection) the callerid received by the foreign asterisk is device <YOURNUMBER>: i.e device <567> If you take a look at etc/asterisk/sip_additional.conf, you can see under the SIP extension
2006 Mar 15
4
misdn problem
I am trying to use misdn insted of zaphfc to drive two billion isdn cards zaphfc is ok, but the problem with cdr and the fact tha you always have to wait the bristuffed version of asterisk took me to try another way. so I downloaded the misdn installation script from beronet for the last version ( I am using asterisk stable 1.2, so now is 1.2.5) wget
2007 May 17
4
how to define a key to decline incoming call
Hi all. We have Snom phones which do have a defined key in order to drop incoming call WITHOUT answering. Pressing that key, a "SIP/2.0 486 Busy Here" message is sent back. We have other phones (I.E. DECT Siemens C450IP, or ATCOM 320 or other) which DO NOT have any key to do that (or the key does not work, as is with Siemens C450 IP ): you have to answer and immediatly after hangup the
2005 Sep 19
1
problems with PRI
Hi, I configured an asterisk box with 1 Digium Wildcard TE110P T1/E1 Card 0 I setup the jumper in e1 position. my zaptel.conf : defaultzone=it loadzone=it #gestione PRI span=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15 # set this to 1-15,17-31 for E1 dchan=16 # set this to 16 for E1 bchan=17-31 # set this to 1-15,17-31 for E1 # Asterisk starts correctly, I see th 30 channels. Anyway I cannot put
2006 Feb 24
2
S100U and TigerJet
Hi all, this is another post about this problem. I installed from scratch a new Suse Linux 10.0, with latest stable asterisk. Moreover I add the lines to /etc/udev/rules.d/50-udev.rules, in order to let the driver create the /dev/zap....... When I plug into usb port my TigerJet adapter, I see on /var/log/messages Feb 24 14:55:02 srvlnx05 kernel: usb 1-2: new full speed USB device using
2004 Feb 17
5
chan_capi problem
Hi to all I've mada up my mind and i tried to change from i4l to chan_capi, following some councelling from the gurus. I compiled it up, and when i try to load it in modules.conf, i get that wonderful message and Asterisk does not start: [chan_capi.so]Feb 17 09:21:40 WARNING[16384]: loader.c:239 ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_get_group Feb