similar to: AMP backup-restore problem

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

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

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.

I don't succed in getting callerId on incoming calls on a zap trunk. I am using a zaphfc card When a call is received, one line in asterisk pbx says -- Executing SetCIDName("Zap/32-1", "") in new stack second parameters should be the caller ID, but it is not set The callerid is not hidden at source, so I think that is some kind of setting in zapata.conf I am using

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

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

Hi all, I would like to eliminate about 150 lines in log /var/log/messages) every time a call is placed/received If I type, on the asterisk console, set verbose 0 the lines in the log disappear, but it appears to me too drastic as a method.... The lines shown in the log don't appear (at least to me) very critical: no problems at all are shown. Isn't any way to turn off this debug ? I

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

....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

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

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

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

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

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

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 ;

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

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

Hi all, I just installed a Suse Linux 9.2 with Samba 3.0.0923 I would like to make this new server a member server of my active directory domain I think I configured almost anything correctly: I succesfully joined the domain via LDAP with net ads join, I can browse user and groups via wbinfo -u and wbinfo -g I can browse user and groups via getent passwd and getent group I can also give file

Is there any way to customize VoiceMail ? I would like to customize the message played to callers sent to the voicemail becouse the extension is busy or otherwise unavailable. Is it a way to record a welcome message and use it ? thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it

I am trying to use language=it in asterisk I downloaded the sound package and installed it I added country=it in indications.conf language=it in sip.conf language=it in iax2.conf everything ok in call from sip and from iax The problem arises in outside call, coming trom CAPI Trunk I try language=it in capi.conf: no result: always language=en I found a german forum, and it seems to be a