similar to: AMP and custom application

Hallo, we have started playing with asterisk about one month ago, and we do like very much what we are experiencing. Now we would like to take some step further towards "standardizing" installed modules, functionalities, tools etc. The "wall" we are facing now is: choosing the right tool for * management. We tried AMP, very powerful but incomplete (CAPI is very important to

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

Is there anyone who is currently using Asterisk as a production H323 gateway? And using which combination of asterisk and H323 (chan_h323, chan_oh323?) The main issue is interoperability with other H323 parties (Cisco AS53xx, Nextone, etc). Searching the mailing list it seems that both h323 and oh323 are not so stable, is it only an impression or using h323 is really not so advisable?

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 ;

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

I have an ISDN card, Billion ISDN PCI Card I tried to use the ZAPHFC, I patched the kernel, I did anything (also followed reccomandation on use on Suse Linux Professional 9.2 --my box is) using bristuff last version. In the end I succesfully compile zaphfc, but I am not able to use the card (a lot of problem running zapcfg, a loto of problem starting asterisk saying about wrong anything (from

I have the following problem when calling from outside my asterisk box : ********************* Extension 'my ISDN phone number' in context 'from-pstn' from '' does not exist. Rejecting call on channel 0/1, span 2 ********************* The card is zaphfc configured (group=2), the calls form internal to outside are perfect. If I had a chan_capi card, I shoud add

Does anyone have any input into integrating asterisk with a alcatel 4400 PBX. Acording to what i've found is that Alcatel uses R2 for E1 -- regards Vikram (http://www.vicramresearch.com)

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.

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

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

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

Kapejod is working on a fix for the CDR problem in bristuff. See below ----- Forwarded message from kpj@junghanns.net ----- Resent-From: tzafrir.cohen@xorcom.com Resent-Date: Fri, 10 Feb 2006 13:01:25 +0200 Resent-Message-ID: <20060210110125.GU16880@xorcom.com> Resent-To: tzafrir@cohens.org.il Envelope-to: tzafrir.cohen@xorcom.com Delivery-date: Fri, 10 Feb 2006 05:19:50 -0500 Date: Fri,

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

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

....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