search for: deflection

Hi, I've recently reinstalled a Diva in my asterisk server (alongside a QuadBRI :-) to test the nice features Armin has been adding in chan_capi. The capi.conf format has changed, so my question is how do I define a deflect= statement for different incoming MSN's? I've tried to define a section for each (group of) MSN with a different "deflect". Is that correct? [DIVA1]

...ROR[98310]: chan_capi.c:1953 capi_handle_msg: received a call waiting CONNECT_IND Nov 16 13:29:05 ERROR[98310]: chan_capi.c:1953 capi_handle_msg: received a call waiting CONNECT_IND Nov 16 13:29:09 ERROR[98310]: chan_capi.c:1953 capi_handle_msg: received a call waiting CONNECT_IND Nothing else. No deflection. No signs of doing anything. What else can i do? Thanks Jens Asterisk 1.0.2-BRIstuffed-0.2.0-RC2 chan_capi-0.3.5 ISDN BRI / Germany

Hello, AFAIK the feature CD (call deflection) is only possible on point-to-multipoint links, is this correct? I've heard about the feature "partial rerouting" which should do the same on point-to-point-links. Is this implemented in either bristuff or chan-capi(-cm)? Thanks in advance, Karsten

I'm using a Asterisk 1.6.1.6 with dahdi. We need to redirect phone calls to a certain number when there is nobody. So I read about call reflection but the call reflection applications on bristuff are not for 1.6.1.6. Are there any other applications or patches that provides call reflection for Asterisk 1.6.1.6?? Greetz TM

Hi list, I am trying to compile and install flac 1.2. I $ ./configure(d) and $ make(d) without any errors or warnings. However I get the following error when $ make check(ing): > Testing 24-bit full-scale deflection streams... > fsd24-01 (--channels=1 --bps=24 -0 -l 16 --lax -m -e -p): encode...ERROR > during encode of fsd24-01 > FAIL: ./test_streams.sh > =================== > 2 of 7 tests failed > =================== Using KUbuntu 7.04 x86_64 using following command: $ ./configure &&am...

Hello List! We're having trouble making call deflection on ISDN PRI. We would like to transfer a call to an external extension but keeping the callerid of the caller so it can be presented to the receiver of the transferred call. At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium hardware TE420B. We've ordered the service (CD) f...

...sandersen+xiphlists@runbox.com> wrote: > Hi list, > > I am trying to compile and install flac 1.2. I $ ./configure(d) and $ > > make(d) without any errors or warnings. However I get the following > error > when $ make check(ing): > > > Testing 24-bit full-scale deflection streams... > > fsd24-01 (--channels=1 --bps=24 -0 -l 16 --lax -m -e -p): > encode...ERROR > > during encode of fsd24-01 > > FAIL: ./test_streams.sh > > =================== > > 2 of 7 tests failed > > =================== > > Using KUbuntu 7.04 x86_64 us...

...: > > Hi list, > > > > I am trying to compile and install flac 1.2. I $ ./configure(d) and $ > > > > make(d) without any errors or warnings. However I get the following > > error > > > > when $ make check(ing): > > > Testing 24-bit full-scale deflection streams... > > > fsd24-01 (--channels=1 --bps=24 -0 -l 16 --lax -m -e -p): > > > > encode...ERROR > > > > > during encode of fsd24-01 > > > FAIL: ./test_streams.sh > > > =================== > > > 2 of 7 tests failed > > > =====...

Hi! Anyone got a patch for call deflection for Zaptel/libpri drivers? Thanks! /hanna -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080415/c69ecdd2/attachment.htm

Hi all, i do have the following setup ISDN BRI Line -> openVOX Card/Asterisk 1.6.2.6/libpri 1.4.11.2 -> Dialplan Dial DAHDI -> ISDN PBX -> ISDN Equipment The user on the ISDN Equipment das enable call forwarding - Teilrufumleitung / Call deflection - so that call will get forwarded by the telco switch - and not using b channels. The forwarding request is coming in on asterisk (i can see it with pri debugging enabled) - but it seems the asterisk does not handle it - does not forward it.... I do have transfer = yes, and facilityenabled=yes se...

Hi, Has someone successfully deflected calls using a Digium BRI board enabled asterisk ? How would you describe your experience ? Which Asterisk, Libpri and Dahdi versions are required for this ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111011/6e746640/attachment.htm>

Hi The only way to have CD service is using: DAHDISendCallreroutingFacility(<destination-5551212>, <original-my-number>, cfu|cfb|cfnr|unknown) or it is possible also with Dial() command before answering the call also ? Regards Babak -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list, I am searching for a possibility to do a certain call transfer method which is called "path replacement" in QSIG. But I want to do that in DSS1 (EuroISDN). If my asterisk does a call transfer I want the machine to signalize on dchan that the call path has to be replaced to a direct connect between the caller and the called, i.e. my machine is to hang up after the transfer and

Hi all, I am trying to get Capi Dial to use a specific outgoing msn. I can't get it to work. If I make a test call to 0703241494 (same isdn line, just one of the other numbers) I don't get CLID at all. Any ideas? ; use 0703241432 as outgoing msn exten => _070.,1,Dial(CAPI/@0703241432:${EXTEN}|30|r) in capi.conf I have: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8

Has anyone a working /etc/asterisk/capi.conf example for Germany or Switzerland using the AVM C4 - ISDN Card. I try to connect asterisk to 3 wires BRI-ISDN (Swisscom). I appreciate your help and it would save me a lot of time, figuring it out by myself. regards, Claudio

...re I actually Dial() it. For instance, if an absent user entered a forwarding number in his (Cisco or Polycom) phone, it will anwser a Dial() with a REDIRECT and asterisk will comply if the context allows. However I'd like to intercept that REDIRECT, get the number and use the telco's call deflection service to deflect the call without answering it. That way, once the deflection is done, no telco lines are used. Whereas with asterisk's default behavior, two lines would be used: one for the incoming call and one for the forwarded call. I've looked in the docs without finding, for a way...

Hi, when I still had ISDN, I was using Hangup(causecode) to send e.g. "Wrong number" to unwelcome callers. Meanwhile, I am only using SIP providers (no PSTN lines any more) and I would like to do similar, i.e. send specific SIP headers. Besides "wrong number", I would especially like to send 302 temp moved with a specified address to deflect certain calls. Is there any way to

Hello, I've been using * for quite a while using AVM C4 card and a few Grandstream Budgetone 102 hard phones plus couple of HandyTone 286. Echo is a big problem. I am getting used to it, but some users complain. Anyone has experience w/ regards to echo comparing the AVM C4 with two HFC-cards? Before shelling out time and money, maybe someone else already has done so, and could tell me if

I've having a big problem after having upgraded to 1.2.4 and chan_capi-cm 0.6.4 When making outgoing calls I don't seem to have any control over the CLI that is presented to the called party -- it can be any one of the MSNs allocated to the line, allocated on what seems to be a random basis. This is on a BT Business Highway line (which is essentially an ISDN2e line with two built-in

I have configured call pickup, and this works fine. Although there are 2 problems, perhaps anyone would know a solution to this; - When I pickup a call from another set, the *8 code keeps being displayed in my screen (Snom 220). I would like it to show the phonenumber of the person calling me. - When a caller that I've answered through Call-Pickup disconnects, my phone does not close