similar to: call deflection support in chan_dahdi, libpri

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

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) from the phone company. The

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>

I did the following: - chan_capi-0.3.5/Makefile: uncommented CFLAGS+=-DDEFLECT_ON_CIRCUITBUSY - recompile asterisk + chan_capi - added /etc/asterisk/capi.conf: deflect=0800123456 ; some 0800 test number - in etc/asterisk/extensions.conf under [tcom-in]: exten => 98765,1,capiCD(0800123456) - made both b channels busy by outcalling on both lines (ISND BRI) - called my msn (98765) by mobile

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

Hi all, i have the following setup PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk 1.6.2.9 -> SIP -> agent Does work quit fine - then agent does have the abibility to transfer a call to a third party - the agent can initiate the transfer over a web interface - it does generate a asterisk manager atxfer request... So agent does initiate transfer - call

--- Daniel Aleksandersen <aleksandersen+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

On 2007-09-15, Josh wrote: > --- Daniel Aleksandersen <aleksandersen+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

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 > ===================

hi all, we do have some troubles with zaptel timing source - we have a setup with 3 telco PRI lines, connected to asterisk digium card 0 - asterisk does some handling - calls are leaving on digium card 1 - going to a siemens hipath - there is some call handling - some of the calls then are going from the hipath over a qsig line to a bosch integral PBX - handling the rest of the calls. To be able

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 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:

The Asterisk Development Team has announced the release of version 1.4.11.2 of libpri. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ This release fixes situation where Q.SIG calling name in FACILITY message was not reported to the upper layer: * pri_facility.c: Q.SIG calling name in FACILITY message not reported to the

The Asterisk Development Team has announced the release of version 1.4.11.2 of libpri. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ This release fixes situation where Q.SIG calling name in FACILITY message was not reported to the upper layer: * pri_facility.c: Q.SIG calling name in FACILITY message not reported to the

Hi, Recently, I made issues after upgrading libpri from 1.4.11.2 to 1.4.11.5. After coming back to 1.4.11.2, everything ran successfully. Unfortunately, I didn't have much time to spend to investigate but I hope I'll soon get an opportunity to try 1.4.11.5 again on another telco line. Basically, the symptoms were that pri show spans kept showing lines to be either down or in alarm. My

The Asterisk Development Team has announced the release of version 1.4.11 of libpri. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ This release contains many fixes and new features, among them being: 1.) Support for NT-PTMP BRI links, including support for multiple TEIs and connecting of BRI phones. 2.) Support for allowing persistent

The Asterisk Development Team has announced the release of version 1.4.11 of libpri. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ This release contains many fixes and new features, among them being: 1.) Support for NT-PTMP BRI links, including support for multiple TEIs and connecting of BRI phones. 2.) Support for allowing persistent

Hi, I'm having trouble capturing calls using the chan_dahdi with dynamic span. Here my settings: chan_dahdi.conf [trunkgroups] [channels] context=default switchtype=national facilityenable=yes rxwink=300 ; Atlas seems to use long (250ms) winks ; where the ring cadence is changed *after* the callerid spill. usecallerid=yes hidecallerid=no

Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein

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]