similar to: Re: no audio after many transfers

For years we've had the following simple context for outgoing calls: [outtrunk] ; match any NANP, and strip leading 1 off exten => _1XXXXXXXXXX,1,StripMSD,1 ; dial outbound on trunk group 1 exten => _XXXXXXXXXX,2,Dial,Zap/g1/${EXTEN} But when I upgraded on Friday to the latest CVSHEAD, this no longer works. If I send 13115552368 to this context, I get a message like pbx.c: Channel

Hi. I'm using /var/spool/asterisk/outgoing files to place automatic calls, but I'm having trouble setting the Caller ID for the second half of the call. In other words, when we call the first number, we want the Caller ID set to our number, but then when we connect them to the second number, we want _their_ number to be the Caller ID. I've tried the following (and various

I'm evaluating using Polycom phones for our call center and I've set up my first phone (a SoundPoint 560) to give it a try. The phone is working and can successfully place and receive calls. But every minute, there's an error in the log file: chan_sip.c: Registration from '<sip:6193644850 at jtsd05>' failed for '192.168.200.99' - Username/auth name

Hi folks, We successfully set up our ccrb app, running on a server in-house. As a first pass, I was originally running it through a terminal connected to the server via ssh, where I''d just navigated to the app''s folder then run ''cruise start''. At that time, I could connect fine to the CCTray service. We''ve subsequently

Hi all. I've been trying to build some new systems, and no matter what I do, if I load the zaptel and tor2 drivers, the system panics within an hour, even with no traffic. These systems are using dual Athlon MP 2800 chips with one, two, or three T400P boards and 2 GB of system memory. I'm currently using Fedora Core 1, but I also went back to our old reliable Red Hat 7.3 and the systems

I've been testing both T400P and TE405P boards and I'm running into some kind of hard limit on the number of simultaneous calls. This is on x86 with 2 Athlon MP 2800+ CPUs running Fedora Core 1. Everything is fine up to 190 channels, but the 191st call fails every time with errors like: Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on Zap/201-1 Dec 14 15:44:00 WARNING[1215]: Failed

I have a directory dumps on my laptop containing several dumps of various levels. local-0-2007-03-03.gz local-4-2007-02-12.gz local-4-2007-02-19.gz local-4-2007-02-26.gz local-4-2007-03-05.gz local-5-2007-03-04.gz local-5-2007-03-06.gz local-5-2007-03-07.gz local-5-2007-03-08.gz Naturally the level-0 is the largest and rarely changes. On the target (access.cims.nyu.edu) I have a directory

I'm trying to filter calls that don't have a proper ANI. This is what I did: ; only if they a real-looking ANI exten=_1XXXXXX1118/_.N.,1,Newt,1118-config ; Otherwise, send them to the loser partyline exten=_1XXXXXX1118,1,Goto(outtrunk,19096611234,1) This properly deals with null ANIs, but for some reason those with ten zeroes get matched by the first line. I also tried to be a bit more

Zitat von Israel Gottlieb <isrlgb at gmail.com>: > If you the c option in the dial command it will send answered > else where sip message to the phone and most ip phones understand that > The cell will always display a missed call? I'm very sorry, but I can't understand what you mean... Could you explain, maybe with an example? Thanks Luca Bertoncello (lucabert at

Eli Gottlieb <eligottlieb at gmail.com> writes: > I compiled and installed it to the prefix /usr, but that's not the > issue. Once I actually compile and install LLVM with CMake by hand, I > get the share/llvm/cmake stuff installed correctly (can those files be > included in "normal" builds, or will LLVM switch to CMake as its > primary build system?). Now

I am trying to configure a channel bank with 24 ports of FXS., but appear to be hitting a roadblock? This worked on v1.4.xx but now just get "SimpleSwitch" and immediate=no/yes don't seem to make a difference?, no matter if under top section, under channel, etc. Chan_dahdi.conf: [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes

Eli Gottlieb <eligottlieb at gmail.com> writes: > So you're saying that the default CMake build of LLVM creates static > libraries that got linked into my shared-object and now require me to > link in everything they require myself? Shouldn't the linker be able > to track down C++ runtime for this? You told CMake to manage your shared library as if it were a pure C

Zitat von Israel Gottlieb <isrlgb at gmail.com>: Shalom, Israel! > Using chan_sip you need to create another ?user aand then dial both > > Using pjsip you can connect 2 devices Thank you. Unfortunately it seems that I don't have pjsip available as package on the OpenWRT where I installed Asterisk... :( I'll create another user. Thanks Luca Bertoncello (lucabert at

Israel Gottlieb <isrlgb at gmail.com> schrieb: > At the end of the Command you could use options one of them is the c (not > apital) which sends a cancel event to the phone > http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Shalom Israel, unfortunately it does not work as expected... I wrote: exten => _00493512222222,n,Dial(SIP/00493512222222&SIP/00493511111111,,Rc)

Hey I need another pair of eyes on this! I would like to add phones numbers to the blacklist from any handset so I did this: exten => _*66XXXXXXXXXX,1,StripMSD,3 exten => _XXXXXXXXXX,2,DBPut,blacklist/BYEXTENSION/1 exten => _XXXXXXXXXX,3,Hangup However what I get in the database is: /blacklist/BYEXTENSION : 1 And BYEXTENSION is not replaced with the actual number

It would be prudent the test for success and continue rather than failure and drop. For example: exten => s,5,GotoIf($["${SYSTEMSTATUS}" != "SUCCESS"]?105:6) That way only the result that you know is good, Will continue a call.. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On

After I actually get everything compiling, install the library, and load it from my Java program, I get the following: > Exception in thread "main" java.lang.UnsatisfiedLinkError: > /usr/lib/libjllvm.so: /usr/lib/libjllvm.so: undefined symbol: > _ZTVN10__cxxabiv120__si_class_type_infoE If I have to guess, this means that the CMake stuff given is linking to the C++ libraries

Hello-- I'd like to do a little processing on external phone numbers from within the asterisk pbx. Fairly simple stuff, but... devilishly hard to make it work so far! 1. I'd like to dial 9 to get an outside line. 2. If the number dialed after the 9 is 754XXXX, I'd like it to go thru unmodified. It's the only local number available here. 3. I'd like all 1 XXX XXX XXXX numbers

Hi folks, The obligatory newbie disclaimer: "Hi, I'm new to Asterisk and I have a couple questions..." OK, now that that's over with: I've just started working for a small CLEC, and I'm trying to sell * to my boss as a replacement for some expensive/inflexible/closed-source software he's been using to provide residential dialtone with for a couple years now.

Does * support ISDN Primary Rate Interface (PRI) - 2B Transfer Capability for T-1/PRI? In other words the ability to take a call and join it to another call and then drop off letting the CO-switch take over. -Kevin Kevin Fjelsted, President AltiCom CTI, Inc. Track Me Down! One number Access, Press 11# during the voice mail message greeting to have me F-O-U-N-D! Phone: 612.259.0722 Fax: