similar to: Xapian 1.2.19 released

hi, I am installing the asterisk-1.2.19 with zaptel-1.2.8 on FC5. I got install installed ok.. after i had disable the xpp_usb module. However, when i try to compile asterisk and having this error I will be glad for your kind response. Goksie "chan_zap.c: In function ?pri_dchannel?: chan_zap.c:9203: error: ?pri_event_setup_ack? has no member named ?call? make[1]: *** [chan_zap.o] Error

The Asterisk development team is proud to release Asterisk versions 1.2.19 and 1.4.5. There has been a very large number of bugs fixed since the last release, including crashes and other critical issues. There were 244 commits to the 1.4 source tree and 74 commits to the 1.2 source tree since the last set of releases. Also, keep in mind that the release branches are only changed to fix

The Asterisk development team is proud to release Asterisk versions 1.2.19 and 1.4.5. There has been a very large number of bugs fixed since the last release, including crashes and other critical issues. There were 244 commits to the 1.4 source tree and 74 commits to the 1.2 source tree since the last set of releases. Also, keep in mind that the release branches are only changed to fix

The Asterisk.org development team has announced the release of Zaptel versions 1.2.18 and 1.4.4. These releases are maintenance releases that fix various known issues. See the ChangeLog included in the releases for a full list of changes. The ChangeLogs are also available separately on the ftp site. Both releases are available as a tarball as well as a patch against the previous release. They

The Asterisk.org development team has announced the release of Zaptel versions 1.2.18 and 1.4.4. These releases are maintenance releases that fix various known issues. See the ChangeLog included in the releases for a full list of changes. The ChangeLogs are also available separately on the ftp site. Both releases are available as a tarball as well as a patch against the previous release. They

I have a "support call" AGI script that has been working flawlessly for a couple of years now. It dumps the customer into a MeetMe conference room, then dials a bunch of support engineers, and connects anyone who accepts the call into the conference room. The conference room is recorded. After the support call is over, the recording is emailed to a list for quality control, etc. It

On my Asterisk installation, I've had to roll back to Zaptel 1.2.19. When I use 1.2.20, I get very bad echo problems. It seems to work OK if I use a quieter-than-normal speaking voice, but at a sufficient sound level, the echo "breaks through" and then never goes away. The problem goes away if I revert to 1.2.19. Hardware in question: Digium TE110P, Adtran TA750 channel bank,

> Asterisk Project Security Advisory - ASA-2007-013 > > +----------------------------------------------------------------------------------+ > | Product | Asterisk | > |----------------------+-----------------------------------------------------------| > | Summary | IAX2

> Asterisk Project Security Advisory - ASA-2007-013 > > +----------------------------------------------------------------------------------+ > | Product | Asterisk | > |----------------------+-----------------------------------------------------------| > | Summary | IAX2

Hi, I would like to implement a queue using a circular strategy, I mean, using roundrobin or rrmemory strategies. However, I am not able to define the order Asterisk will call the agents once a new call arrives in the queue. Seems that Asterisk will always define its order as the queues.conf file is read, and most of times this order is different from the one I want (for each queue in

Hi, I have two asterisk with the Digium TDM400 installed on the first and the TDM800 installed on the second. Both systems are linux Debian 4.0 whith kernel 2.6.18 and asterisk 1.2.24. Often the cards stop answering calls, and I can't make or receive calls; I need to reboot the system or manually reload the zaptel modules to restore it. I've tried zaptel versions 1.2.18, 1.2.19 and 1.2.20

Hello, I have a problem with MoH at attended transfers. - Mobile A dials into Asterisk - Asterisk dials another Mobile B - Mobile B presses *1 for attended transfer and for example 20 to dial extension 20 - Asterisk sends "Remote hold" message to Mobile A, so the carrier of Mobile A starts playing it's own music-on-hold - Mobile B hang up, so Mobile A should be connected to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I've been using for a long time asterisk-perl-0.08 for prepaid card applications, and I've identified a problem with the last releases of asterisk-1.2, installed with Trixbox. The command get_variable() raises a signal SIGPIPE when it is called (whatever the variable to get). I made tests with Asterisk 1.2.20, 1.2.21 and 1.2.22, and I

Hi all, Perhaps someone can give me a hint i the right direction... Sometimes dtmf is recognized, sometimes not. I'm using 1.2.19 asterisk with misdn for my hfc card. When i got in incoming sip-call, dtmf is recognized, When i phone my self (isdn-phone or gsm-phone) no problem with dtmf When SOME (not all) people phone me (isdn-incoming) DTMF is not recognized. How come? Either it works

On Fri, Mar 04, 2016 at 04:57:30PM +0000, Daniel P. Berrange wrote: > What version of libvirt ? We had an unfortunate implementation mistake > which caused us to waste tonnes of time in memmove(). versions since > 1.2.19 have a fix that addresses it Ah, this is on F23, which has: libvirt-1.2.18.2-2.fc23.x86_64 There's nothing more recent right now in either updates or

Hi, I have a problem with people that are calling from analog lines. We have a block of numbers 12345 - 0 to -99. Most calls are transmitting the whole number including the extension. There's no problem with that. But people calling from analog lines are connected to our asterisk box as soon as they finish dialing 12345. They don't get a chance to dial an extension. Just inserting a

Hi all, I upgraded from asterisk 1.2.23 and zaptel 1.2.19 to asterisk 1.4.18 and zaptel 1.4.12.1 I use polycom 501 phones internally. Everything seems fine. I can pick up the phone and call out, calls coming in work just fine. The issue I see is when the system first calls me, then calls someone else. This works if its polycom to polycom. I hear audio full channel. If I do polycom to external

Hi, I am afraid I have a question to ask of you all. I have just completed setting up a mail server running on Ubuntu Server 10.04. It has postfix, dovecot 1.2.19, LDAP and squirrelmail as the webmail. I have also created virtual users accounts on the system through LDAP. I can send and receive mails which is great. Now, what I am trying to do is to set up a vacation auto-reply on the

Hi all, i am having a major asterisk problem. I think it started around 1.2.19 but could also be 1.2.18 zaptel or 6.5.10 snom 360. basically we start getting busy signals, all our 4 line hunt group is busy, i then check the channels and there are open calls that were hung up long ago. i thought it was a zap problem but then i saw the same problem with iax2 calls. its becoming a huge issue

Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 when i ztcfg -vvv im having this error message and the E1 is not getting up. "cas