similar to: Missing connect indication on pri?

Hi all, When I was making calls from an IP phone, through a X100P, to PSTN, the following error was encountered. -- Executing Dial("SIP/25086937-aa6c", "Zap/1/91713545") in new stack -- Called 1/91713545 -- Zap/1-1 answered SIP/25086937-aa6c Aug 6 00:12:53 WARNING[3983]: chan_zap.c:4717 zt_indicate: Don't know how to set condition 16 on channel Zap/1-1 --

We are wholesale supplier of DJ equipment here in Taiwan.We have various model in stock such as Guitar,Amps/Effects,Bass,Drums,Keyboards,Live Sound in stock.interested buyers should contact us at the below address yangmiaofenggear at yahoo.com yangmiaofenggear at hotmail.com Pioneer DJM 909 ...................$500USD PIoneer DJM 700 ...................$450USD Pioneer DJM 800

I downloaded asterisk-12.4.0 (asterisk-12-current.tar.gz, dated 10-JUL_2014) from http://downloads.asterisk.org/pub/telephony/asterisk/. The packaged configured OK. However, I'm getting a compile error: use of undeclared identifier 'pvt'. I've found similar reports for other versions (asterisk-11 and dongle; macports; a couple others), but nothing for 12.4. Any suggestions to

Hi, I'm looking for a toturial or manpage describing all the thing that you can set with sysctl on RHEL 6 or CentOS 6. It apperas the the default /etc/sysctl.conf coming with the distribution gives a couple of errors on bridgen. Any hints??? Adrian -- Adri P. van Bloois Antonlaan 104 email: adrian at pa0rda.nl 3701 VG Zeist voice: +31-(0)-30-6912741 The Netherlands fax: NONE 52

Morning, On a DC: [root at BPCTASRVSDC003 ~]# samba-tool drs showrepl | grep failure 0 consecutive failure(s). 0 consecutive failure(s). 0 consecutive failure(s). 0 consecutive failure(s). 0 consecutive failure(s). 0 consecutive failure(s). 0 consecutive failure(s). 0 consecutive failure(s). 0 consecutive failure(s). 0 consecutive failure(s). 0 consecutive failure(s). 0

On 06/05/2015 03:48 PM, John R Pierce wrote: > On 6/5/2015 1:44 PM, g wrote: >> for what it would take to do so, you are better off getting buying >> better quality at a little more cost. > > I've had pretty good luck with the basic models of Plantronics > headsets. They tend to be well made. The Logitech stuff I've > bought has often broken within a

Hej! I have the following problem: I would like to do partial correlations on non-parametric data. I checked "pcor" (Computes the partial correlation between two variables given a set of other variables) but I do not know how to change to a Spearman Rank Correlation method [pcor(c("BCDNA","ImProd","A365"),var(PCor))] Here''s a glimpse of

This apparently was a problem that i was not aware since i redid my mail server configuration to make use of Postfixadmin. I have update the conf files using the postfix document fromthe down load. The download document seems to be fpr an early version of dovecot (I am using 2.2.4) as it focuse of dovecot.conf which is now broken into file in the conf.d subdirectory I get "dovecot:

The Asterisk.org development team is pleased to announced the release of Asterisk release candidates 1.6.0.7-rc2, 1.6.1.0-rc3, and beta release 1.6.2.0-beta1. Additionally, new release candidates of Asterisk-Addons 1.6.0.2-rc1 and 1.6.1.0-rc3 have been created. Note that the 1.6.1 series of Asterisk-Addons is compatible with both Asterisk 1.6.1 and 1.6.2 branches. These releases are available for

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.0.7. Asterisk 1.6.0.7 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release resolves an issue where IMAP voicemail message retrieval and Message Waiting Indication (MWI) would not work properly with the same mailbox name in multiple voicemail contexts. This release also fixes a

Hi, I've looked around the archives and have spent a while on voip-info.org but not found an answer so forgive me if this is in a FAQ somewhere. We've got several Asterisk servers with E1 cards (some Digium, some Sangoma). We provide non geographic numbers for customers and route calls to their existing phone numbers. Calls come in over the PSTN and into Asterisk. This works perfectly

What is the best way to authenticate users in SMB4 DC on Linux workstation? I'm using pam_winbind, but sometimes its very slow... -- []'s Jefferson B. Limeira jbl at internexxus.com.br https://br.linkedin.com/in/jlimeira (41) 9928-8628

Hi, I've got a RH72 box called FILESERV acting as a PDC for a while now. It's been doing fine with Win2k clients. The version of the PDC Samba is 2.2.3a-1. I've now added a RH72/samba-2.2.1a-4 machine called GOA to the domain. I've set it to use FILESERV as the password server. On GOA I created a local linux user that's name matches the one in the Windows domain (peter). It

An example of how slow is... [root at CTA1PAPAN001645 ~]# time id teste uid=16777232(teste) gid=16777216(domain users) grupos=16777216(domain users),16777220(operacao),16777222(BUILTIN\users) real 1m15.981s user 0m0.005s sys 0m0.007s According this documentation, if I want use File Sharing without AD modifications only option is Winbind (idmap_rid).

Hi all, After successful joined a new DC (my third), no new DNS entry was created. The end of samba_dnsupdate --fail-immediately --verbose Calling nsupdate for A bp.net 192.168.1.240 (add) Outgoing update query: ;; ->>HEADER<<- opcode: UPDATE, status: NOERROR, id: 0 ;; flags:; ZONE: 0, PREREQ: 0, UPDATE: 0, ADDITIONAL: 0 ;; UPDATE SECTION: bp.net. 900 IN A 192.168.1.240

===================================================================== Asterisk-11.5.1 Centos6 app_confbrige.c ===================================================================== APP: MyConfbridgeCount(Confbridgename,variablename) it will return no of user in conference if conference is created or else zero. Task: Using Dailplan user want to retrive no of user in conference '6050'

Hello, I have problem to make SIP calls, i have asterisk + PC InterP4 + Digium TDM400P here is the content of the sip.conf: [SIP_PROVIDER] type=peer fromuser=testcomclient username=testcomclient secret=testr host=IP_SIP_PROVIDER ;dtmfmode=rfc2833 context=interne canreinvite=no ;allerid=Beer disallow=all allow=ulaw allow=gsm allow=g723.1 ; Asterisk only

It seems consistant after dialing dozens of times that the call that doesn't go through is the one the gets the log message "No echocancellation requested" (chan_zap.c) and the "Scheduleing timer" (channel.c) in the middle of receiving the DTMF tones. I'm now using the T400P card last week very simular problems the the T100P (although I think I was actually loosing

I finally got a 10G card that is recognized by FreeBSD (9.1-stable): ... ix0: <Intel(R) PRO/10GbE PCI-Express Network Driver, Version - 2.4.8> port 0xecc0-0xecdf mem 0xd9e80000-0xd9efffff,0xd9ff8000-0xd9ffbfff irq 40 at device 0.0 on pci4 ix0: Using MSIX interrupts with 9 vectors ix0: RX Descriptors exceed system mbuf max, using default instead! ix0: Ethernet address: 90:e2:ba:29:c0:54

===================================================================== Asterisk-11.5.1 Centos6 app_confbrige.c ===================================================================== APP: MyConfbridgeCount(Confbridgename,variablename) it will return no of user in conference if conference is created or else zero. Task: Using Dailplan user want to retrive no of user in conference '6050'