similar to: DAHDI 2.6 and OPENVOX cards

B.H. Hello, i'm using AMI Originate action (with async=true) to send outgoing calls to a SIP trunk (using asterisk-java library to connect to AMI). The problem is that in case of failed originate, OriginateResponse event is returning only the reason code which is sometimes not sufficient to determine the real cause of failure. Also, there's no way to link between the channel that was

B.H. Hello, all. We have noticed many calls on our PBX get "stuck" - the other end sends BYE, and our side sends ACK but the call remains active (no hangup event on AMI, the call is listed in 'core show channels') and it's impossible to hang up until asterisk is restarted. Asterisk's log shows lots of messages like this: chan_sip.c: Autodestruct on dialog .... with

B.H. Hello! We have several Asterik boxes that are connected to PSTN using PRI cards and they are interconnected using IAX2 trunks so that incoming calls are delivered from PSTN to the servers they belong to. In past we were using asterisk 1.4 on the server that is receiving IAX connections and everything worked as expected. Recently, we have switched to a newer box with asterisk 1.8.22 and

B.H. Hello, all :-) We have a cluster of Asterisk (v. 11.9) servers that host IVR applications. The servers work behind SIP proxy (kamailio) for load balancing. All servers are in 2 processor configuration, 8-10 cores per CPU. When a particular server gets about 500 concurrent calls, the sound quality begins to degrade, the sound plays slowly and with clicks. As far as i understand, it's

B.H. Hello, all. We have a dialer software that runs outgoing telephony campaigns. We have been using it successfully with PRI cards, now we're evaluating it's use also with a SIP trunk. Most of the things run perfectly good without a need to change anything except for dial string, but there's some strange problem with asterisk interpreting SIP result codes. Our software is written

B.H. Hello, all I'm using Asterisk 11.7, connected to PSTN using SIP trunk. I'm looking for a way to get data from INVITE's SDP. Specifically, i would like to get a value of o= for incoming call from PSTN because it contains data about the operator that the call originates from. I have googled for a solution and found this patch:

Hello all- My client has purchased these two OpenVox cards and I'm configuring a system with Asterisk 1.6. In the past I have used bristuff and libpri with older versions of Asterisk, but now I would like to upgrade to Asterisk 1.6. Question, should I be using mISDN or libpri for these cards when they are in the same system, or does DAHDI now support both cards under asterisk 1.6

Hello- I have a system with one D410P and one B200P (both OpenVox). All is well with the D410P, inbound and outbound, and I can initiate calls on the B200P BRI span, but there may be something wrong with my inbound BRI setup: there is no indication of an inbound call when I dial in to it from the PSTN. When I run "pri intense debug" and make a call to the BRI span, I can see a

Hello- After configuring DAHDI and starting asterisk, I get the following message continuously on the Asterisk console: WARNING[2057]: chan_dahdi.c:4158 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! My card is a D410P configured for E1, only the first span is configured, and configuration snippets are as follows: From /etc/dahdi/system.conf:

hello, all of users: sorry, resend it again for clarifying the message. I have implemented cha_ss7 in china. initially, the chan_ss7 can not support the call group. i modify the code. now the problem is that, both sides can hear the ring, but i can not hear the voice from each other. i think the ss7 does not send the voice steam to the destination. in chan_ss7, i added:

Hello, I am trying to get a Wildcard TE110P to work in a relatively modern HP Proliant DL385p Gen8 server. Being a potent 12 core Opteron server I expected no problems. Much to my dismay the dahdi_test results are constantly terrible: # dahdi_test Opened pseudo dahdi interface, measuring accuracy... 89.101% 89.195% 89.142% 88.957% 88.953% 89.115% 89.089% 89.134% 89.066% 89.021% 88.933% 89.044%

We have a customer that has Asterisk 1.4.12.1, Zaptel 1.4.5.1, Asterisk-Addons 1.4.3. running on a Dell Poweredge 1900 server (Dual Core Xeon, 4gb RAM, 500gb Raid 5). Until a month ago they had two TE120P cards and everything was working fine. Since they needed to add a third E1 line we decided to change one of the TE120P cards with a TE210P. After the change we had a couple of crashes (server

Hello, I'm thinking of ordering an OpenVox A400P01 (A400P + 1 PORT FXO Bundle) for use in a old IBM 8364 thin client: http://www.openvox.com.cn/products_detail.php?genre_id=9&id=28 http://silicon-verl.de/home/flo/software/netstation-8364/ Has someone already used this hardware with Asterisk, especially on a small piece of hardware like this, and could offer some feedback? Thank you.

Hi I am looking for a source for an OpenVox card. Has anyone purchased through http://www.voiplink.com or do you normally use another vendor or OpenVox.cn directly? thanks Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090705/6d90b996/attachment.htm

Hello Has someone used the OpenVox A400P01 (ie. a supposedly Digium-compatible A400P board with a single FXO module www.openvox.com.cn/products_detail.php?genre_id=9&id=28) successfully? I've put it in an older PC with a Gigabyte GA-7ZX motherboard, then a more recent PC with an Asrock K8NF4G-SATA2: dmesg returns nothing :-/ Is there something specific that needs to be done in either

Hi, This is tech-support from OpenVox, would you mind to send email to tim.june at openvox.cn for more details about G400P issue? Or contact me via IM below for better communication. Regards, MSN: tim.june at msn.cn Gtalk: tim.june666 at gmail.com Skype: tim.jjune OpenVox Communication Co. Ltd. Quick Support: http://wiki.openvox.cn/index.php/OpenVox_Quick_Support

Hi i need to install a server with this hardware: 1 OpenVox B800P 1 OpenVox A800P01 4 OpenVox FXS-100 FXS100 4 OctWare SoftEcho SOFTECHO Do you suggest 1.2 or 1.4 branch? Is now 1.4 stable ? I've tried 1.4 the last year but i've experienced many problems with misdn drivers. Thanks -- /*************/ nik600 https://sourceforge.net/projects/ccmanager

Hi i've got an openvox a800p01 with 1 FXO and 4 FSX i've done the following: - downloaded zaptel-1.4.7.1 > >> - downloaded the file opvxa1200.c > >> - copied in zaptel-1.4.7.1/ > >> - edited makefile adding opvxa1200 in the modules and the voice > >> opvxa1200.o : zaptel.h wctdm.h > >> - edited zaptel.sysconfig adding MODULES="$MODULES

hello, I have the A400P01 from OpenVox. Is necesary to install all this packages? http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.16.tar.gz http://ftp.digium.com/pub/libpri/releases/libpri-1.2.4.tar.gz or just with asterisk and zaptel is enough. thanks a lot -------------- next part -------------- An HTML attachment was

Hello If someone had the opportunity of trying those two analog cards, how do they compare? Digium's sells for $150 while OpenVox's sells for $95. Thanks.