search for: pbxforall

...really poor, doesn't matter what phrase I give it to convert. None of the other 8KHz voices I have ever used were this bad. It doesn't seem good enough system to be used in a commercial system. Is there any better quality text-to-voice engine? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101023/5c56d155/attachment.htm

...eavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101030/7af3ae2e/attachment.htm

Only 100? We had a single server over 300. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Saturday, October 30, 2010 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Under heavy attack My count has reached 100 for the day. The server serves doesn't serve

...erience in this type of monitoring and automization. I'll be using SNMP data from asterisk servers and endpoints. If anybody has any suggestion which SCADA software can fit in such a VoIP solution, your guidance will be highly appreciated. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20101215/23e46bcd/attachment.htm>

...terisk does support NI-1 ans NI-2, but wanted to get your opinion if you have used these protocols on an Asterisk box and if there were any things to consider. If anybody has experience with Primus, it'll be more helpful. Thanks Zeeshan A Zakaria -- www.visionvoip.com www.ilovetovoip.com www.pbxforall.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110121/bc7e691a/attachment.htm>

...I need to do E1 to T1 conversion for a project, and was wondering if there exists a card with both E1 and T1 on it. Or is it possible to use two separate cards in an asterisk box, one for E1 and one for T1? (Please don't mention aculab or adtran) Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101022/826f2c8f/attachment.htm

...in the past, won't deal again.) I talked to Digium and the sales guy said their TE420 card can *supposedly* do it as it can have ports configured as a mix of E1s and T1s. Has anybody used this card for the purpose of T1 to E1 conversion? Regards Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101022/244b8f71/attachment.htm

Hi all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, I'll make a call, ore receive a call, and there will be NO sound. The strange part is that both endpoints are public IP addresses so NAT isn't in play and a sniffer trace reveals that the packets simply aren't being sent. It only seems to happen on a particular trunk. The same phone

Hello everybody, does anybody know if BLF is correctly working in Asterisk 1.4.*? I'm particularly interested in Asterisk 1.4.25. Thanks in advance! Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101029/f6bd1f95/attachment.htm

Is the a CLI command that shows all channels in use at one time? (Whether IAX, SIP, SCCP, etc)? As well, when I "SIP SHOW CHANNELS" I see phones registering showing as channels in use. Is there a way to filter this output? Thanks! MD

Hi, I am facing issue while generating a dial plan for the following case: all caller should be asked a code to enter than All the callers should be connected one extension. also tell me testing scenario : I have pbx setup and currently I have soft phones to use as extension. Currently I have created a dial plan using vdp I tried submitting it here but I don't know how to extract text

Hi I'm sorry for the my trivial quest. I Have asterisk 1.4 with TDM 400 with FXO and FXS, and works fine from several months. Now I want to connect a device to TDMFXS that want a ring frequecy of 25 hz to activate: i am italian, and usually the ring freq is 20 hz. The other time (I have used that device several times with other asterisk installation) I have modified /etc//modprobe.conf and

Hi ? I ?wonder if?anyone could give some light on SIP NAT. I've having a friken headache with SIP NAT 1 way audio. Client - NAT? - NAT - Server Client can hear users from server side but server cant hear client. ? Ive tried every possible settings externip set localip set NAT= yes / route directmedia yes/ no ? Ive check the sip headers in the debug mode and its using the external address in

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

Hi, please forgive me for this (hopefully) simple question. I cannot seem to find an answer or solution while searching around. I want to be able to call in to my server using my cell phone and be able to set call forwarding for my extension and enter a phone number and also be able to call in to that extension and disable the call forwarding. I see I can do this through the ARI web interface

Over the last two weeks, we have had at least two "incidents" where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further).

Hello Everyone, We are running asterisk 1.2.x version in production environment since last 5 year and we have no issue at all, But now time to upgrade. and i heard about 1.8 which has introduce many features. I am wondering should I use asterisk 1.8 in production ? or should I go with 1.4 or 1.6 stable version? I would like if you suggest me which version would be good for production since