similar to: Voip-talk?

Hi all My * implementation is working brilliantly with only one small fault left to kill. I'm using IAXTalk from Telappliant for my incoming/outgoing calls to the pstn network; if I set my codec to GSM everything works great - no pauses but quality is a bit poor. If it set the codec to alaw (I think I'm using the correct one - I'm in the UK) I get intermittent pauses on the call.

Does anyone know of a BYOD provider that terminates calls to NCFA numbers (UK 'national rate'). I enjoyed broadvoices unlimited to those numbers, but this is getting silly now, it doesnt work and no answer if after switching to a new provider it will ever work. Can anyone suggest an alternative provider that serves NCFA? -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605

Hi, I have installed the latest version of asterisk(1.4.0-beta3), and built app_rxfax/txfax. I'm using spandsp from here, http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20061207.tar.gz Everything builds ok. I had to manually apply the patch from the site so configure would spot spandsp libraries. However, when I try dialing my virtual fax extension (either from a phone or

Hello folks, i upgraded to the actual CVS head from yesterday (27.5.) but can not get incoming SIP calls from my provider (sipgate). If someone calls my number, my asterisk responds with the following error: May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request: Failed to authenticate user "<CallerID>" <sip:<CallerID>@217.10.66.11>;tag=as38e9693c I

Hi, Anyone used this service, any comments on reliability/support? Thanks John

Any recommendations for a VoIP provider in the UK? I have a few guys in a field office in the UK with SIP phones and a VPN tunnel back to a working Asterisk setup in the US. The Asterisk setup has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US offices, so they can call vendors, customers etc in the US at local rates. I'd like to get the same thing for the UK, so that UK

Hi, Despite much searching, I can't find anything quite like the problem we seem to be experiencing with our recently activated asterisk pbx. Of four people answering several calls per five minutes, I have reports of the occasional call starting with dead air. The user picks up the call from the queue, and they can hear that the line is too quiet. If they start talking, the caller either

I went on a service call yesterday to find an asterisk system with a T100P card on a Qwest PRI and a TDM40B card connected to fax machines. The TDM40B LEDs were not lit, and the system did not respond to keyboard input. However, calls were being processed for the PRI and 7960 phones. I replaced the TDM40B card with a new one, and the system now seems to be ok. But I'm wondering, why

Hello Everybody, I was going thru the C code of Asterisk. Does anybody know how does one go about modifying the C code of Asterisk? Please do reply. Regards, Bharat M. Sarvan EZZI BPO Pvt Ltd., PUNE. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello List. I am setting up asterisk as a central dialplan, voicemail and conference solution, connected to 12 Cisco 1760 Routers running Call Manager Express IOS distributed around the world. This is all done over VPN. These routers all have PSTN access in their respective country. So far all is good, and Asterisks interopability with the Cisco CME using SIP is very good, although

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Ive just installed a TDM4000P with 4 fxos. The zaptel config is fine, zttest comes back with configured. If i call a line when zttest it shows on the display,and then goes when the line drops. In * when a call comes in, it follows my dialplan and answers the call according to the log, but IT DOESN'T actually pick up the call, i.e. it continues ringing. I'm using KS signalling, and

Hi all, Does anyone know of any providers that can offer local numbers based in the UK via IAX or SIP? We're looking at getting a number based in the UK. Thanks! -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

Hello! I've been reading through the archives on this list for the last 8-10 months. There are some reports on success with tftp autoconfiguration with a given cfg.txt format but really vague. Has anybody successfully done this without using GAPS, or has anybody got a correctly formatted cfg.txt file that works (from GAPS). I would be happy to write a script or a java program that

Hi, I am looking for voip termination all over the world especially based on IAX or SIP. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040124/25e16736/attachment.htm

Hi all, I'm running Asterisk since two days, and it's really one of the phatest software available on the net!!! Respect!!! I have connected Asterisk as a call manager for a cisco gatekeeper. Everything works fine internal, but if I want to ring to a PSTN over another call manager, which is connected over ISDN, I get the following output. Has anyone experience in this or can help me?

Have anyone tried to interface BT's Broadband Voice with asterisk? Kannaiyan

I am new to Asterisk so can I start by apologising if this question has been asked and answered already. I'm in the UK using BT for two incoming lines, one on Wildcard TDM400P and the other on Wildcard X100P. I also have a SIP connection to voiptalk.org. Incoming calls via SIP/broadband ring on extensions immediately. However, an incoming call via PSTN is displayed on the CLI as an

hi: Just starting out with *, and I'm planning to heed the advice to start simple and small, but the goal i'm aiming for eventually is: *-based pbx for 10-20 seat small business, based in the UK. Users will have PoE SIP hardphones. So far so good, but two questions, both UK-specific, relating to connection to the outside world (PSTN or VoIP): - are there any UK-based VoIP providers

I'm planning on setting up some remote agents and before doing so, I did some simple PING tests to measure latency. The average latency I got was 250ms. Does anyone have experience in terms of quality of calls when there is such high latency? Can anyone comment? Thanks, Waldo