similar to: Flash Zap trunk from a Sipura

Hi, the topic says it all really. Does the Sipura 3000 detect and report UK clid correctly? thanks Mike

test only. good luck! james.zhu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070807/0fd2b827/attachment.htm

Hello, I'm wondering if anyone else has been observing problems lately with 1.4.18 and higher releases of asterisk not properly recording calls. When using MixMonitor, the resulting file is only a few bytes long. I think this is because asterisk is doing Native bridging even though MixMonitor should block that. Did something change around the release of 1.4.18 that would have changed

I don't suppose anyone might know why I hear ringing transposed over itself when I place a call out via PRI? SIP to SIP is fine SIP to IAX is fine SIP to PRI is always transposed I mean sometimes you don't notice it much because it's lined up right, but other times you'll hear a really long ring (starts sounding normal, then sounds "weird" -- like two rings played at

Hello, We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to the telco, users mainly use snom 320/300 SIP phones. When dialing to an external phone number with custom ringback music, users reported that they could not hear the music but can only hear the standard ring tone generated by the system. Is there any kind of settings need to allow the ringback music pass to the

I've seen lots about presence and Polycom phones recently. I've got one here for evaluation but noticed other phones only seem to appear busy when they initiate a call. If they receive a call, they still show as available. Is this a config problem on my part, or is that as far as presence is working right now? Thanks! Trev

This may not be the right place to ask, but I have an annoying issue with a Sipura/SPA1000-2.0.10(e) ATA device connected to an Asterisk box. (The system is remote to me, so I've only been able to observe this by dialling into a VoIP phone on-site, then run commands on the box remotely!) First of all it's all working fine connected to an Asterisk box and the user can make/take calls

We have a PRI from Telepacific. Asterisk 1.2 and a Sangoma A101 T1 card. Outgoing calls to certain toll-fee (8XX) numbers fail -- we hear ringing but the calls are never answered. All other calls, and most toll-free numbers are not affected. The numbers that are affected are all travel related companies (United Airlines, American Airlines, US Air, Starwood Hotels, etc.) we cannot connect to

It seems to never fail - after 3 to 5 minutes SIP -> IAX calls drop audio on one side. I place a call out through voipjet, and call quality is flawless. However a few minutes later the person who I'm talking to can no longer hear me. I can still hear them. What should I look for to resolve this? Has anyone else had this problem? Using last night's CVS this problem still exists.

Hello, I know not too long ago I saw /something/ _somewhere_ about an adjustment to call parking that allowed blind transfers from SIP phones to park a call and still be able to hear the parking lot stall number. Unfortunately, I have no idea where I saw that (google turned up little, couldn't find it on the list either). I'm using Sipura SPA-2000 adapters and it doesn't seem to

Hi, I am about to buy ip pbx asterisk system but what ip phones do you recommend? Are polycom ip all functional with the ip pbx system??? Be waiting.thanks a lot Marlo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050811/16cc52cd/attachment.htm

Hello. Has anyone run into problems accessing voicemail with the Sipura SPA-2002's? Our SPA-2000's work fine (registers fine, able to make and receive calls properly & also able to access voicemail). We've configured the 2002's exactly the same way. However, with the SPA-2002 we're unable to access the voicemail system (though it does register fine and is able to

In my hunt to track down my echo issues, I tried disabling all echo cancellation, suppression, adaption, on my SPA-2000 (Advanced section of the config, under Line 1/2). Then calling from one local extension to another. (SPA-2000 Line1, to Line2 on the same device) I was pretty shocked with the results, the echo was HORRIBLE! I even tried 3 different analog phones. Now, once I turned the echo

Hi I am setting up a Calling card Plat form I have incoming toll number, the provider charges incoming calls I see some spammers( competetors) keep calling my toll. so iam getting huge invoices how can i identify those kind of spammers and block the callerID for some time any suggestions or example could help me ram -------------- next part -------------- An HTML attachment was scrubbed...

Hi all, This is my first post to the list. I have searched the net far and wide but can't find an answer to this problem. When I have call forward working or use the voicemail from a SIP phone, the first part of the message is always cut off. So instead of hearing "call forward cancelled" I hear "l forward cancelled". Or in voicemail I hear "edian mail"

Not able to compile asterisk,zaptel,libpri in /usr/src -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100208/fa5ff126/attachment.htm

Hi My call flow is T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN Call is placed in reverse direction - from PSTN to T38 Gateway. T38 GW reinvites with T38, and asterisk passes it along to the SIP provider. The SIP provider challenges it and asterisk reponds to the Challenge with INVITE with Auth credentials...however, the Asterisk changes the SDP and replaces the T38

Hello, I have some Aastra 9112i's in production that almost function flawlessly. The problem I'm having is when a caller is put on hold they do not hear hold music. If they are on hold for too long (~ a minute?) they are hung up on. All other phones including Aastra 480i and Sipura/Linksys ATAs all seem to be working fine. Is this a quirk anyone else has experienced? Any

Hello, Has anyone tried the combination of asterisk 1.2.24, zaptel 1.4.5 and HPEC 9.00.003? In particular, with a hardware configuration similar to: Module 0: Installed -- AUTO FXO (FCC mode) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (3 modules) I have two fully independent systems

Hello list, I have several SPA-2000's and 3000's scattered about the Internet (all behind NATs). Because I do not qualify as an ITSP, Sipura will not license their "Sipura Profile Compiler" so that I can have the units remote upgrade, remote re-configure, etc (via TFTP or HTTP). This is extremely annoying. Right now if I have to make a config change to any of these