similar to: Presence Fully Supported?

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

Hello, In my quest to create several proof of concepts for what can be done with Asterisk, I've run into a bit of a problem. I have a pair of SPA-2000's acting as off premise extensions for an analog line. When a call waiting call comes in, the caller id information makes it though the ULAW codec and displays on the caller id box, however asterisk doesn't seem to want to pick

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

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

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

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

Hello, To assist Matt's efforts in bug 3554 (2BCT & CNAM), I'm hoping someone can provide a dump of the setup and related messages from a PBX that supports outgoing Station Name to the CO. As suggested in the bug, I tried to ask my telco for a dump of the setup messages for a client that supports this but was told to contact my vendor as they cannot provide that information.

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.

Hi - Writing to see if someone can suggest whether a problem warrants a bug report. It concerns the use of stepFlexclust in the flexclust package. The problem concerns the size of clusters returned. Versions: R-2.7.0 on Windows XP; RODBC_1.2-3 code snippet: r8 <- stepFlexclust(df,8,nrep=100,FUN=kcca, family=kccaFamily("kmedians")) summary(r8) ## returns cluster sizes of 51, 115,

https://bugs.freedesktop.org/show_bug.cgi?id=66255 Priority: medium Bug ID: 66255 Assignee: nouveau at lists.freedesktop.org Summary: Enabling Xinerama with nouveau driver causes Segmentation fault QA Contact: xorg-team at lists.x.org Severity: critical Classification: Unclassified OS: Linux (All)

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"

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Hello, I have dovecot setup to pull passwords stored in plain from a database. My allowed authentication mechanisms are cram-md5 and digest-md5. Cram works but digest fails on my evolution client with and authentication failed. The server is running dovecot 0.99.14-1 on debian sarge. The user logging in is test at advanced-reality.com. Some potentially relevant portions of dovecot.conf are:

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

I have let's say a reception that is comprised of 2 zap extensions and a mobile phone to dial using ISDN through Capi. I want to have a delay before starting dialing the mobile phone so that it rings only when the call has been unanswered for say 25 seconds. I tried to use Capi/2106994444:ww6935555555 but without any success. There is any way to do it or the code has to be modified ? Thanks

Hi everyone I am trying to get my scanner working under linux and to develop a driver I need to capture the data. I have downloaded wine-0.9.2-mdk.i586.rpm and installed it. The wine developers guide (section 3.1) mentions setting read and write values in the config file. The new version of wine does not have a config file, so I would be grateful if someone would let me know how to set read

Oops I sent the last message back to the sender, not the list.... ---------- Forwarded Message ---------- Subject: Re: [Wine] Fwd: Monitoring Parallel Port Date: Thursday 08 Dec 2005 12:10 From: Trev Jackson <trev@g7pvs.freeserve.co.uk> To: Uwe Bonnes <bon@elektron.ikp.physik.tu-darmstadt.de> On Thursday 08 Dec 2005 09:15, you wrote: > >>>>> "Trev" ==

Hi, I have two asterisk servers with the same version of 1.4.29.1. The first server named it as MYE1. MYE1 is an incoming server that can accept incoming calls from PSTN(ZAP E1). The second server is a pbx functions server and named it as MYPBX(SIP). The sip.conf of MYE1 likes below: [MYPBX] type=peer host=mypbx.abc.com nat=no disallow=all allow=g729 canreinvite=yes qualify=no context=default