Re: [Asterisk-Users] sipura spa2 + asterisk bug ?

Tofik Suleymanov wrote: 1. assume 1-st line is in use 2. after dialing 2-nd line from outside i immediately go to the voicemail announcement (also i immediately go to voicemail if i dial from extension to extension both of which are on the same sipura device) Check what response code appears

Re: [Asterisk-Users] Set caller ID for outgoing PRI calls

On Tuesday 28 March 2006 16:33, Tomislav Vojvodic wrote: > Is that what you were asking? My question is: how can I set specific caller id for outgoing PRI calls? pgpqCyPf2WORz.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Ea

[Asterisk-Users] Re: Agent in multiple queues?

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... > > Just add the agent to both queues, * will take care of the rest. > l. I have tried to put agents in groups and then join groups to specific queue. It doesn't work. I don't know is the problem because one agent can't be in more groups

Re: [Asterisk-Users] AstCC

Il Neofita wrote: Hi, I am wondering if it is possible with astcc to make a second call without hangup and be oblige to re-enter all the codes. Any idea how to do? Thank you Yep, one way is to ask for the account code from the dialplan, save it to a var like CARDNO and pass that to astcc.

[Asterisk-Users] Realtime mapping problem after svn upgrade

hi all.   i upgraded my asterisk today via svn but now my oracle realtime is not longer working it always say: Mar 29 08:10:54 WARNING[3876] config.c: Realtime mapping for 'sippeers' found to engine 'oracle', but the engine is not availableMar 29 08:10:54 NOTICE[3876] chan_sip.c: Registration fr

Re: [Asterisk-Users] Dropped calls

Chris Mason (Lists) wrote: I have been experiencing dropped calls on my iax2 connections between my Asterisk server and my ITSP providers, I use Teliax and Voxee but it seems to happen on both so I don't think it is the provider. I don't see any packet loss at the time so I don't think it is

Re: [Asterisk-Users] Voicemail limit?

> Is there an account limit for voicemail? I have 80+ users in >the voicemail and I can only reach the 70-ieth user. If there is a >limit how can I increase it to hundred for example? I've only seen this with something like Voicemail(123&124&125&126) (i.e. leaving the same voicemail in

Re: [Asterisk-Users] RTP frame size location?

John Todd wrote: Google has given me too many responses, so I'll ask the list: Where in the Asterisk rtp source code can I find the default millisecond frame size? I've looked around for obvious pointers, but it's not clear. I'd like to "force" my Asterisk server to use a certain frame si

Re: [Asterisk-Users] Receptionist Phones

For those of us that only need a small handful of these receptionist phones (for me it is 2), it should not be nearly as much of a problem, correct? For example I only need 2 phones with 60 (well, I can get 54 atm, but would like to expand even more). Assuming everybody picked up their ph

Re: [Asterisk-Users] Dial out .call files File permissions??

Tomislav Vojvodic wrote: If you put 777 on some file, that means that anyone can read/write/execute that file.. I think that file ownership isn't important in that case, but if you want to set 'precise' permissions so that only user 'asterisk' can deal with it.. then type chmod 644 filename (

Re: [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk

Hello, VICIDIAL supports this when using VICIDIAL for inbound and/or outbound calling. Blind monitoring, barging in on the call and hijacking the customer from the agent. MATT--- On 3/28/06, Devraj Mukherjee <[EMAIL PROTECTED]> wrote: > http://www.voip-info.org/wiki-Asterisk+manager+API > > I h

RE: [Asterisk-Users] Transferring calls - BUG0003710

Olle, xxx.187.142.20 is a phone (not sure if it's the caller or callee) xxx.187.142.203 is an OpenSER system, which sits in between the phones and Asterisk. I guess OpenSER must be changing the callid when it forwards SIP messages to Asterisk. It's surprising that no one else seems to have en

Re: [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk

http://www.voip-info.org/wiki-Asterisk+manager+API I have been doing some work with the Asterisk Management API and there is a commadn where you can transfer a call. This is what you may be looking for Not sure, trying to be as helpful as I can On 3/29/06, Steve Totaro <[EMAIL PROTECTED]> wrote:

RE: [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk

I do not think so but it would be a great feature. -Original Message- From: Cory Andrews [mailto:[EMAIL PROTECTED] Sent: Tue 3/28/2006 9:59 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] Call Monitoring / Call Takeo

[Asterisk-Users] Call Monitoring / Call Takeover with Asterisk

Does Asterisk support, in a call center type environment, the ability for a supervisor to monitor a call between a system user and a 3rd party, and allow them to physically take over the call.  For instance if a call center supervisor is randomlay monitoring agent calls, and for some reason

RE: [Asterisk-Users] Voicemail limit?

Hi, I'm using version 1.0.9. In the sip.conf or any configuration file, is there any parameter that specify the size of users? Thanks, Ryan At 03:21 AM 3/28/2006, Watkins, Bradley wrote: What version of Asterisk are you running? The site that I have with that many users is 1.2.0 (plu

RE: [Asterisk-Users] Receptionist Phones

Well the problem with the sidecar is simple. Just try to light all lights three times within one second. If you have 50 keys there is already hell breaking loose. If you cascade side cars and say have 100 LED, this is a real Xmas tree. The CPU drowns in XML notifications. We already had trouble, an

RE: [Asterisk-Users] Receptionist Phones

So how did the Polycom with sidecars work? I like the idea of a dedicated FOP display but not sure why you would need it if you have a Polycom with sidecars. > -Original Message- > From: Jerry Jones [mailto:[EMAIL PROTECTED] > Sent: Tuesday, March 28, 2006 7:28 AM > To: Asterisk Users Ma

Re: [Asterisk-Users] Dialogic d/4 PCI

Rene Nelson wrote: Is this card supported in the Open source version of Asterisk? If so has anyone had success implementing one? Any help will be greatly appreciated. The list of supported hardware is at http://www.asterisk.org/hardware If your card is on the list you will need to call Digiu

[Asterisk-Users] IAX2 errors

Hi, all. I have problems with iax2, when try to communicate with one third server, asterisk reports the following errors in server's, could help me? Server A it speaks It with C in iax and Server B it speaks with D in iax, but Server A it does not obtain to speak with B in iax, reports the followin

[Asterisk-Users] dial plan logic

Just starting to enjoy the full features of asterisk, I do have a couple questions though, that I can't seem to find answers for in the wiki, just wondering if someone could light my way. after a caller has made their choice of options in the dial plan, I would like them to be placed on "hold"

Re: [Asterisk-Users] Psgw

Haven't tried this product myself, but according to their spec it's only 1 call. There's another free SIP-Skype gateway from www.nch.com.au called uplink. http://www.nch.com.au/skypetosip/index.html Giordano Grandis wrote: Hi all, anyone never used PSGW as gateway beeween * and SkyPe? If yes

RE: [Asterisk-Users] RTP frame size location?

Check out bugid 5162 on Mantis. It allowed per peer/user packetization settings. It is in need of much love, but as-is should not be too far from applying to the 1.2.X series. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Tuesday,

[Asterisk-Users] Dialogic d/4 PCI

Is this card supported in the Open source version of Asterisk?  If so has anyone had success implementing one?  Any help will be greatly appreciated.     ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIB

Re: [Asterisk-Users] Transferring calls - BUG0003710

29 mar 2006 kl. 01.03 skrev Douglas Garstang: I made the post below earlier today. I'v since removed all NAT from the equation and the problem still persists. Basically I am trying to transfer a call. The transferring phone sends a REFER message to asterisk with a call id that Asterisk doe

Re: [Asterisk-Users] Softphone accepting URL

Bruno de Assumpção Loureiro a écrit : Does anyone know a softphone that can accept URLs during a call and open that page in the default browser when the call is answered? I Know DIAX and the IDEFISK, only pro version.I need another ones. It can be using the cmd SetURL Regards. May I advertise

RE: [Asterisk-Users] Asterisk 1.2.6, VMWare, & Playback/BackgroundGSM prompts

I think you might have misunderstood my point. From a horsepower perspective you are right. However, from a latency perspective, virtual machines suffer from too much skew in their timeslices. If the server is performing background tasks the impact is unperceivable. If the server is performing r

Re: [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P

Andy Kuo wrote: Hi, We have Rhino channel bank and TE406, and are thinking of doing the same thing. What modem or modem pool are you using? We are dialing out to an ISP (for backup if our broadband connection is down), not providing dial in access. We are doing some dial up modem to dial up

RE: [Asterisk-Users] Asterisk Tools for OSX

Yaah! I'm a Mac fan. PPC mini in my home office and Intel dual core mini in our audio video room. I'm a fan of JackenIAX softphone and look forward to any OS X integration with Asterisk. Thanks and keep us posted. Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED

[Asterisk-Users] H323 Info

Hi,I compiled for my asterisk 1.2.4 the openh323but when I give this commandh.323 show codecsI receive thisAllowed Codecs: Table: Set:I cannot test with msn if everything is working since I am outside and I cannot access to the firewall. Someone can tell me if I need to install the oh323 si

[Asterisk-Users] Asterisk Tools for OSX

Hello Asterisk Users, I am an Objective-C enthusiast and have been writing some clever tools to integrate Asterisk functionality with Mac OS X applications. Please find my project on http://www.sf.net/projects/astrxtools4osx/ The objectives of my project are as follows 1. Implement an Objective

[Asterisk-Users] Transferring calls - BUG0003710

I made the post below earlier today. I'v since removed all NAT from the equation and the problem still persists. Basically I am trying to transfer a call. The transferring phone sends a REFER message to asterisk with a call id that Asterisk doesn't know about. Surely, surely someone else mus

[Asterisk-Users] RTP frame size location?

Google has given me too many responses, so I'll ask the list: Where in the Asterisk rtp source code can I find the default millisecond frame size? I've looked around for obvious pointers, but it's not clear. I'd like to "force" my Asterisk server to use a certain frame size all the time. (

[Asterisk-Users] Can realtime extensions be used within AEL contexts?

Can realtime extensions be used within AEL contexts? I've tried without success. Also did some Googling and looking around on the Wiki, but couldn't find much. Thanks, Mike Clark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Us

[Asterisk-Users] Asterisk to MySQL Data Lookup Warning Message?

Hi All, I'm getting a strange warning message when I perform a MYSQL data lookup. The operation performs fine, I retrive the data I'm looking for and continue on through the dial sequence without an issue. I'm wondering if this warning message is something to be concerned about, can't find an

Re: [Asterisk-Users] RXgain

a te110p. Any thoughts on why? I have not unloaded he modules and reloaded them as it is during the day. Does this even need to be done to take effect; I did restart the asterisk service. just fyi, since taking out the modules and re-inserting them won't take much longer than restarting the aste

Re: [Asterisk-Users] Problems Configuring Cisco 12SP+

Derek, Thank you for your quick response. If i do a skinny debug, nothing happens :( And this shows my CLI: server*CLI> skinny show lines 101 1 101N N server*CLI> skinny show devices Name DeviceId IP

Re: [Asterisk-Users] How to disable event_log?

On 15:49, Tue 28 Mar 06, steve wrote: > the astersisk.conf file has a pointer to where the log files are stored: > astlogdir => /var/log/asterisk > > maybe if you changed it to point to /dev/null/ they would just vanish. I > can't vouch if this would work but its a suggestion. This will disable

[Asterisk-Users] How to disable event_log?

the astersisk.conf file has a pointer to where the log files are stored: astlogdir => /var/log/asterisk maybe if you changed it to point to /dev/null/ they would just vanish. I can't vouch if this would work but its a suggestion. Enjoy, Steve Date: Tue, 28 Mar 2006 00:31:41 +0200 From: Roger

[Asterisk-Users] RXgain

I have really cranked up the rxgain on a t-1 trunk in Zapata.conf. It seems to have no effect although I raised it to 7 from zero. I am using a te110p. Any thoughts on why? I have not unloaded he modules and reloaded them as it is during the day. Does this even need to be done to take effec

Re: [Asterisk-Users] Softphone accepting URL

In a inbound queue() environment, you can also have the same result using QueueMetrics and any soft or hard phone. l. In data Tue, 28 Mar 2006 21:26:13 +0200, Bruno de Assumpção Loureiro"" <[EMAIL PROTECTED]> ha scritto: Does anyone know a softphone that can accept URLs during a call and

Re: [Asterisk-Users] Problems Configuring Cisco 12SP+

Juanjo Portela wrote: > Hi, > > After reading this valuable forum and the voip-info wiki and follow > all the steps , but my Cisco 12SP+ remains unregistered. > > These are my config files: > > skinny.conf > [general] > port = 2000 ; Port to bind to, default tcp/2000 > bindaddr = 17

Re: [Asterisk-Users] queue caveats

On 22:18, Tue 28 Mar 06, lenz wrote: > Yes - I think so at least, this is surely true as of * 1.2.4, and have > seen nothing in the ChangeLog to suggest it has been changed. > By the way, this is usually an annoyance, as Agents would often like to do > an attended transfer, not an unattended on

Re: [Asterisk-Users] Redirect problem/bug/feature

This is just like you if you would be doing this: Dial(Local/[EMAIL PROTECTED],22,t) Which if Local/944 has then: Dial(Local/[EMAIL PROTECTED]) and exten => 904,1 has: Dial (Sip/904,,90,t) Then asterisk will dial sip/904 for 22 seconds. and whatever follows the first dial (Dial(Local/[EMAIL PROTEC

Re: [Asterisk-Users] AAH Mailing list

Dan Elder wrote: any pointers to where this list is? I dont see it on the sourceforge pages. It appears that they've moved to a forum only format, judging by the number of messages posted on the Sourceforge site. Doug ___ --Bandwidth and Colo

Re: [Asterisk-Users] Polycom IP 301 is slow

I've read all the messages here.. seems everyone's forgotten the original problem. Yes, the polycoms are very slow at rebooting when configuring through its http server, not only that, but it's also inconvenient because it reboots for every change (each page) which means you have to pass throu

Re: [Asterisk-Users] Monitoring question

Hello Bjorn, this will very likely be the solution to your problem: http://www.oinko.net/astrecipes/index.php?n=119 Then you can use QueueMetrics - http://queuemetrics.loway.it - or something similar to listen to both pieces of the call through the web interface. Hope this helps, l. In da

Re: [Asterisk-Users] Softphone accepting URL

Bruno de Assumpção Loureiro wrote: > Does anyone know a softphone that can accept URLs during a call and > open that page in the default browser when the call is answered? I > Know DIAX and the IDEFISK, only pro version.I need another ones. IAXPhone from Sokol & Associates can do that as well. ___

Re: [Asterisk-Users] Agent in multiple queues?

Just add the agent to both queues, * will take care of the rest. l. In data Tue, 28 Mar 2006 16:13:13 +0200, Matt <[EMAIL PROTECTED]> ha scritto: Hi, What do I need to do to put an agent into two queues? The idea being that the agent will get the call no matter which queue it comes into?

Re: [Asterisk-Users] queue caveats

Yes - I think so at least, this is surely true as of * 1.2.4, and have seen nothing in the ChangeLog to suggest it has been changed. By the way, this is usually an annoyance, as Agents would often like to do an attended transfer, not an unattended one. Anybody has better solutions to this pro

[Asterisk-Users] Problems Configuring Cisco 12SP+

Hi, After reading this valuable forum and the voip-info wiki and follow all the steps , but my Cisco 12SP+ remains unregistered. These are my config files: skinny.conf [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 172.20.1.1 ; Address to bind to dateForma

Re: [Asterisk-Users] How to send announcement after called has picked up the phone?

Does your phone show callerid? Prefix the callerid with the provider it came from: SetCIDName(SIP1:${CALLERIDNAME}) or SetCIDName(SIP2:${CALLERIDNAME}) or SetCIDName(VB:${CALLERIDNAME}) (maybe for voipbuster) or SetCIDName(SG:${CALLERIDNAME}) (maybe for sipgate) voipbuster and sipgate are just

[Asterisk-Users] Asterisk & SMP: Is irqbalance Redundant on 2.6 Kernels?

Asterisk users, I posted the following email to the Fedora users list and it got no responses, so now I'm calling on your expertise. Please take a look at it and share your knowledge on the subject with me. Additionally,

Re: [Asterisk-Users] How to send announcement after called has picked up the phone?

Why not just set the callerid with added info about the caller? Say, a number before/after the number/name. That way it's immediate, and you know when it's ringing. Aaron On Tue, 28 Mar 2006, Benoit Panizzon wrote: Hi I would like to send a text to the called person when he picks up the p

Re: [Asterisk-Users] How to send announcement after called has picked up the phone?

Use the M option to execute a macro and have the macro play a file, based on the arguments passed in the dial command, or you could use the playfile option from app_dial. show application dial is your friend here. On 3/28/06, Benoit Panizzon <[EMAIL PROTECTED]> wrote: > Hi > > I would like to send

[Asterisk-Users] Redirect problem/bug/feature

I have a major problem with SIP redirects, or maybe just do not understand how they are supposed to work. We are using Cisco 7940s and 7960s with SIP version 6.3. Asterisk 1.2.5. A call come in to extension 944 over the IAX trunk. Extension 944 has forward all to extension 904 set on the phone. Ac

[Asterisk-Users] AAH Mailing list

any pointers to where this list is? I dont see it on the sourceforge pages. Hans Witvliet wrote: > aah-handbook (version 1.6) doesn't spill a single character about bri > and "tfot" doesn't spill much paper of the subject either ;-( > > Any suggestions/pointers > > Hans > You may want to try t

Re: [Asterisk-Users] Echo cancellation

Hmm, that control-enter thing got the better of me and sent the message before I was done. Just wanted to mention that if a call weren't in progress, he would have seen "Zap/1-1 is not a known channel" Moj Steve Davies wrote: On 3/28/06, Giordano Grandis <[EMAIL PROTECTED]> wrote: Hi all,

Re: [Asterisk-Users] Echo cancellation

Seems Giordano only would have gotten a response to his query of the status of Zap/1-1 if a call _were_ in progress, hence the '-1'. Otherwise there would have been: Steve Davies wrote: On 3/28/06, Giordano Grandis <[EMAIL PROTECTED]> wrote: Hi all, I'm using bristuff 0.2.0 RC8o with a HFC pc

[Asterisk-Users] How to send announcement after called has picked up the phone?

Hi I would like to send a text to the called person when he picks up the phone before the call gets connected through. Is there a way to do this? Example: I'm registered to multiple SIP providers. They come in to a context each and then get through to my phone. Now I would like to send myself a

RE: [Asterisk-Users] Flash Operator Panel

Make sure you have port 4445 opened up if you are using any sort of firewall for your asterisk server - like iptables. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Thursday, March 16, 2006 9:16 AM To: Asterisk Users Mailing List - Non-Co

RE: [Asterisk-Users] Asterisk 1.2.6, VMWare, & Playback/Background GSM prompts

For what it is worth I had the same experience with VMWare server. I got better sound from my 5 year old workstation. From: [EMAIL PROTECTED] on behalf of [EMAIL PROTECTED] Sent: Tue 3/28/2006 12:40 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Use

Re: [Asterisk-Users] Asterisk eating CPU

On Tue, 28 Mar 2006 14:07:20 -0500, Script Head <[EMAIL PROTECTED]> wrote: and one the processes utilizes way more CPU than any other. According to htop, it used 7:59:XX of CPU time. Once I kill asterisk and restart, another process does the same thing while others are running smoothly. This d

[Asterisk-Users] Softphone accepting URL

Does anyone know a softphone that can accept URLs during a call and open that page in the default browser when the call is answered? I Know DIAX and the IDEFISK, only pro version.I need another ones. It can be using the cmd SetURL Regards. -- Bruno de Assumpção Loureiro msn: [EMAIL PROTECTED] __

[Asterisk-Users] Asterisk eating CPU

I have asterisk running user a user/group asterisk/asterisk like so su - asterisk safe_asterisk and one the processes utilizes way more CPU than any other. According to htop, it used 7:59:XX of CPU time. Once I kill asterisk and restart, another process does the same thing while others are runnin

[Asterisk-Users] Re: Unable to authenticate password - VM

dtmfmode=auto in sip.conf did it. I guess I need to keep reading, hadn't encountered that instruction yet! Thanks for your help. -Ben >I'm unable to get into voicemail to retrieve messages. ___ --Bandwidth and Colocation provided by Easynews.com --

RE: [Asterisk-Users] Bluetooth headsetin handsfree modewith SJPhoneorX-lite

I didn't change it. I just had a heckofa time getting it to load. Unfortunately, I neglected to document the process, since this was on my home PC and I wasn't planning on replicating it. If I can find some time, I'll see if I can recreate the process. -Original Message- From: [EMAIL PROTE

Re: [Asterisk-Users] Cisco 7970

Hello, can I use same settings and config files with Cisco IP Phone 7910 ? :)On 3/24/06, jason justman <[EMAIL PROTECTED]> wrote: Best bet is to get "Asterisk Chan_Sccp" http://chan-sccp.berlios.de/1.) setup your /etc/asterisk/sccp.conf with something like:[devices]type= 7970   

RE: [Asterisk-Users] Asterisk 1.2.6, VMWare, & Playback/Background GSM prompts

On Tue, 2006-03-28 at 13:08 -0500, Technical Support wrote: > You can't reliably run a real-time application (like asterisk) on a > virtual machine. You will get better performance from an old PC than > a VM on a new top-end PC. Sorry > > MD H, I would have to say a properly configured GSX

RE: [Asterisk-Users] Asterisk 1.2.6, VMWare, & Playback/Background GSM prompts

You can't reliably run a real-time application (like asterisk) on a virtual machine.  You will get better performance from an old PC than a VM on a new top-end PC.  Sorry   MD From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]Sent: Tuesday, March 28, 2006 12:4

Re: [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P

Hi, We have Rhino channel bank and TE406, and are thinking of doing the same thing. What modem or modem pool are you using? Can Asterisk serve as an access server/gateway to the internet? Please share your experience. Thank you. Andy On 3/28/06, Don Pobanz <[EMAIL PROTECTED]> wrote: > Nico Gief

Re: R: [Asterisk-Users] Echo cancellation

Steve Davies wrote: > Use 'zap show channel ...' on the channel where the call is in > progress, and it should show that EC is enabled. Except that currently it does _not_ reflect when the EC has been disabled due to the receipt of the CED tone (sent by a FAX or modem). Basically, it reflects that

Re: [Asterisk-Users] Receptionist Phones

What I read on snom's website was the _currently_ only one sidecar can be hooked up at a time. It sounds like they are working on getting multiple sidecars chained together but have not got all of the bugs worked out. I am kind of in the same boat. Our current system offers 60 buttons on

[Asterisk-Users] Asterisk 1.2.6, VMWare, & Playback/Background GSM prompts

I've spent the past week experimenting with [EMAIL PROTECTED] 2.6, andthen Asterisk 1.2.6 individually, on VMWare Workstation 5.5. I have anentirely IP (hard & soft)phone setup (IAX and SIP) so I have norequirements to support any Digium PCI cards, etc.All in Asterisk works extremely well except f

[Asterisk-Users] WARNINGS For SIP call

I am getting the following warnings on the Asterisk when i try a call parking scenario. I use Ciso 7920 phones and Cisco 2800   Executing ParkedCall("SIP/192.168.50.2-088cde00", "366") in new stack    -- Channel SIP/192.168.50.2-088cde00 connected to parked call 366Mar 28 17:07:36 WARNING[10027]:

[Asterisk-Users] Monitoring question

Hello all!   We’ve been thinking of using the monitor or the mixmonitor application for a while. However, we have met some basic problems in getting this to work as planned:   First of all, most of our calls come in through a call queue. There’s a monitoring option in the queue, and it

Re: [Asterisk-Users] Unable to authenticate password - VM

Ben, The first thing I would check is if DTMF mode is set up correctly for your SIP channel. The Voicemail application is trying to read the user and password as DTMF tones and if it can't, it will take it as empty. That is why there is nothing between the '' marks. dtmfmode = auto in sip.con

R: R: [Asterisk-Users] Echo cancellation

I did it Steve, but on some calls i still have the EC on OFF. What can i check? Could it depend of my zapata.conf ? Thanks Giordano Grandis -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Steve Davies Inviato: martedì 28 marzo 2006 17.08 A: Asteri

Re: [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P

Nico Giefing wrote: Is it possible to establish a ISDN DIAL up Connection and Analog Dial up Connection (V90) trough asterisk with Digium TE405? I do the v90 dial up. The modem is connected to an Adtran 750 channel bank. Our DID trunks are on a T1 line to the phone company. If you have anal

Re: [Asterisk-Users] Unable to authenticate password - VM

looks like a DTMF mode issue. How you calling Asterisk? SIP Phone? make sure dtmfmode in sip.conf is set correctly. On 3/28/06, Ben Gore <[EMAIL PROTECTED]> wrote: > Hello: > > I am relatively new to Asterisk, so this hopefully is an easy question for > someone: > > I'm unable to get into voicemai

RE: [Asterisk-Users] web meetme instructions

Sorry for the late reply, I was away on vacation.   Version 1.2 was created by Areski and I extended it to include the scheduling functions.  I guess I should get an account on the Wiki and make some updates.   If all you need is a tool for monitoring conferences, version 1.2 is the way to

[Asterisk-Users] Psgw

Hi all, anyone never used PSGW as gateway beeween * and SkyPe? If yes, how does it works? How many session could I have on a single user ?   Thanks all   Giordano   ThanksThis e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have rece

[Asterisk-Users] CID passthrough

This may be an issue for the other business list, but does know how I can set a2billing to pass the CID of the calling party, and not my poviders' numbers. My current provider does allow me to pass my own CID data, I just can find in a2billingany ideas?

Re: [Asterisk-Users] sipura spa2 + asterisk bug ?

Vahan Yerkanian wrote: Tofik Suleymanov wrote: Thanks all for replying tommorow i'll try to do like you suggested. One more quick question: why Sipuras cant do more than 1 g.729 channel at a time ? Insufficient CPU power to process 2 g729 streams. Is this somehow related to g.729 licensing

Re: [Asterisk-Users] Snom 360 problems

I may need a consultant that can help with some problems. We had 2 smaller satellite offices on the same asterisk systems with no trouble. We've just upgraded the main office and hit troubles. There's no going back because we've entirely outgrown our old system. Any * consultants around Cin

[Asterisk-Users] Unable to authenticate password - VM

Hello: I am relatively new to Asterisk, so this hopefully is an easy question for someone: I'm unable to get into voicemail to retrieve messages. Since I'm learning, the conf files are very basic. The relevant portions of the voicemail.conf is: [general] format=wav [EMAIL PROTECTED] append=yes m

RE: [Asterisk-Users] NATted phones transferring calls - BUG0003710

> -Original Message- > From: Alexander Lopez [mailto:[EMAIL PROTECTED] > Sent: Tuesday, March 28, 2006 9:04 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] NATted phones transferring calls - > BUG0003710 > > > I wont go into the details of

RE: [Asterisk-Users] Receptionist Phones

Very true. I am currently debating whether or not to offer it as an option for my employer's system. As it currently stands, we do not have everyone's extensions on a button. With the snom 360 plus the expansion we still don't have them all. While I'm sure it would be 'better than nothing' fro

Re: [Asterisk-Users] Problems with wcte11xp module

--- [EMAIL PROTECTED] wrote: > It means that you are loading the digium card up > with incorrect values. > Ok when I modprobe wcte11xp I get the following message ZT_CHANCONFIG failed on channel 26: No such device or address Any ideas? Regards Jon Jon Farmer Telford, Shropshire, UK

RE: [Asterisk-Users] automatic callback when busy

Sorry for delay. I never tried personally but received this recipe from the author of the first one in this wiki page: http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+reverse+hold Keep me infomed if works. -- Mimmus > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EM

Re: [Asterisk-Users] Comments on Gafachi

At 02:39 PM 3/25/2006 -0500, you wrote: Has or does anyone use Gafachi s (www.gafachi.com) origination (and termination services)? If so, what can you tell me about their call quality, and have you had any problems with the service?   Comments are appreciated!   Thanks, Wes Baehr Ability Busi

RE: [Asterisk-Users] NATted phones transferring calls - BUG0003710

I wont go into the details of NATs and how they work, they are beyond the scope of this fourum,even though it is important that we understand NATs and their function as it pertains to Asterisk. If Asterisk is not in the media path and it CANNOT transfer the call if it is NATed. If you have canrein

RE: [Asterisk-Users] Agents on DND still receiving calls...

Do a trace with ngrep or ethereal and see if the phone is sending back a sip DECLINE or similar. If it is, I'd say you have an Asterisk bug. Doug. > -Original Message- > From: Stephen Kratzer [mailto:[EMAIL PROTECTED] > Sent: Tuesday, March 28, 2006 7:51 AM > To: asterisk-users@lists.dig

Re: [Asterisk-Users] Receptionist Phones

Bob McDowell wrote: > Can you chain these to get more that 42 buttons? I need about 60... > > > Bob McDowell 42+12 is fairly near the 60 target. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation provided by

[Asterisk-Users] Bluetooth stack for cordless telephone

Hi, Anyone know where I can download or purchase a Bluetooth stack that supports the CTP (Cordless Telephone Profile). Apparently this in the only way to get the answer/hangup button on a wireless headset to work with SIP soft phones. I have looked at the Widcomm site and they are not selling

[Asterisk-Users] Dropped calls

I have been experiencing dropped calls on my iax2 connections between my Asterisk server and my ITSP providers, I use Teliax and Voxee but it seems to happen on both so I don't think it is the provider. I don't see any packet loss at the time so I don't think it is poor Internet connectivity, w

[Asterisk-Users] NATted phones transferring calls - BUG0003710

I made a call from 3254102 to 2944093. I then tried to do a transfer to 3254107. IP addresses have been changed to protect the innocent. It appears this related to bug 3710. It's unclear from the bug if the problem has been fixed or not. If it hasn't, then this seems pretty serious and would I

Re: [Asterisk-Users] Receptionist Phones

We installed a snom with 3 sidecars. Kinda worked, but had so many quirks they had us replace with a Polycom. All their other phones were of the poly variety. We installed a dedicated lcd running FOP for display. Receptionist was much happier. One of the key problems was she like to set the

Re: R: [Asterisk-Users] Echo cancellation

Use 'zap show channel ...' on the channel where the call is in progress, and it should show that EC is enabled. I assume that you stopped/started Asterisk since you enabled echo cancelling in zapata.conf. A reload is not sufficient. Cheers, Steve On 3/28/06, Giordano Grandis <[EMAIL PROTECTED]>

Re: [Asterisk-Users] aah 2.7 / BRI

Hans Witvliet wrote: First encounter with * Just downloaded & installed aah-2.7 Started up AMP, but i can not find any reference towards isdn. I presume there has to be some configuration done for my Eicon-Diva-pro. Does aah actually support isdn-bri? On the mail-archive i found some references

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