[Asterisk-Users] Snom200 strange sound problem

Hello group, I've a rather strange problem with my Snom200 telephone. I'm using it in combination with SER, asterisk and rtpproxy. The telephone is behind NAT and connects to SER. It can be called without any problem from any Client on asterisk or SER. But whenever I make a call to asterisk or

[Asterisk-Users] Mysql support

I'm setting up Asterisk in a large organisation. I'm a little worried about reload (or restart graceful) every time I have to add/move/change one SIP user. I have been looking for the implementation of mysql, but I cannot find some good advise on the wiki. If I add MYSQL_FRIENDS - will I

[Asterisk-Users] Do I *need* to compile zaptel?

Hi List, I have installed a debian system through a knoppix CD and the knx2hd tool. I am using 2.6.7 kernel because 2.4 doesn't seem to support the network card I have in the box - which is a drag: from what I have read so far asterisk works better with 2.4. After doing a apt-get update

[Asterisk-Users] Dropped call

Hi all We have had Asterisk drop calls every now and then, does anyone know why this happens? It is seldom but does happen. We have plenty of memory in the server. Regards Doug Reid Director Stormcorp Network Solutions (Pty) Ltd Tel:+27 11 807 1141 Fax:+27 11 807 3504 Mobile: +27 83

Re: [Asterisk-Users] Dropped call

do you check irq misses? what is your set up? os? ,kernel?, full voip?, voip+tdm? On Fri, 29 Oct 2004 08:54:42 +0200, Doug Reid -Stormcorp [EMAIL PROTECTED] wrote: Hi all We have had Asterisk drop calls every now and then, does anyone know why this happens? It is seldom but does happen. We

RE: [Asterisk-Users] - ACAN - the AsteriskComprehensive ArchiveNetwork (was RE: GPL thoughts)

Who's talking about docs? We're referring to applets, auxiliary programs, shell scripts, perl creations and such. [EMAIL PROTECTED] wrote: Why not just use voip-info.org for this? It already seems to be the site for all things asterisk. Is there a problem with many people contributing

[Asterisk-Users] Re: ISDN-Problem with Quadbri behind Tenovis

Hello kietlak and others, Any clues to what happens here? Seems the communication asterisk=Tenovis does not work. And why is the cause not handled in chan_zap? Send your zaptel.conf and ANZG output from AOGD. Sorry, but i only managed apprenticeship on all those 3-letter-acronyms, the

[Asterisk-Users] Problem with Dial (in v.1.0.2)

I updated with version 1.0.2 of asterisk. And I experience some troubles with application Dial. In version 1.0 there was no problem. The situation is as follows. I have defined some user type=friend (for a SIP phone). But this phone is disconnected from the network. When I try to dial more then

Re: [Asterisk-Users] Do I *need* to compile zaptel?

Dave Cotton wrote: On Fri, 2004-10-29 at 10:48 +0400, Jean-Michel Hiver wrote: Now I don't have any digium hardware in this box, so I wanted to use the ztdummy driver before starting asterisk. However 'modprobe ztdummy' tells me that module ztdummy is not found. Is there a way to install

Re: [Asterisk-Users] Do I *need* to compile zaptel?

On Fri, 2004-10-29 at 11:52 +0400, Jean-Michel Hiver wrote: wild guess My guess is that the debian is distributing the 1.0 stable branch, which might happen to have no support for 2.6.x kernels, and that I need to cvs checkout the head? /wild guess Yes, Mandrake has * in its contribs

[Asterisk-Users] Modifying CDR data?

hi I've written a small AGI thing to allow lots of stuff, including diverts. If a call is placed to a diverted number, a new call is initiated from * to that number. Simple. But then, to make billing sane, I need to change the 'dst' in CDR to reflect the number diverted to. How can I do this?

Re: [Asterisk-Users] Snom200 strange sound problem

Hi, sounds for me like a typical NAT problem. I would recommend to use a more recent version like 3.52 or 3.56 you can find here http://www.snom.com/download/share/ Regards, Sven On Friday 29 October 2004 08:12, Pamela Weis wrote: Hello group, I've a rather strange problem with my Snom200

[Asterisk-Users] zapata.conf - callwaiting

Hi, I have callwaiting enabled by my telco, but my wife hates callwaiting so I tried to switch it off in * but it doesn't work: zapata.conf [channels] busydetect=yes ; to test when a line is hung-up busycount=6 ; to prevent suprious hangups echotraining=800 echocancel=yes immediate=no

Re: [Asterisk-Users] Can bad person with SIPp attack Asterisk ?

Any more info how to configure Asterisk to limit the number of calls concurrently ? Thanks in advance, Robert. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 29, 2004 12:50 AM Subject: RE: [Asterisk-Users] Can bad person with SIPp attack

RE: [Asterisk-Users] Can bad person with SIPp attack Asterisk ?

Check these url's http://www.voip-info.org/wiki-Asterisk+cmd+CheckGroup http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup http://www.voip-info.org/wiki-Asterisk+cmd+GetGroupCount Niels -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman

[Asterisk-Users] Asterisk with Nortel BCM

Hello, does anyone has an experience with connecting Asterisk to Nortel's BCM (http://www.nortelnetworks.com/products/01/eedge/bcm.html)? I would like to make this working using some voip protocol IAX, SIP, but it looks like Nortel's can't do that? My scenario is Nortel's BCM in central office

[Asterisk-Users] call another server

Good day all We have a few of asterisk server running sip and zaptel/voicetronix cards. All the server works 100 for calling in and out of the pstn and internally Some of the server are owned by the same company so I configured IAX so they can call different extensions at different branches

Re: [Asterisk-Users] Dialing a # in phone number?

I found the answer to the problem I was having with Asterisk, an X100P, a Cell Socket with Nokia phone and dialing the # character. It wasn't a problem with the # character at all - the problem was that I had the txgain set to -4.5 (which was working well on an analogue phone line). I

Re: [Asterisk-Users] Automatic codec selection

Hello, this was only an example ... The problem is, that one of my phone has a VERY VERY bad voice quality when it's not using G729, so i have to transcode if the other side is not able to use the same codec. But my main question is, if it's possible to select the codecs depending on the connected

RE: [Asterisk-Users] Automatic codec selection

I would like that too.. The easiest for me would be to be able to DISABLE codec translations of any kind Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, October 29, 2004 12:19 PM To: Asterisk Users Mailing List

[Asterisk-Users] Echo in CAPI channels

Hello all, Recently we change our traditional pbx with an Asterisk pbx, All works fine, but almost all incoming calls that we recibe, we ear our own voice like an echo in our phone. The person who we are talking ear us fine, but we ear echo and sometimes interferences on the phone. We are using

Re: [Asterisk-Users] Echo in CAPI channels

I also get echo also but whats interesting is that I only get an echo if the person at the far end (via. the telephone company) is using an analogue phone line. If they are on ISDN or a cell phone there is no echo. I'm only using basic ISDN (not primary rate) connections with chan_capi and

[Asterisk-Users] chan_sccp and Cisco 7940

Hi everyone, I have a Cisco 7940 and I'm using chan_sccp with it (chan_skinny does work fine but it seems to be very featureless compaired to chan_sccp - caller Id being probably the biggest reason to use the latter). I can make call on the 7940 but I cant answer them. The phone rings but

[Asterisk-Users] Strange problem with TDM400 FXO in UK

Hi Folks, I have a Rev H TDM01B board which seems to be working pretty well. However sometimes when I dial out on the Zap channel I get the Congestion signal on my SIP phone and a message logged in the log file saying :- Unable to create channel of type 'Zap' followed by :- Dial argument

[Asterisk-Users] Asterisk Sipphone

Title: Asterisk Sipphone Someone can tell me if my sip.conf and my extension.conf are correctly for setup Asterisk with SipPhone? I'm a newbie about Asterisk, install it on a Suse 9.1 with a wildcard x100p *SIP.CONF* [general] port=5060 binaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw

[Asterisk-Users] sip.conf registration

Sorry to be a pain reposting this question, but a mail server problem resulted in my last message missing the boat slightly: How do I associate a SIP entity with a registered account on a PSTN gateway? I have 2 register lines and 2 entities in sip.conf. When I dial into asterisk from the PSTN

[Asterisk-Users] DTMF problem

I'm really new to asterisk, but everything works fine beside DTMF recognition. As you can see in output generated by asterisk, I constantly receive following line "Private structure not found in send_digit." for every DTMF digit I enter on h323 side. I have analog phone atached to

Re: [Asterisk-Users] disa hangs up on me

I have confirmed that DISA is the culprit. If I remove DISA from the s exten, ti works as I would expect -- I can dial internal extensions after getting in via iax. DISA is an important part of the office dialplan, though, as it allows us to call in from outside and get an internal line to dial

Re: [Asterisk-Users] Echo in CAPI channels

On Fri, 29 Oct 2004, Derek Conniffe wrote: I also get echo also but whats interesting is that I only get an echo if the person at the far end (via. the telephone company) is using an analogue phone line. If they are on ISDN or a cell phone there is no echo. I'm only using basic ISDN

RE: [Asterisk-Users] * connectionto home automation server

The server is from Gira (www.gira.de) The server has an ethernet connection and it's own html server but also an s0 bus. From what I understand is that the communication over the s0 bus is supposed to go via Gira and that it's not meant to be accessed directly. (a packet sniffer will be my

Re: [Asterisk-Users] Echo in CAPI channels

Hi, I have exactly same problem. I hear echo (I guess it's more like reverb) but only if calling from analog phone. I've already asked this group about it, but didn't receive any solution answer. Maybe we should try to contact avm or something. I also didn't find any more info what these all

[Asterisk-Users] pbx_loopback.so failed

Hi, Strange error message showing up. Not sure what's causing it, hope someone on the list has seen it before. Using Asterisk on a Fedora2 using the 2.6.9 kernel. [EMAIL PROTECTED] bin]# uname -r 2.6.9 Installed Zaptel and looks fine, not got the TDM400P card yet. Downloaded Asterisk using

Re: [Asterisk-Users] Echo in CAPI channels

Hi, who does this echo canceling if you use ordinary ISDN phone (you hear no echo) ? Regards, robert. - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 29, 2004 1:12 PM Subject: Re:

Re: [Asterisk-Users] pbx_loopback.so failed

The dangers of mixing different CVS versions, fixed it anyway. I started with CVS Head, don't work with Chan_capi so switched to stable V1.0, that did'nt work. So I d/l 1.0.1 and had much more success. However, in doing all the above it left several redunant SO libraries. I guess Asterisk

[Asterisk-Users] ISDN EDSS1 protocol support

Hi all, I have to implement the following: -- | 10 voice channels |---| Prov E1 | 256 kbit/s for VoIP | Asterisk IP-PBX | | 256 kbit/s for Data

[Asterisk-Users] eyebeam video

Does anybody have the miracle setting required to get the video portion of eyebeam from Xten to actually work. All I get is blank screen. Or is it the product itself that is not quite ready for prime time yet MarcG. ___

Re: [Asterisk-Users] eyebeam video

On Fri, 29 Oct 2004 [EMAIL PROTECTED] wrote: Does anybody have the miracle setting required to get the video portion of eyebeam from Xten to actually work. All I get is blank screen. Last time I looked it seemed that Asterisk did not allow the addition of the video stream after the call

Re: [Asterisk-Users] eyebeam video

got it working with my webcam just as is use the wizard to detect and add video for sip [EMAIL PROTECTED] wrote: Does anybody have the miracle setting required to get the video portion of eyebeam from Xten to actually work. All I get is blank screen.

Re: [Asterisk-Users] Echo in CAPI channels

I've been wondering about this too. I've now got two telephone systems side by side - my old system is an analogue PBX connected to ZyXel routers (Prestige 100s) which give me POTS lines from the ISDN NT1 boxes and its only since I've started with Asterisk that I've come across echo (I have

Re: [Asterisk-Users] Can bad person with SIPp attack Asterisk ?

On Fri, 2004-10-29 at 05:20, Robert Rozman wrote: Any more info how to configure Asterisk to limit the number of calls concurrently ? This is done with app_groupcount and the SetGroup, CheckGroup, and GetGroupCount applications. More info here --

[Asterisk-Users] FOP 0.17 - Agent setup

I have searched the wiki and the web - to no avail. What data do I need to enter in the op_buttons.cfg file (and any other cfg file) in order to monitor an agent ? As an example, I have agent 5002 in the queue LID Help! Many thanks. Julian ___

[Asterisk-Users] Re: Multi-office topology suggestions

The issue is this: How can I have a phone number in a city over 1000 miles connect to the Asterisk box in an economical way? for one phone number, look at the sipura spa-3000 or its clones randy ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] Queue.conf, maxlen = 5 , but what happens with the 6. caller ?

In my queue.conf I have: ; Maximum number of people waiting in the queue (0 for unlimited);maxlen = 5 - Log: -- Executing Goto("SIP/242.112.162.21-081a8b90", "veksel|s|1") in new stack -- Goto (veksel,s,1) -- Executing Answer("SIP/242.112.162.21-081a8b90", "") in new

[Asterisk-Users] Re: Benjk's Question Why FXS

Wolf Paul wrote: How about a school strapped for cash, with around 60 POTS phones on hand and an almost free source of another 60? Versus a cost (here in Austria) of $99 for the cheapest VoIP phone (the cheapest Grandstream model). Of course that also means that FXS is only of interest if I can

[Asterisk-Users] Re: Benjk's Question Why FXS

Andrew Kohlsmith wrote: IP Phones require massive rewiring of your network infrastructure -- throwing those phones with the built-in switches in the mix is just asking for trouble. -A. I agree - if you are have a hub based architecture. But not if you are using switches. And, sharing existing

[Asterisk-Users] Grandstream HT486 and FAX

Hi All, I was trying to test to send a fax to an international number. Here's the setup: FAX -- HT486 -- SIP PROXY -- GATEWAY -- PSTN -- FAX Unfortunately I haven't been able to do it, I read somewhere that fax uses G711 only, is this true? because our gateway provider uses only G729. does

[Asterisk-Users] Security question (permissions)

I have a FWD account (511205) which should work directly into my Asterisk server. I have a sipgate.de connection as well. I have two local phone sets 601 and 602 I want that my friends (and you???) can call me, but cannot call out through sipgate.de However, I have a FWD account at work and

[Asterisk-Users] voicemail transfer on busy fails

We upgraded our software to current levels and now if an extension is busy the transfer to voicemail fails with the following message. CLI -- Starting simple switch on 'Zap/7-1' -- Executing Dial(Zap/7-1, Zap/7|15|tr) in new stack Oct 29 08:54:08 NOTICE[245775]: app_dial.c:763

Re: [Asterisk-Users] Echo in CAPI channels

On Fri, 29 Oct 2004, Derek Conniffe wrote: I've been wondering about this too. I've now got two telephone systems side by side - my old system is an analogue PBX connected to ZyXel routers (Prestige 100s) which give me POTS lines from the ISDN NT1 boxes and its only since I've started

[Asterisk-Users] wake-up

Can anybody tell me how to install and use wake-up Thanks! bye Ronald begin:vcard fn:Ronald Wiplinger n:Wiplinger;Ronald email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED]

RE: [Asterisk-Users] Asterisk with Nortel BCM

Use H.323 and in the BCM set the protocol to Other. Do you HAVE to use the BCM? It's a really horrible system. I worked for many years in tech support, and I've been involved in BCMs since the beta trials of version 1.0, four years ago. I know BCM, and I can tell you that it is one of the worst

Re: [Asterisk-Users] wake-up

On Fri, 2004-10-29 at 22:18 +0800, Ronald Wiplinger wrote: Can anybody tell me how to install and use wake-up You have a lot of reading to do, start here http://www.voip-info.org/wiki-Asterisk+auto-dial+out -- Steven Critchfield [EMAIL PROTECTED]

RE: [Asterisk-Users] zapata.conf - callwaiting

Ahh, good, I'm not the only one who hates call-waiting. You'll have to get call-waiting turned off by your telco, there's nothing you can really do in * about this. The callwaiting settings are for FXS ports, not for FXO ports. Once an FXO port is connected, the CW beep will be heard on the

Re: [Asterisk-Users] Grandstream HT486 and FAX

Hi Ronald, G.729 will not carry FAX. G.711 will, but over IP it still tends to be flaky. Reliable FAX over IP requires T.38. Steve Ronald Ramos wrote: Hi All, I was trying to test to send a fax to an international number. Here's the setup: FAX -- HT486 -- SIP PROXY -- GATEWAY -- PSTN -- FAX

[Asterisk-Users] Nothing but static on new install with TDM11B

While I have successfully compiled all the packages, I haven't been able to get asterisk to provide a dialtone for a phone attached to the TDM11B - I get static (at least I've got it to produce some noise). I'm running on a Red Hat 8 box. I've been following the instructions at:

Re: [Asterisk-Users] Echo in CAPI channels

I'm telephone company connections only (due to only having a 64Kbps fixed internet connection). Its definitely relating to the far end because it only happens when I'm talking to a person using an analogue line on the far end but the question asked by Robert Rozman a couple of hours ago is

Re: [Asterisk-Users] Queue.conf, maxlen = 5 , but what happens with the 6. caller ?

Stig Thune wrote: Because of bandwidth limitations, I have to reduce the incomming calls to 5. So when people call, and are placed in queue.. and the 6. caller comes in, he/she gets a welcome message, and then is left with a empty sound. How can I change, so that the 6. caller gets a message like:

Re: [Asterisk-Users] Do I *need* to compile zaptel?

On Fri, 29 Oct 2004 11:52:36 +0400, Jean-Michel Hiver [EMAIL PROTECTED] wrote: Dave Cotton wrote: On Fri, 2004-10-29 at 10:48 +0400, Jean-Michel Hiver wrote: Now I don't have any digium hardware in this box, so I wanted to use the ztdummy driver before starting asterisk. However

Re: [Asterisk-Users] Echo in CAPI channels

On Fri, 29 Oct 2004, Derek Conniffe wrote: I'm telephone company connections only (due to only having a 64Kbps fixed internet connection). Its definitely relating to the far end because it only happens when I'm talking to a person using an analogue line on the far end but the question

[Asterisk-Users] Re: question about asterisk

Date: Fri, 29 Oct 2004 12:23:06 +0900 From: Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] question about asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type:

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 3, Issue 410

i have a audio problem between sip and h323. First my installation: Debian Sarge Asterisk 1.0.1 Gnugk 2.0.8 Asterisk register a prefix to gnugk. Communication from sip to sip and h323 to h323 is working. When i now call from the siphone (three tested) the h323 phone (also three tested) the

Re: [Asterisk-Users] Echo in CAPI channels

Hi Peter, Ah yes - I get you now - sorry for misunderstanding and I think you've hit the nail on the head. I have a couple of budgetones and a Cisco 7940. I do have a few FXS cards so I'll have to put one in the * server and see if there is no echo when I talk to someone on a remote POTS line

Re: [Asterisk-Users] sip - h323 audio problem

i have a audio problem between sip and h323. First my installation: Debian Sarge Asterisk 1.0.1 Gnugk 2.0.8 Asterisk register a prefix to gnugk. Communication from sip to sip and h323 to h323 is working. When i now call from the siphone (three tested) the h323 phone (also three tested) the

Re: [Asterisk-Users] Echo in CAPI channels

Hi, who does this echo canceling if you use ordinary ISDN phone (you hear no echo) ? Regards, robert. Often noone. But you don't notice it because the delay is so low. So it just sounds like sidetone - where the phone plays some of your voice back in your ear. When you start

Re: [Asterisk-Users] Suggestion re: SIP/NAT/*

On Thu, 28 Oct 2004 14:45:46 -0600, Ryan Courtnage [EMAIL PROTECTED] wrote: Yep, you can do this, just requires some port forwarding and special considerations in sip.conf. You are missing the point. There is no *solution* to SIP NAT traversal. All there is are *workarounds*, otherwise known as

Re: [Asterisk-Users] Echo in CAPI channels

Please keep us posted on progress. There are more users watching you :-). Regards, Robert. - Original Message - From: Derek Conniffe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 29, 2004 5:08 PM Subject: Re:

Re: [Asterisk-Users] Re: question about asterisk

On Fri, 29 Oct 2004 07:57:03 -0600, Olger Merlos Valverde [EMAIL PROTECTED] wrote: Ok, only one question :), this card, (X100P) looks like one modem analog :) it's the same?? or have some diferents... Yes, the X100P is full-duplex, comes with a warranty that it will work with Asterisk and

Re: [Asterisk-Users] Re: question about asterisk

On Fri, 2004-10-29 at 09:57, Olger Merlos Valverde wrote: Ok, only one question :), this card, (X100P) looks like one modem analog :) it's the same?? or have some diferents... Very thanks... :) It is an analog modem with a very specific Intel/Ambient chipset that the wcfxo driver supports.

[Asterisk-Users] Outbound IAX calls stop ringing remote phone, yet can still pick up

Greetings, I've recently encountered some strange behavior placing outbound calls using IAX via a VOIP provider. Intermittently, calls placed will ring the called phone a couple times, then the ringing stops. However, even after a few seconds of silence i can pick up the phone and the call

Re: [Asterisk-Users] Outbound IAX calls stop ringing remote phone, yet can still pick up

add an r to the end to your dial statement - Original Message - From: Stephen David [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 29, 2004 12:06 PM Subject: [Asterisk-Users] Outbound IAX calls stop ringing remote phone,yet can still pick up Greetings, I've recently

Re: [Asterisk-Users] Echo in CAPI channels

:-) - Original Message - From: Robert Rozman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 29, 2004 11:32 AM Subject: Re: [Asterisk-Users] Echo in CAPI channels Please keep us posted on progress. There are more

Re: [Asterisk-Users] Outbound IAX calls stop ringing remote phone, yet can still pick up

That will, of course HIDE any BUSY or telco messages. And the caller will never know of they dialed an invalid number or of the number they dialed is busy. Do you think he REALLY wants that? Steve Totaro wrote: add an r to the end to your dial statement - Original Message - From:

Re: [Asterisk-Users] Outbound IAX calls stop ringing remote phone, yet can still pick up

then what is REALLY the solution? - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 29, 2004 12:08 PM Subject: Re: [Asterisk-Users] Outbound IAX calls stop ringing remote

Re: [Asterisk-Users] Outbound IAX calls stop ringing remote phone, yet can still pick up

Its almost the same as dialtone. Not needed but added for user ease and comfort. - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 29, 2004 12:23 PM Subject: Re:

[Asterisk-Users] AddQueueMember and call distribution

Is there anyway to get Agents (members of the queue) logged out if they do not answer a call. I'm assuming that autologoff doesn't work with AddQueueMember. I've searched the wiki and google to death... Currently, when a call comes in to the queue, one member just rings and rings and rings. It

[Asterisk-Users] Snom 190/220

Good Day list, I have spent better part of the morning reading through the user group messages and have found some people stating that they are able to get the Transfer Button to work on the Snom 190/220 However, I am unable to find HOW they pulled this off. When I press the transfer button it

Re: [Asterisk-Users] Echo in CAPI channels

If you use a voip technology phone such as a softphone or a hard ip phone you will have at an absolute minimum an additional 40ms round trip time, but more realisitcally twice that at least. That moves the echo from the sidetone to actually being perceived as an echo. That what is seems.

Re: [Asterisk-Users] Suggestion re: SIP/NAT/*

NONSENSE Benjamin on Asterisk Mailing Lists wrote: On Thu, 28 Oct 2004 14:45:46 -0600, Ryan Courtnage [EMAIL PROTECTED] wrote: Yep, you can do this, just requires some port forwarding and special considerations in sip.conf. You are missing the point. There is no *solution* to SIP NAT

[Asterisk-Users] Asterisk with own IP address

If a server running Asterisk has its own IP address while SIP phones are behind a NAT router on a different IP address, the phones should be able to contact and register with the Asterisk server, right? I'm sure the answer is yes, because I can register my SIP phones with other Asterisk

RE: [Asterisk-Users] Suggestion re: SIP/NAT/*

Karl Are you saying it is nonsense that there difficulties using Asterisk and SIP behind a NAT server. Or are you saying it is nonsense that SIP and NAT are dangerous together? Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karl Brose

RE: [Asterisk-Users] G729 and Sipura.

I am having the same problem, it doesn't work on my SNOM either. Below is my sip.conf .. On both sipura and SNOM I get same results, I can hear voice but not send voice. sip show channels: Peer User/ANRCall ID Seq (Tx/Rx) Format xx.xx.xx.xx 811 3c26723bacd

Re: [Asterisk-Users] AddQueueMember and call distribution

Sean, We're in the same boat. What I have ended up doing was to use the roundrobin strategy. Round robin will ring each queue member one after the other in a loop until someone answers. It still doesn't address the issue with auto-logout. I have been toying with the idea of making a custom

Re: [Asterisk-Users] Suggestion re: SIP/NAT/*

Also karl, what are you basing your statement on ? *g On Fri, 29 Oct 2004 18:01:50 +0100, Bill Seddon [EMAIL PROTECTED] wrote: Karl Are you saying it is nonsense that there difficulties using Asterisk and SIP behind a NAT server. Or are you saying it is nonsense that SIP and NAT are

Re: [Asterisk-Users] Snom 190/220

Ronald, No luck is getting the Transfer button working here. However, there is a Transfer button that lights up on the LCD when you are on a call. I believe this is the one that people are most likely referring to when they say they got things working. It would be nice to be able to program

Re: [Asterisk-Users] Asterisk with own IP address

http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions #8 - Original Message - From: Bill Seddon [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Friday, October 29, 2004 12:58 PM Subject: [Asterisk-Users] Asterisk with own IP

Re: [Asterisk-Users] Suggestion re: SIP/NAT/*

I would agree that it is not good to suggest or impliment a solution that is not a Best Practice unless it is a last resort. - Original Message - From: Bill Seddon [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Friday, October

RE: [Asterisk-Users] G729 and Sipura.

On Fri, 2004-10-29 at 12:17, Matt Schulte wrote: I am having the same problem, it doesn't work on my SNOM either. Below is my sip.conf .. On both sipura and SNOM I get same results, I can hear voice but not send voice. When you do a show g729 on the CLI do you get that the license is

RE: [Asterisk-Users] G729 and Sipura.

Yes it shows up. turns out my SNOM maybe a piece of sh*t. It almost looks like Asterisk is responding with g723, or am I backwards./ This is part of SIP debug. I called Digium support, they blamed the SIP devices.. ideas ?? --snip Capabilities: us -

[Asterisk-Users] sip phones...

By any chance has anyone been able to get any of the 3com phones to work with asterisk? If not is there anyone out there who want's to give it a go? -- Cheers, Mikel King Optimized Computer Solutions, INC 39 West Fourteenth Street Second Floor New York, NY 10011 http://www.ocsny.com t:

[Asterisk-Users] Hiss on Line, No ringing thru VoicePulse?

Hi All, Relatively new to *, but I've got it basically working. I'm having a bit of trouble and need a pointer on where to start. I have a TDM400P with one FXS and one FXO (Developer's kit), and also am using IAXComm on a PC with a headset and an Intel 810 sound card. Calls from the headset

Re: [Asterisk-Users] Suggestion re: SIP/NAT/*

All there is are *workarounds*, otherwise known as bad and rather dangerous hacks. Whether it works or not is highly dependent on external factors that you don't usually control. It also depends on the type of NAT/PAT your router is using, ie the router's particular NAT/PAT implementation.

Re: [Asterisk-Users] iLBC/PCM16 Huge Cost

Michael Loftis wrote: Little tireds now so you may have already done all this but make sure you have latest libpri, zaptel, and asterisk, in that order. It seems that disabling MMX support in zaptel fixed *all* my problems, from hold music, to iLBC times, to random crashes. That's odd though,

[Asterisk-Users] Anyone using Voipjet?

I've used them for calls terminating in the US with good results. I happened to put through a call to Romania today and it seemed the person was hearing me very much lagged behind. The actual asterisk IAX figure given was like 80 ms which is usually pretty decent for talking to someone.

Re: [Asterisk-Users] Snom 190/220

Jeremy Rusnak wrote: It would be nice to be able to program the transfer button behavior on these phones. I connected a Snom190 (fw3.56, sip) to * a few days ago and was surprised to see that not only the transfer button but also the programmable buttons on the side (with the leds) worked

Re: [Asterisk-Users] FOP 0.17 - Agent setup

Hi Julian, On Fri, 29 Oct 2004 14:20:21 +0100, Asterisk [EMAIL PROTECTED] wrote: I have searched the wiki and the web - to no avail. What data do I need to enter in the op_buttons.cfg file (and any other cfg file) in order to monitor an agent ? As an example, I have agent 5002 in the

Re: [Asterisk-Users] AddQueueMember and call distribution

Hi Jeremy, On Fri, 29 Oct 2004 13:19:33 -0400, Jeremy Rusnak [EMAIL PROTECTED] wrote: Sean, We're in the same boat. What I have ended up doing was to use the roundrobin strategy. Round robin will ring each queue member one after the other in a loop until someone answers. [snip] For the

[Asterisk-Users] Newbie question - pickup call waiting on an analog trunk

I have a TDM31B with one FXO port. The phone company provides call waiting and the ability to switchover to that party. My problem is that now that I'm running it through asterisk, if I'm on a call and I get another call, I get the caller id info and the tones, but I can't switchover (via

Re: [Asterisk-Users] Hiss on Line, No ringing thru VoicePulse?

is there a mute setting, this can help isolate if its the mic or not. - Original Message - From: Scott Sharkey [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 29, 2004 1:59 PM Subject: [Asterisk-Users] Hiss on Line, No ringing thru VoicePulse? Hi All, Relatively new to *,

[Asterisk-Users] Is NuFone messing up for anybody else?

Weve been using NuFone for about 2 months, pushing an average of 10,000 minutes a month of Long distance. There have been minor quality issues in the past, maybe to one area code or another. However, Ive noticed today more problems than usual. Gotten several calls. Has anybody else noticed

Re: [Asterisk-Users] Suggestion re: SIP/NAT/*

Thanks to everyone for your input. I've chosen to register my * server with FWD's IAX service, and have my remote SIP users register as FWD clients. I think this will solve my biggest problems, and give me the added benefit of having voice mail available if my * server is offline. -

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