RE: [Asterisk-Users] dtmf problem

Bart, We have has similar issues with BroadVoice in the past. From what I understand they had problems with DTMF depending on which proxy you register to. This is a bug that related to their session border controllers which should have been resolved. Looking at your config your first each account

Re: [Asterisk-Users] 3 Phone Call Qualtiy Issues

On Dec 20, 2005, at 1:01 PM, Rhonda Herron wrote: Hi, I have been battling with the following problems for a while and was hoping someone could shed some light on the subject. I am using AT320 402 IAX2 phones with 1.49 firmware (latest) connected to an Asterisk server running [EMAIL PROTECT

Re: [Asterisk-Users] MFC/R2

Hi     I have a few question about unicall       What version of unicall is stable  0.0.2e or 0.0.3pre8   i am using 0.0.3pre8 but i have a litle problems   1.- I can't send or recived any fax. i put the faxdetect in the unicall.conf but your code ignore this. 2.- When the user dial some numbers th

Re: [Asterisk-Users] fxs woes...

hello again, coming back to the battle i fight with the fxs's, i put the card in another asterisk box and it seemd to work out just fine. could it be a problem with the pci bus? how am i supposed to debug that!?! any other ideas as to where the problem lies? -saad - Original Message - From

RE: [Asterisk-Users] Latest Source

Brian, you can take your reply and shove it up your ass. I followed their directions, and made the mistake of using an IP instead of a FQDN and as a result it didn't work. I posted a question to the group. Ooer! Kevin Fleming replied with a solution. Question, answer. Solved. Don't make

RE: [Asterisk-Users] Latest Source

Point taken. That's how I got the IP address I used... I simply resolved the fqdn to the IP address. I'll try with the fqdn tomorrow (and I don't think it said 'specifically' that I 'had' to use a fqdn). -Original Message- From: Peter Bowyer [mailto:[EMAIL PROTECTED]

RE: [Asterisk-Users] Latest Source

Peter. I followed the directions at that URL, and it failed. Hence the subject 'Latest source' and my question where I posted what I'd done and the error I got. Kevin Fleming replied to it with a solution. -Original Message- From: Peter Bowyer [mailto:[EMAIL PROTECTED]

Re: [Asterisk-Users] Latest Source

Peter Bowyer wrote: Any chance you could learn to use email? Replying to an unrelated thread is unlikely to win you many friends. It might be bit hard to follow for the RTFM-challenged, but the page with the clearly-too-obvious URL 'http://www.asterisk.org/download',which is reached from the

Re: [Asterisk-Users] Latest Source

On 21/12/05, Douglas Garstang <[EMAIL PROTECTED]> wrote: > They still support cvs? I was reading their patch docs (not that I want to > make a patch), but it said to use: > > [EMAIL PROTECTED] ~]# svn checkout http://216.27.40.102/svn/asterisk/trunk > asterisk No, it said to use svn.digium.com,

Re: [Asterisk-Users] Latest Source

On 21/12/05, Douglas Garstang <[EMAIL PROTECTED]> wrote: > No idea on this. Can't find it on digium's web site. How do we download the > latest source for Asterisk? Looks like they switched to SVN from CVS? Never > used it... is there a Linux client for it? > > Doug. Any chance you could learn to

RE: [Asterisk-Users] Busy recognition

I've found that I can only get ${DIALSTATUS} to work reliably if I have a qualify= directive in sip.conf against the user. If you don't put the qualify= directive in there, Asterisk will just ring and ring until the ring timeout occurs. If you do put it in there, and Asterisk has had a chance to

Re: [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?

Interesting that Skype cant prevent itself becoming a super-node unlike KaZaa. Wonder what that does to capped ADSL lines in South Africa...RobOn 12/19/05, Paul Hewlett <[EMAIL PROTECTED]> wrote: On Monday 19 December 2005 14:23, Francesco Peeters (Asterisk) wrote:> On Mon, December 19, 2005 11:33

[Asterisk-Users] Unicall Problem with fax

Hi      I can't recive or send any fax  using a unical truck, i put  faxdetect=incoming in the unicall.conf but asterisk ignore  it   The log show   Dec 21 06:21:10 VERBOSE[22400]:  [chan_unicall.so] => (Unified call processing (UniCall))Dec 21 06:21:10 DEBUG[22400]: Parsing /etc/asterisk/unicall.c

RE: [Asterisk-Users] SIP Subscriptions

Yes. SER has been a saviour here. That was how we managed to get registration info to all the Asterisk boxes, but using SER. Phones register with OpenSER and it forwards() the registrations out. Seems to be working for now, but that solution isn't without it's warts too. -Original

RE: Re: [Asterisk-Users] SIP Subscriptions

>-Original Message- >From: C F [mailto:[EMAIL PROTECTED] >Sent: Tue 12/20/2005 9:58 PM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Cc: >Subject: Re: Re: [Asterisk-Users] SIP Subscriptions

Re: [Asterisk-Users] SIP Subscriptions

I don't think anyone will deny that subscriptions have long been a neglected feature.  You know... SER might do the trick. :)  The default answer for all things SIP that Asterisk does not do. On 12/20/05, Douglas Garstang <[EMAIL PROTECTED]> wrote: So I see that if you issue a 'reload' command in A

RE: Re: [Asterisk-Users] SIP Subscriptions

Like what? I'm being very serious and firmly believe in what I said below. What's your issue with it? I'd really like to know, seriously, because I just can't understand why others don't have a problem with this. -Original Message- From: Steve Totaro [mailto:[EMAIL PROTE

RE: [Asterisk-Users] SIP Subscriptions

I respectfully disagree that my complaints are idle. Digium as 'Digium™ is the creator and primary developer of Asterisk™, ' state that it is 'enterprise-grade '. I feel that the things I have complained about disqualify it as being enterprise-grade. That's not idle. If it's enterprise-grade, th

Re: [Asterisk-Users] Re: Mulitple voicemail on mulitple phones

Steven wrote: What is the proper way to email to multiple email addresses. I have been intending to also email my cell phone when there is a message, but have yet to try different options like comma, semicolon, etc. Use an alias in your mail transfer agent (MTA) to expand it into the addresses

Re: [Asterisk-Users] How do I remove the temp greeting?!?!

tracinet wrote: I don't know about you, but my option 4 says "change password" yet when I press it, it does give me the option to remove the temp greeting. A bit confusing - I agree. That means you haven't updated your sound files to match the code you've installed :-) ___

Re: [Asterisk-Users] RFC 3262 PRACK

Trond Andersen wrote: Does Asterisk have SIP support for PRACK/100rel? No. Asterisk does also seem to filter out some SIP header fields. Is there a way I can force Asterisk to pass on ALL SIP header fields? Asterisk is not a proxy. It does not pass on (or filter out) any header fields, it

Re: [Asterisk-Users] Latest Source

Douglas Garstang wrote: [EMAIL PROTECTED] ~]# svn checkout http://216.27.40.102/svn/asterisk/trunk asterisk and I get: svn: PROPFIND request failed on '/' svn: PROPFIND of '/': 200 OK (http://216.27.40.102) Had to use IP address... no DNS on test box... The Apache installation on that syste

Re: Re: [Asterisk-Users] SIP Subscriptions

Lets get this clear once and for all, you are a Fuc*ing retard, and this will be the last (I hope) you will hear from me in this post. And now to the post: On 12/20/05, Douglas Garstang <[EMAIL PROTECTED]> wrote: > Well, I know I will be attacked for saying this, as cursed are those who say > an

RE: [Asterisk-Users] Latest Source

Times like now, I miss Critchfield... > > Yes, I saw that earlier and may have taken more notice of that note if the > poster had documented the symptoms associated with the fix (like maybe the > errors etc). I saw nothing there to relate that note back to the problem I > was having. At that poin

RE: Re: [Asterisk-Users] SIP Subscriptions

I hope you are just having a breakdown and aren't normally like this. > > Well, I know I will be attacked for saying this, as cursed are those who > say anything bad about Asterisk, but for an application that is > 'enterprise-grade' as Digium tourts, it has several flaws that IMHO > disqualify

Re: [Asterisk-Users] Asterisk and Adtran TA 750 Channel Bank -- odd behavior (help!)

I just wanted to follow up with the solution for the searchable archives. To summarize: Adtran TA 750 w/ FXO lines from a Nortel Meridien Problem: DTMF digits not being decoded properly Solution: Set the TX/RX Attenuation on the TA750 to 9.0db (max) The sound is a little low now, but DTMF is w

RE: [Asterisk-Users] Latest Source

Yes, I saw that earlier and may have taken more notice of that note if the poster had documented the symptoms associated with the fix (like maybe the errors etc). I saw nothing there to relate that note back to the problem I was having. At that point I also didn't know udev was related.

Re: [Asterisk-Users] Latest Source

ignore my last message, i'm retarded... too many D's with first names :) On Dec 20, 2005, at 10:27 PM, Darren Wiebe wrote: http://www.voip-info.org/wiki-Asterisk+timer+ztdummy Search that page for udev. Douglas Garstang wrote: There's no info on the wiki about udev in relation to ztdummy.

RE: [Asterisk-Users] Latest Source

Yes, I saw that earlier and may have taken more notice of that note if the poster had documented the symptoms associated with the fix (like maybe the errors etc). I saw nothing there to relate that note back to the problem I was having. At that point I also didn't know udev was related.

Re: [Asterisk-Users] Latest Source

It's a wiki, if you don't like the fact that the command "make install" actually warns about it when you're running udev, then add it to the page. On Dec 20, 2005, at 10:27 PM, Darren Wiebe wrote: http://www.voip-info.org/wiki-Asterisk+timer+ztdummy Search that page for udev. Douglas Ga

Re: [Asterisk-Users] Latest Source

Last time I checked google gave me 494 results in 0.59 seconds for the following query: http://www.google.com/search?hl=en&q=udev+site%3Avoip-info.org&btnG=Google+Search and 131 results in 0.59 seconds for the following: http://www.google.com/search?hl=en&lr=&q=udev+ztdummy+site%3Avoip-info.org&btn

Re: [Asterisk-Users] SIP Subscriptions

Your complaints are idle, because you provide no real solution to the problem, other than "digium needs better qa" and various other comments about how crappy asterisk is. I've been working with asterisk for 8-9 months now, have been utterly pissed off at the system for weeks, but it takes

[Asterisk-Users] dtmf problem

hi everyone, I do have 2 lines with broadvoice. >From 2 days on one line my dtmf tones are not passed to asterisk server. It siply goes through the extensions routine acting link it did not receive any tone. Could it be problem with my config??? It looks like this:(it worked for last 1.5 year) n

Re: [Asterisk-Users] Latest Source

http://www.voip-info.org/wiki-Asterisk+timer+ztdummy Search that page for udev. Douglas Garstang wrote: There's no info on the wiki about udev in relation to ztdummy. I've been involved with Linux since 1993, and I've never even heard of udev. Yes, I'll get called stupid, but I can't know e

RE: Re: [Asterisk-Users] SIP Subscriptions

Well, I know I will be attacked for saying this, as cursed are those who say anything bad about Asterisk, but for an application that is 'enterprise-grade' as Digium tourts, it has several flaws that IMHO disqualify it as 'enterprise-grade'. I mean, really, c'mon... think about it... an applicat

RE: [Asterisk-Users] SIP Subscriptions

Since when where you in a position to come to the conclusion that my complaints are idle? My complains and questions are valid, just like anyone else's. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Tue 12/20/2005 9:08 PM To: Aste

RE: [Asterisk-Users] Latest Source

There's no info on the wiki about udev in relation to ztdummy. I've been involved with Linux since 1993, and I've never even heard of udev. Yes, I'll get called stupid, but I can't know everything. Considering that there was no link to udev anywhere in the ztdummy docs, I really didn't know to e

[Asterisk-Users] Digium E1 Card Modprobe problems

e configsHi guys, I Hav just installed asterisk on a server which has a VIA M800 motherboard and we have 1 TE110 E1 card installed. After zaptel.conf was edited, asterisk runs and the E1 card is active. However after each restart, first Kudze tells me the E1 card has been removed and

Re: [Asterisk-Users] SIP Subscriptions

Douglas, the asterisk community is largely made up of volunteers. I have worked with volunteers in other places. Here is one thing they have in common. A volunteer does not usually feel obligated to put up with non constructive criticism. I smiled as I read some of your posts. You may not

Re: [Asterisk-Users] SIP Subscriptions

This email is a perfect example. "Your issue with that is...?" Your sarcastic attitude is grating on a lot of people's nerves, which is what his email was about. If you can't put in constructive criticism with a better attitude and less "asterisk sucks", then go somewhere else with your c

Re: [Asterisk-Users] Latest Source

ztdummy and udev comes to mind On 12/20/05, Douglas Garstang <[EMAIL PROTECTED]> wrote: > Oh, I'm sorry. If you could provide a detailed list of other specific > situations where I forgot to RTFM, I'd appreciate it. Thanks. > > -Original Message- > From: C F [mailto:[EMAIL

RE: [Asterisk-Users] SIP Subscriptions

I don't know what the rules are for this list, but it wouldn't be much of a stretch to assume that personal attacks are grounds for removal. While we're at it, why does my reluctance to deal with contracts, legal fees and international boundaries make me a 'fuc*ing retard'? -Origin

RE: [Asterisk-Users] Latest Source

Oh, I'm sorry. If you could provide a detailed list of other specific situations where I forgot to RTFM, I'd appreciate it. Thanks. -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Tue 12/20/2005 8:56 PM To: Asterisk Users Mailing List - Non-

Re: [Asterisk-Users] Latest Source

In M$outlook click on Tools > Options select Preference then Email-Option then play around with on Replies and Forwards. Again you forgot to RTFM. On 12/20/05, Douglas Garstang <[EMAIL PROTECTED]> wrote: > I would if I knew how... Fraid I'm spending all my time on Asterisk, and not > enough on M

RE: [Asterisk-Users] SIP Subscriptions

Seems someone has some anger management issues. As I just stated in a previous post, it seems you have issues with me asking valid questions. I'm not sure why that is. The long email you rattled off with all my questions where quite valid. Your issue with that is.? -Original Mes

RE: [Asterisk-Users] SIP Subscriptions

Good grief. Your perceptions of reality are quite warped. It seems you have a problem with me asking a lot of valid questions. Why exactly is that? -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Tue 12/20/2005 7:24 PM To: Asterisk Users M

RE: [Asterisk-Users] Latest Source

I would if I knew how... Fraid I'm spending all my time on Asterisk, and not enough on Microsoft Outlook. No idea how to turn this on in Outlook. -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Tue 12/20/2005 6:39 PM To: Asterisk Us

Re: [Asterisk-Users] IVR Capacity

Serge Schumacher wrote: Hi,   Do you think * could play around 300 voicemenu messages simoultanously?   You would need to test, but assuming this is an 'IVR' you should also take the following into consideration. 300 channels is a lot of business, and thought you might

Re: Re: [Asterisk-Users] SIP Subscriptions

But the beans and cabbage remind me more of duggie. PaulH > Aaron Daniel <[EMAIL PROTECTED]> wrote: > > OH, and don't forget pizza! > > Aaron > On Dec 20, 2005, at 9:02 PM, [EMAIL PROTECTED] wrote: > > >> C F <[EMAIL PROTECTED]> wrote: > >> > >> On 12/20/05, Douglas Garstang <[EMAIL PROTECTED

Re: [Asterisk-Users] Asterisk based IVR/VoiceMail Server for a Unified Messaging suite

I am pretty sure Asterisk can handle point 1. PaulH > ammad jami <[EMAIL PROTECTED]> wrote: > > Hello: > > Please go through this mail, it won't take much time. > > For a Unified Messaging system, I have to develop an IVR/Voicemail > system > based on asterisk. > ** > Following are the deta

Re: [Asterisk-Users] SIP Subscriptions

OH, and don't forget pizza! Aaron On Dec 20, 2005, at 9:02 PM, [EMAIL PROTECTED] wrote: C F <[EMAIL PROTECTED]> wrote: On 12/20/05, Douglas Garstang <[EMAIL PROTECTED]> wrote: Digium needs people like me, if they read this list that is. They sure don't seem to be able to make real-world fun

[Asterisk-Users] low audio volume on recorded .wav voicemail messages

The audio volume of voicemail messages (msgNNN.wav) is rather low. Is there a parameter/option to adjust gain? In my voicemail.conf, I use these formats: format=wav49|gsm|wav Maybe I should use a different format? ___ --Bandwidth and Colocation pro

Re: Re: [Asterisk-Users] SIP Subscriptions

> C F <[EMAIL PROTECTED]> wrote: > > On 12/20/05, Douglas Garstang <[EMAIL PROTECTED]> wrote: > > Digium needs people like me, if they read this list that is. They sure > don't seem to be able to make real-world functionality decisions on > their own. > > > > You are so right, how did we do wit

Re: [Asterisk-Users] Sangoma E1 board Experience

hi Steve - 1ms ensured good EC convergence, using software EC. Adding delay really degrades the performance of an EC adaption loop. It may be a block size of 2 or Yes, but... As mentioned some pentiums are not that happy about a 1ms block and some oy the algorithms in the voice path nee

Re: [Asterisk-Users] Linking existing channels through Managerinterface. Is it possible?

There are many reasons that is not desirable, least of which is that when one party hangs up the other party is still on the line. The other bad part is that it takes more resources to do meetme and if someone is running a bare-bones asterisk setup without a hardware timer or ztdummy then meetme wo

Re: [Asterisk-Users] Sangoma E1 board Experience

hi David, Thanks. 1. For lowest "cost" echo cancellation in software you need the lowest possible delay between outbound and inbound data streams, which demands the minimum possible buffering. Not sure I understand what you actually mean? You need to buffer voice anyway for echo cancel itse

Re: [Asterisk-Users] Re: Mulitple voicemail on mulitple phones

On 12/20/05, Steven <[EMAIL PROTECTED]> wrote: > What is the proper way to email to multiple email addresses. > I have been intending to also email my cell phone when there is a message, > but have yet to try different options like comma, > semicolon, etc. 1. Create an email address that is an a

Re: [Asterisk-Users] Linking existing channels through Managerinterface. Is it possible?

The other obvious option is to redirect them both to a meetme room. On 12/20/05, Matt Florell <[EMAIL PROTECTED]> wrote: > I wrote a patch to do just this quite a while ago. Have been using it > in production since Asterisk 1.0.6. Here's the bug tracker link: > http://bugs.digium.com/view.php?id=

[Asterisk-Users] Asterisk based IVR/VoiceMail Server for a Unified Messaging suite

Hello:   Please go through this mail, it won't take much time.   For a Unified Messaging system, I have to develop an IVR/Voicemail system based on asterisk.    Following are the details of the voice mail / IVR (Interactive voice Response) system to be developed:  Develop using Asterisk 100 simul

Re: [Asterisk-Users] Sangoma E1 board Experience

You're still not answering his question. :-) The TJ320 has no buffering capability but the Silabs parts have nothing to do with that, and neither do the framers on the higher-density TDM cards. I think his question is more "why don't these cards have bigger PCM buffers and interrupt less o

Re: [Asterisk-Users] SIP Subscriptions

On 12/20/05, Douglas Garstang <[EMAIL PROTECTED]> wrote: > I don't think the bounties are worth the costs asociated with contracts, > legal fees, international boundaries etc. Right, Avaya is. This guy is a Fuc*ing retard. Douglas, did you see a doc the last few days??? _

[Asterisk-Users] Asterisk based IVR/VoiceMail Server for a Unified Messaging suite

Hello:   Please go through this mail, it won't take much time.   For a Unified Messaging system, I have to develop an IVR/Voicemail system based on asterisk.    Following are the details of the voice mail / IVR (Interactive voice Response) system to be developed:  Develop using Asterisk 100 simul

Re: [Asterisk-Users] SIP Subscriptions

On 12/20/05, Douglas Garstang <[EMAIL PROTECTED]> wrote: > Digium needs people like me, if they read this list that is. They sure don't > seem to be able to make real-world functionality decisions on their own. > You are so right, how did we do without you the last few years??? __

Re: [Asterisk-Users] SIP Subscriptions

Google tells me that the first message from you on this list is on Dec 6, while there might be an error in that, I doubt that Google is off by more than a week. (http://www.google.com/search?hl=en&q=%22Douglas+Garstang%22+site%3Adigium.com&btnG=Google+Search) We have done well overhere on this lis

RE: [Asterisk-Users] What is the best Dell Machine for Asterisk?

We have more than a hundred Dell servers in production at customers. We use them because we can have them serviced easily, just about anywhere. They are principally 1850s and 2850s, or their predecessors, in T1 and larger applications. The reported IRQ problems are easily avoided if, for examp

Re: [Asterisk-Users] What is the best Dell Machine for Asterisk?

On 12/18/05, Hiu Yen Onn <[EMAIL PROTECTED]> wrote: > >>hi all, > >> > >>What is the best Asterisk-compliant for Dell machine is recommended? I > >>will have roughly 400 users in a production office. thanks!! > >>___ We turned up a T/E411P card with the

Re: [Asterisk-Users] Asterisk Hardware recomendation

hi, i am also looking for a hardware specfication that suit asterisk. would u mind to show all your hardware spec for your asterisk server??? thanks Roman Volf wrote: Krystian Filiks wrote: What about plain g729? My main concern is the Hardware, anyone that can tell me if this Supermicro 6

RE: [Asterisk-Users] Latest Source

Doug, Can you turn on > indenting on replies? Your emails are hard to figure out who is saying what. Thanks, Steve > > They still support cvs? I was reading their patch docs (not that I want to > make a patch), but it said to use: > > [EMAIL PROTECTED] ~]# svn checkout http://216.27.40.102/svn

Re: [Asterisk-Users] What is the best Dell Machine for Asterisk?

Last time I checked Dell does support Linux. http://configure.us.dell.com/dellstore/config.aspx?c=us&cs=04&kc=6W300&l=en&oc=pe2850-max&s=bsd Scroll down to the OS chose and you'll see out of 26 choices only 4 are M$, the rest minus one are for a total of 21 linux choices. On 12/19/05, Walt Reed <[

Re: [Asterisk-Users] 1.2.1 Queues

Enought ranting now to the problem: On 12/20/05, Douglas Garstang <[EMAIL PROTECTED]> wrote: > > Are queues in 1.2.1 completely screwed up or what? > The 'announce' command doesn't do anything. I'm getting this on the console > which may be related: > > Dec 20 10:32:00 WARNING[3467]: file.c:508 as

Re: [Asterisk-Users] Latest Source

Apparently there is a daily mirror copying svn -> cvs, so it will not be truly bleeding edge, but really close. Moj Tony Hoyle wrote: Douglas Garstang wrote: No idea on this. Can't find it on digium's web site. How do we download the latest source for Asterisk? Looks like they switched to SV

Re: [Asterisk-Users] 1.2.1 Queues

Does digium have any sort of QA process for Asterisk??? Yes, they do. See this link: http://www.digium.com/index.php?menu=product_detail&product=ABE "Proven Reliability Through Regression Testing" among other things for $995 Seriously, though, you can discuss problems without the attitude and

RE: [Asterisk-Users] Latest Source

They still support cvs? I was reading their patch docs (not that I want to make a patch), but it said to use: [EMAIL PROTECTED] ~]# svn checkout http://216.27.40.102/svn/asterisk/trunk asterisk and I get: svn: PROPFIND request failed on '/' svn: PROPFIND of '/': 200 OK (http://216.27.40.102) H

Re: [Asterisk-Users] Latest Source

Douglas Garstang wrote: No idea on this. Can't find it on digium's web site. How do we download the latest source for Asterisk? Looks like they switched to SVN from CVS? Never used it... is there a Linux client for it? I've just done an update now and it works fine... Are you using the right

[Asterisk-Users] Latest Source

No idea on this. Can't find it on digium's web site. How do we download the latest source for Asterisk? Looks like they switched to SVN from CVS? Never used it... is there a Linux client for it?   Doug. -Original Message-From: Rehan Ahmed [mailto:[EMAIL PROTECTED]Sent: Tuesday,

RE: [Asterisk-Users] RE: SIP Subscriptions

No... I'm using standard 1.2.1. I could... and I may I'll keep this in mind. But... for now I'm not sure if I want to introduce more variables. -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 20, 2005 4:58 PM To: Asterisk Users Mailing List - Non-

Re: [Asterisk-Users] Affordable IP Phones for Asterisk

We're using a Budgetone 101 ($60) SIP phone. It works pretty well. No echo cancellation, though, which is a little annoying when used somewhere with significant ping-times to the server. Phil Rehan Ahmed wrote: Hello Dakota, I have a few that i can ship you from vida21.com

RE: [Asterisk-Users] IVR Capacity

No unfortunatly this is not an option, we would never be able to reach them all with 48 hours. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: mercredi 21 décembre 2005 00:04 To: Asterisk Users Mailing List - Non-Commercial Discussi

[Asterisk-Users] Unicall E1 Error in Mexico

Hi     I have a weired problem. when i make a call with some numer the unicall show me a error.   For example when i dial 30003300 in mexico city then log show   MFC/R2 UniCall/3 R2 prot. err. [2/  40/Group I   /DNIS ] cause 32769 - T1 timed outDec 21 00:22:46 WARNING[17649]: MFC

Re: [Asterisk-Users] RE: SIP Subscriptions

On 12/20/05, Douglas Garstang <[EMAIL PROTECTED]> wrote: > Dang. So we still have a subscription buildup. I didn't even realise Asterisk > was ignoring the expiry. Pardon me while I cry! > Are you running a version of the code later than this past Friday? http://svn.digium.com/view/asterisk?re

Re: [Asterisk-Users] Got SUBSCRIBE for extensions without hint

Many thanks, Douglas - that was it! On Dec 20, 2005, at 3:49 PM, Douglas Garstang wrote: Anthony, Your getting SIP SUBSCRIBE messages from the Polycom phones. Check the -directory file and look for directory entries that have 1. For each of these, the phone tries to send a SUBSCRIBE to the

Re: [Asterisk-Users] Affordable IP Phones for Asterisk

Hello Dakota,   I have a few that i can ship you from vida21.com for 70$ each they with me in the US.   The client is now using cisco i have 70 pcs with me   Rehan   On 12/20/05, Dakota <[EMAIL PROTECTED]> wrote: Are there any IP Phones that can work with Asterisk, that cost less than $60?if so, wh

RE: [Asterisk-Users] Meetme and ztdummy

Thanks. That did the trick. :) -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 20, 2005 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Meetme and ztdummy On 12/20/05, Douglas Garstang <[EMAIL PROTECT

Re: [Asterisk-Users] Got SUBSCRIBE for extensions without hint

On 12/20/05, Anthony Rodgers <[EMAIL PROTECTED]> wrote: > Hi there, > > I'm getting a bunch of these errors from Polycom phones in 1.2.1: > > ERROR[24301]: chan_sip.c:10790 handle_request_subscribe: Got SUBSCRIBE > for extensions without hint. Please add hint to 4003 in context > internal > > I've

RE: [Asterisk-Users] RE: SIP Subscriptions

Dang. So we still have a subscription buildup. I didn't even realise Asterisk was ignoring the expiry. Pardon me while I cry! -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 20, 2005 4:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

RE: [Asterisk-Users] Got SUBSCRIBE for extensions without hint

Anthony, Your getting SIP SUBSCRIBE messages from the Polycom phones. Check the -directory file and look for directory entries that have 1. For each of these, the phone tries to send a SUBSCRIBE to the (contact) field. If you don't have corresponding hint extensions in your dialplan, Asterisk

Re: [Asterisk-Users] RE: SIP Subscriptions

On 12/20/05, Douglas Garstang <[EMAIL PROTECTED]> wrote: > I don't think it's an expiry issue. When a phone reboots before the expiry, > it sends a new subscription to Asterisk. Asterisk doesn't remove the old one. > If subscriptions where keyed like registrations, the new one would just > overw

[Asterisk-Users] Got SUBSCRIBE for extensions without hint

Hi there, I'm getting a bunch of these errors from Polycom phones in 1.2.1: ERROR[24301]: chan_sip.c:10790 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 4003 in context internal I've searched the Wiki and archives to no avail - what do these errors

Re: [Asterisk-Users] Meetme and ztdummy

On 12/20/05, Douglas Garstang <[EMAIL PROTECTED]> wrote: > > > Can't get meetme to work. The ztdummy driver is loaded: > > [EMAIL PROTECTED] asterisk-1.2.1]# lsmod | grep ztdummy > ztdummy 7748 0 > zaptel196612 1 ztdummy > > Although it won't load under normal cond

RE: [Asterisk-Users] RE: SIP Subscriptions

I don't think it's an expiry issue. When a phone reboots before the expiry, it sends a new subscription to Asterisk. Asterisk doesn't remove the old one. If subscriptions where keyed like registrations, the new one would just overwrite the old one. -Original Message- From: BJ Weschke [m

[Asterisk-Users] Meetme and ztdummy

Can't get meetme to work. The ztdummy driver is loaded:[EMAIL PROTECTED] asterisk-1.2.1]# lsmod | grep ztdummyztdummy 7748  0zaptel    196612  1 ztdummyAlthough it won't load under normal conditions. I have to 'modprobe ztdummy' manually:[EMAIL PROTECTED] aster

Re: [Asterisk-Users] RE: SIP Subscriptions

On 12/20/05, Douglas Garstang <[EMAIL PROTECTED]> wrote: > You know, there isn't even a 'sip delete subscription' command available. If > there was, it might be possible to write some third party scripts which > interact via the Manager Interface to control subscriptions. Without even a > delete

[Asterisk-Users] RE: SIP Subscriptions

You know, there isn't even a 'sip delete subscription' command available. If there was, it might be possible to write some third party scripts which interact via the Manager Interface to control subscriptions. Without even a delete function though, it's pretty much impossible to do anything at a

Re: [Asterisk-Users] SPANDSP & TX RX Fax paid support wanted.

Rehan Ahmed a écrit : Hello, I need some paid help for installation of Spandsp and RX and Txfaxes for me so i can learn what is the error i am having, so far i have this problem patching file Makefile Hunk #1 succeeded at 52 (offset 2 lines). Hunk #2 FAILED at 98. 1 out of 2 hunks FAILED -

[Asterisk-Users] Re: Can you time limit access to a trunk?

David Tillman <[EMAIL PROTECTED]> writes: > I have an issue where I have an extension (that is powered down and > disconnected) > still connected to a trunk (Sipura SPA3000) na dhas been for 11 hours. > > First of all, I don't know how it got in that condition. > > Secondly, is there a way I can l

Re: [Asterisk-Users] IVR Capacity

Why not use a much simpler solution, such as finding a way to schedule different employees/offices/regions what have you to call in at different times? Or better yet don't use an IVR at all. Yeah or even better, call /THEM/ at regular intervals until they answer the survey. Use the mighty p

RE: [Asterisk-Users] SIP Subscriptions

Realtime, as stated by Digium, does not work with sip users. This isn't related to sip subscriptions. -Original Message-From: Gary Reuter [mailto:[EMAIL PROTECTED]Sent: Tuesday, December 20, 2005 3:54 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:

[Asterisk-Users] SPANDSP & TX RX Fax paid support wanted.

Hello,    I need some paid help for installation of  Spandsp and RX and Txfaxes for me so i can learn what is the error i am having, so far i have this problem patching file Makefile Hunk #1 succeeded at 52 (offset 2 lines). Hunk #2 FAILED at 98. 1 out of 2 hunks FAILED -- saving rejects to file M

Re: [Asterisk-Users] SIP Subscriptions

Everybody is entitled to their own opinion. I believe Kevin Fleming indicated that the Digium "todo" list was flexible if enough $$$ of funding were involved. Maybe that would interest you more. Darren Wiebe [EMAIL PROTECTED] Douglas Garstang wrote: I don't think the bounties are worth the

Re: [Asterisk-Users] SIP Subscriptions

Doesn't realtime allow you to manipulate users without ever having to issue a reload command?I'd expect the command 'reload' to flush any existing information from memory and use with freshly loaded information anyway, and that goes for any application, not just asterisk. On 12/20/05, Douglas Garst

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