Re: [Asterisk-Users] Connecting two phones with different codecs

>From memory, it's really down to making the right selections in sip.conf We did a large installation, with phones at the Head Office using g711 and phones at remote sites using g729. Asterisk happily transcoded for us. Which was great. PalH On Wed, 2006-02-15 at 15:37 +1100, Lisa Wolf wrote:

[Asterisk-Users] ATA186 V2.15.ms upgrade

Hi, I want have upgrade my ATA186 v2.15ms to H.323/SIP ButI dont have a cisco acount yet can some body help me with the ata18x-v2-16-030401a-1.zip file ? THX . weiming. ___ --Bandwidth and Colocation provided by Easynews.co

[Asterisk-Users] Connecting two phones with different codecs

I've got a situation here that I thought was trivial. I have two phones, and an asterisk box. The first phone knows about g723, alaw and g729, as does the second phone. sip.conf has allows for those codecs. Now, within the dialplan I make a determination which codec will be used (this is a si

Re: [Asterisk-Users] Telmex PRI line configuration problem

no cdp enable - 1) Is it possible to have CAS framing with no R2? You cannot have CAS and PRI at the same time. Thats for sure. Also CAS goes hand-in-hand with R2. CAS means Channel Associated Signalling and MFC-R2 is the Signalling ty

Re: RE : [Asterisk-Users] To connect between more than 2 asterisk server [links needed ]

--- [EMAIL PROTECTED] wrote: > Hello, > > I have an IAX2 trunk like this running well with > IAX2 and SIP users mixed at > each side. > Runing like a charm :-) > Don't forget to add username definition from this > example. > To avoid too much load for your CPUs with > transcoding, tempt to have

[Asterisk-Users] Re: dual TE410, both span 3 is broken (Josh Krueger)

On Sun, Feb 12, 2006 at 04:30:50PM -0700, [EMAIL PROTECTED] wrote: > I've seen a similar problem before. Span 3 was throwing errors for > (what seemed to be) no reason at all. After some testing it seemed > that the number of errors thrown on Span 3 had a relationship to the > temperature ins

[Asterisk-Users] asterisk t.38 pass

is there recomended source files for t.38 pass? latest cvs does not work for me. is it possible publish working src? turby ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list T

Re: [Asterisk-Users] Polycom buddy watch limit of 7

Mike Pollitt wrote: > I've got a Polycom 601 with the sidecar unit all working with extension > hints and what Polycom calls the Buddy Watch feature. I can see the state of > extensions, but there seems to be a limit of 7 that I can monitor at any one > time. Polycom is apparently not going to in

Re: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's

www.interast.com Kyle andrew matthews wrote: sorry its really http://connect.voicepulse.com/ On 2/14/06, *andrew matthews* <[EMAIL PROTECTED] > wrote: http://connect.voicepulse.net They support astrisk, with iax2 :) On 2/14/06, *Jim Robinson* < [EM

[Asterisk-Users] Firmware version 1.3.1 released for Aastra IP phones

Title: Firmware version 1.3.1 released for Aastra IP phones Aastra Telecom has released SIP v1.3.1 firmware for the Aastra range of IP phones (480i, 480iCT, 9112i and 9133i). The firmware and release notes (no updated admin and user guides yet) are available for download at: http://www.aastr

Re: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's

sorry its really http://connect.voicepulse.com/On 2/14/06, andrew matthews <[EMAIL PROTECTED] > wrote: http://connect.voicepulse.net They support astrisk, with iax2 :)On 2/14/06, Jim Robinson < [EMAIL PROTECTED]> wrote: Hi Folks,Can anyone give me some good recommendations for VoIP providrs thats

Re: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's

http://connect.voicepulse.net They support astrisk, with iax2 :)On 2/14/06, Jim Robinson <[EMAIL PROTECTED]> wrote: Hi Folks,Can anyone give me some good recommendations for VoIP providrs thatsupport Asterisk PBX's?  We're based in Georgia and I having a hard timefinding anyoneRegards,JimPS -

RE: [Asterisk-Users] Multiple AGI Issues

Hi Freddi Thanks for the reply. Neat ideas there, but a couple of issues. 1. Don't want to have to jump around between the FastAGI and the dial plan. Our plan is to have NO customer data in the dialplan, as all data will be contained within MySQL. We don't want to have to make _any_ edits to th

[Asterisk-Users] Polycom buddy watch limit of 7

Hi All -- I've got a Polycom 601 with the sidecar unit all working with extension hints and what Polycom calls the Buddy Watch feature. I can see the state of extensions, but there seems to be a limit of 7 that I can monitor at any one time. I've put in a call to my distributor (this is how Polyc

[Asterisk-Users] Adjusting frequency asterisk sends NOTIFY's to ATA's at for MWI.

I'm trying to figure out how Asterisk decides how often it will send SIP NOTIFY's to an ATA when a voicemail message is waiting for the user on the server. >From watching, it seems to be completely random. Sometimes 10 seconds apart, then 33 seconds, then 13 seconds, etc. Each time causes a "rin

Re: [Asterisk-Users] Call centre - * hang's up

On Tue, 2006-02-14 at 11:34 -0500, Time Bandit wrote: > > When agent tries to transfer a phone call (*2 - att transfer) he hangs up. > > Why? When a phone call isn't from queue then att transfer works fine. > > > > In features conf I have *1 for recording, *2 for att transfer and #1 for > > blind

Re: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls

Thanks for all your responses. The reason we would not go through a provider is that I run Asterisk phone systems, we have access to bandwidth, and I can do this myself for a fraction of the cost. Cheers ___ --Bandwidth and Colocation provided by Easynew

RE: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls

> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Peter Corlett > Sent: Tuesday, February 14, 2006 9:01 AM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Solution for 1 time blast of > 200,000 recorded calls > > > Ron Seny

[Asterisk-Users] Bug in AMP 1.10.010 in sip outbound callerid

If you define a sip peer, wheather or not you put an entry in the field OUTBOUND CID, if you dial an external extension (let's say an extension on another asterisk server, connected via IAX2 connection) the callerid received by the foreign asterisk is device : i.e device <567> If you take a look a

re:[Asterisk-Users] Multiple AGI Issues

To: "Asterisk Users Mailing List - Non-Commercial Discussion" I've got several issues with AGI/FastAGI 1. When an AGI script sends a command to Asterisk via stdin, why does Asterisk block and not return a result until the command is complete? Specifically, the dial command. If I send a

Re: [Asterisk-Users] Fax to Email with Asterisk and Lucent TNT

On Tue, 14 Feb 2006, Matthew Crocker wrote: On Feb 14, 2006, at 4:33 PM, [EMAIL PROTECTED] wrote: nvfaxdetect / nvbackgrounddetect, along with spandsp+rxfax/txfax is your answer. http://www.voip-info.org/wiki-NVFaxDetect I didn't think this was scalable to a PRI full of fax calls. I'm trying t

Re: [Asterisk-Users] Fax to Email with Asterisk and Lucent TNT

On Tue, 2006-02-14 at 17:26 -0500, Matthew Crocker wrote: > On Feb 14, 2006, at 4:33 PM, [EMAIL PROTECTED] wrote: > > > On Tue, 14 Feb 2006, Matthew Crocker wrote: > >> I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd > >> like to be able to direct an inbound fax call into my TNT,

RE: [Asterisk-Users] Podget or Similar

Speaking of script launching, how would one fire-and-forget a script in Asterisk? It seems that as it is currently configured, if the called hangs up, the script aborts. Thanks, Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell

Re: [Asterisk-Users] TDM04B/TDM2401E Card

Hello,   The cards you referenced are BRI cards. We are located in México and do not use ISDN.  Instead I could install a optional E1 interface to the D-1232 and purchase for example a TE110P card from Digium.   But this solution will cost a lot more money and I am limited with my budget. I don

Re: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls

Well, I'm not real sure on whether I like the idea or not bug Anyway, here is an app that I wrote for something similar to this. It was for notifying customers of events,etc. http://www.astpp.org/index.php?n=Misc.AutoDialOut Darren Wiebe [EMAIL PROTECTED] Ron Senykoff wrote: Hi, I'

Re: [Asterisk-Users] Traffic prioritization and 'class of service' for SIP

Ouch, Sprint wants $200 for the priviledge. I couldn't get approval for that yet until we are closer to switching over more lines to voip. Is it possible to do something equivelent or close without Sprint's help? It seems like they are implmenting the equivelent of: service-policy input IPC

Re: [Asterisk-Users] Traffic prioritization and 'class of service' for SIP

What you are doing is changing the priority of packets that you are sending to the internet, you'll have to throttle the bandwidth for incoming packets (or better yet, have sprint do it on their router).  What you are doing will help if you are getting bad calls when someine is uploading something

RE: [Asterisk-Users] Grandstream hold one way audio -URGENT

We are using the same phones in our office with firmware 1.0.1.13 and have no issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: Tuesday, February 14, 2006 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re

Re: [Asterisk-Users] Fax to Email with Asterisk and Lucent TNT

On Feb 14, 2006, at 4:33 PM, [EMAIL PROTECTED] wrote: On Tue, 14 Feb 2006, Matthew Crocker wrote: I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd like to be able to direct an inbound fax call into my TNT, have it answer the fax and send the image file over to Asterisk, or som

RE: [Asterisk-Users] Solution for 1 time blast of 200, 000 recordedcalls

> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > [EMAIL PROTECTED] > Sent: Tuesday, February 14, 2006 11:06 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Solution for 1 time blast of > 200, 000

[Asterisk-Users] Good VoIP providers that support Asterisk PBX's

Hi Folks, Can anyone give me some good recommendations for VoIP providrs that support Asterisk PBX's? We're based in Georgia and I having a hard time finding anyone Regards, Jim PS - If you could CC me in on the reply I would greatly appreciate it! jim(-A T-)linux-sp.com _

Re: [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...

Hi Giorgio: That seems like a kind of a kludge. I would rather have the program work right, than adding a work around. Dan of Littlejohnsconsulting has told me of one problem in ARI that he is fixing but I do not understand how it will fix the issue yet?? I will let you know as I find out mor

RE: [Asterisk-Users] Podget or Similar

That's not a bad way to do it, but I'm also getting headline news. I have something that works (once I like it I'll tweak the wiki with it), but wouldn't mind something better. Podcasts for weather by zip are available from 'pirateweather.com'. These allow offloading the festival-like functions

RE: [Asterisk-Users] SIP Register

First impressions telling me you want to check your phone settings. What phone are you using and what are the config settings? Mark -Original Message- From: Tomislav Parèina [mailto:[EMAIL PROTECTED] Sent: Tuesday, 14 February 2006 9:01 PM To: Asterisk Users Mailing List - Non-Commercial

RE: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls

Yes. I have customers doing that all the time. They are service provider specialize in political campages. Going back to topic, it he is only doing this one time, why doesn't just find a service provider company to do it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTE

Re: [Asterisk-Users] fax pass-through

after upgrade from 1.0.x to 1.2.x i cannot send faxes my topology: PSTN<-wct4xxp-asterisk- -sip- ata (ht496,ht488,asus vp100) - samsung sf2500 fax is there someone with this scenario? it is working? thanks (ip connectivity is good, codec alaw, 0% success) log: Feb 13 23:50:35 DEBUG[27914] ch

Re: [Asterisk-Users] Grandstream hold one way audio -URGENT

On Tue, 14 Feb 2006, Ronald Voermans wrote: At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times t

[Asterisk-Users] can't dial zap extensions?

Ok, got my last issue sorted, now another one. I can call out fine on this zap channel which is connected to a carrier access bank 1 channel bank, using asterisk 1.2 (aah2.0), I can call out, call other extensions & such.. but I cannot call into this zap extension, it always says the user is on

Re: [Asterisk-Users] Dazed and Confused

I have the same problem on the same server...But I have just 3 PCI slots and the 3 are with digium cards. One of then is a TE406P with only one link connected, so there are a lot of red alarms.I'd like to have the blue light back on my server =) .. Any one already solve this?On 11/17/05, Simone Cit

Re: [Asterisk-Users] Grandstream hold one way audio -URGENT

Yes, we have and we just got rid of them because of it. We use higher end phones like Polycom, Snom and Cisco now. On 2/14/06, Ronald Voermans <[EMAIL PROTECTED]> wrote: > > Hi all, > > At our customer site i've installed one asterisk server with 20 Grandstream > GXP2000's. Firmware 1.0.1.9. Wh

Re: [Asterisk-Users] Fax to Email with Asterisk and Lucent TNT

On Tue, 14 Feb 2006, Matthew Crocker wrote: I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd like to be able to direct an inbound fax call into my TNT, have it answer the fax and send the image file over to Asterisk, or some other system to deliver to an e-mail address(s). I'm no

RE: [Asterisk-Users] Fax to Email with Asterisk and Lucent TNT

We've setup automatic printing of faxes from asterisk (spool to a net queue or direct to local printer), use the sendmail capability of emailing to an executable (alias). Why send an image over to asterisk from the TNT? MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECT

[Asterisk-Users] RE: ZAP extension, DTMF?

Please ignore my last query about DTMF on ZAP, turned out to be an echo can issue.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/

Re: [Asterisk-Users] Fax to Email with Asterisk and Lucent TNT

On Tue, 2006-02-14 at 14:53 -0500, Matthew Crocker wrote: > Hello, > > I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd like > to be able to direct an inbound fax call into my TNT, have it answer > the fax and send the image file over to Asterisk, or some other > system to de

Re: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls

Here in Arizona I have gotten recorded calls from politicians. There are apps to allow for this feature. If you want to know more contact me offline. Kyle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUB

Re: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls

> I just did a quick office poll and everyone agreed if a party candidate > did this to them, they would vote for the candidate's opponent. The office > is rarely unanimous in political matters so this was a pretty interesting > result to me. > > I'm pretty sure the feeling is universal. > > Like I

[Asterisk-Users] Changes to sip.conf in 1.2.x ?

Is there something significantly different in 1.2.x sip.conf that would prevent clients from registering with the server? We installed 1.2.3 and copied over sip.conf from a production 1.0.9 box and clients can not register. Asterisk just replies with "unauthorized". Here is a sample from sip.con

Re: [Asterisk-Users] Softphone and 911

Softphone or hard phone doesnt matter if the service provider has the right connections to provide E911 service. We are setting up E911 compliance right now with our service. Its not as easy as just updating the address, it take time, its not instant. Kyle Matt wrote: Greetings to all, Can a

[Asterisk-Users] Asterisk and Snom 360

Is anyone using the SNOM 360 as a reception console with Asterisk? We are trying to have the ability to view whether an extension is on or off hook, or ringing with the Snom, which seems to work fine. The issue is that we are having difficulty picking up calls and transferring. Anyone have exp

[Asterisk-Users] Not passing CALLER id on in follow me script

Hello People,     I was wondering if you could take a look at this script,   exten => 505,1,dial(iax2/6311${EXTEN},t,25) exten => 505,2,playback(pls-wait-connect-call) exten => 505,3,set(NewCaller=${CALLERID(num)}) exten => 505,4,Set(CALLERID(num)=0${CALLERID(num)}) exten =>

Re: [Asterisk-Users] Instant Messaging: with SIP or XMPP

I use Wildfire with the asterisk IM plugin, and it seems to work really well. I'm running Trillian pro (Trillian basic does not support Jabber, it's an added cost option ~$25), and our clients are running Exodus. I do know GAIM doesn't show the status, only that the user is unavailable. Wild

[Asterisk-Users] ZAP extension, DTMF?

hey all, trying to get a zap extension to work & I can dial out normally with it, but if I try to access any of the features (i.e. *97 for voicemail) the zap channel doesn't hear it, and all i get is dialtone. Is there a dialplan setting or something to make the zap channels recognize keys like

RE: [Asterisk-Users] Podget or Similar

Bob, why not do an adaptation on the [EMAIL PROTECTED] weather service? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Bo

[Asterisk-Users] Grandstream hold one way audio -URGENT

Hi all,   At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes,

[Asterisk-Users] Fax to Email with Asterisk and Lucent TNT

Hello, I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd like to be able to direct an inbound fax call into my TNT, have it answer the fax and send the image file over to Asterisk, or some other system to deliver to an e-mail address(s). I'm not sure if I need Asterisk to

Re: [Asterisk-Users] TE411P Really Bad Echo

On Feb 12, 2006, at 6:25 PM, Mike Pollitt wrote: Hi Rob –   Is it possible to disable the onboard echo canceller so that one may try the software cancellers instead?   I have the TE110P and am experiencing the same bad echo problems that I can’t seem to effect by fiddling with the echo canc

[Asterisk-Users] How to create latency on purpose

Hi All, I have a Digium card in my Asterisk server configured as pri_net and I want to introduce latency on it in order to simulate PSTN conditions and test some echo canceller hardware. Is it possible to purposefully introduce latency and echo in a controlled fashion in order to do so? Thanks...

Re: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls

On Tue, 14 Feb 2006, Ron Senykoff wrote: It's a basic GOTV (Get Out The Vote) drive, with just a short message to encourage people to come out to the polls. It has nothing to do with asking for any money, etc. Just a short message to people who belong to the party from their candidate. I just d

[Asterisk-Users] Multiple AGI Issues

I've got several issues with AGI/FastAGI 1. When an AGI script sends a command to Asterisk via stdin, why does Asterisk block and not return a result until the command is complete? Specifically, the dial command. If I send a Dial command to Asterisk, I don't get a return result until AFTER the

Re: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls

On Tue, 14 Feb 2006, Ron Senykoff wrote: I'm helping out with a political campaign and would like to use asterisk to blast out about 200,000 calls with a short message from the candidate. there is a special place in hell reserved for people who do this. my advice: don't. -Dan

[Asterisk-Users] asterisk and S.E.R.

Hi, I have some questions :   If a client connected to a S.E.R. use as codecs only G729, and I want to call and give him a message in gsm or wav format using the manager API from asterisk server? This will work directly or it's necesary a codec converter? My asterisk has the codec g729 as we

[Asterisk-Users] Softphone and 911

Greetings to all, Can anyone think of a reason that a Softphone would not be compatible with the F.C.C's order for E911? If the user is able to update their address when they move their laptop, etc. ___ --Bandwidth and Colocation provided by Easynews.co

[Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US?

We are looking to lose the TDM400P in favor of an ISDN-BRI solution. This should get rid of static on the line (at least any static generated by our half of the circuit), right? I am a total virgin to ISDN. I understand that we need two BRI circuits to provide four voice channels, and that the har

RE: [Asterisk-Users] Instant Messaging: with SIP or XMPP

Have you looked at Wildfire (was Jive Messenger) with the Asterisk-IM plugin? It seems to work fairly well in my experience. I have it running here at home and also on one client's network. The XMPP client they provide (Spark) is a bit primitive, but something like Trillian also supports the requir

Re: [Asterisk-Users] Codec issue with my iaxy

Wilson Pickett wrote: I just bought a new IAXy box and am only achieving one way calling. Both iax.conf and the IAXy support ulaw and gsm. When I try to call, Does the IAXy now support anything but ulaw or alaw? The original one didn't. Dont know. All i know is that i had ulaw enable

RE: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls

> I'm helping out with a political campaign and would like to use > asterisk to blast out about 200,000 calls with a short > message from the candidate. I can't speak for anyone else, but I'd find it very difficult ethically to be involved in this, even assuming it's legal. Check very carefully

RE: [Asterisk-Users] Skilled API consultant required - preferablywith Salesforce.com intergration

Because it's a new service without much traction and I need something now and also because this will be a fairly large project and as such I don't want to use a third party service to collect a 'fee'. Dean > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PR

RE: [Asterisk-Users] odd 'digital' sound artifacts [solved]

Apparently I didn't do the first test correctly where I said I made sure I had only two cards, each on their own IRQ and still got artifacts. When I repeated the test today, with both cards on their own IRQ, I got no artifacts at all, after shuffeling the cards around for a bit I was able to get ea

Re: [Asterisk-Users] Traffic prioritization and 'class of service' for SIP

David Choo wrote: Hi, Consider doing rate limiting / bandwidth reservation. It worked heaps of wonders for mine! That's good to hear. BTW- Am I doing this right? Here are the relevent chunks of my config on my router: ! ! class-map Platinum match access-group 101 ! ! policy-map IP

RE: [Asterisk-Users] Dial command to connect two channelsand bypassasterisk server

Why not? If the client that's behind the NAT is able accept media directly from Asterisk (which is in the public network), it should be able accept media directly from another client. Otherwise, the whole STUN scheme is not going to work. -Original Message- From: [EMAIL PROTECTED] [mail

Re: [Asterisk-Users] Rough Two Days

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 14 February 2006 10:56, Sean Cook wrote: >> we had originally purchased a tdm2400 with echo cancellation but >> couldn't fit it in the chassis. >> >> Spent 3 - 4 hours tracking down the source of the echo to no >> avail, enabled mark2 echo

RE: [Asterisk-Users] TAPI Recommendations

Bob, There are issues with AstTapi and Windows XP. I suggest you go to the Asttapi bug list (on sourceforge.com) and check for patches etc., I have submitted a couple of bug reports with no response to date. I am in the process (with some success) of debugging the issue myself, and can send you a

Re: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls

> Ron Senykoff <[EMAIL PROTECTED]> wrote: > > I'm helping out with a political campaign and would like to use asterisk > > to blast out about 200,000 calls with a short message from the candidate. > > Can you tell me which party this is for, so I can ensure I never vote for > them? It's a basic GO

Re: [Asterisk-Users] Traffic prioritization and 'class of service' for SIP

Rick Smith wrote: Phil; What link ? Your question is a bit vauge, but here are some relevent urls: Sprint CoS request form (a 2 pager, with some great links to a guidelines doc and faq): http://www.sprintlink.net/maint/cos_template.cgi QoS: http://www.voip-info.org/wiki/view/QoS Phil

Re: [Asterisk-Users] Dial command to connect two channels and bypassasterisk server

Moises Silva wrote: > Unless you use SIP ALG (Application Layer Gateway) like the module in > netfilter to set the expectations? correct? If that exists on every NAT firewall in the path to both clients, yes. ___ --Bandwidth and Colocation provided by Ea

RE: [Asterisk-Users] Nat, SIP, Realtime problem

Just wanted to also say this does not happen to all users behind a NAT box on RR or DSL line just a few. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Tuesday, February 14, 2006 11:19 AM To: Asterisk Users Mailing List - Non-Com

RE: [Asterisk-Users] Nat, SIP, Realtime problem

I'm using realtime caching. Here is my sip.conf file [general] callerid=unavailable context=default allowguest=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes nat=yes canreinvite=no rtcachefriends=yes allow=ulaw allow=g729 All other information about the sip clint is keep in the db Thanks again!

[Asterisk-Users] Instant Messaging: with SIP or XMPP

well, My fulltime job is to add (i.e code if have to) some instant msging to our asterisk pbx. I have no idea, where to start. i want contribute write code, test etc. pls advice. there is SIP extension method called "MESSAGE" which get IM done. Does Asterisk support both SIMPLE and XMPP. tha

Re: [Asterisk-Users] Skilled API consultant required - preferably with Salesforce.com intergration

why not try www.asteriskhelpdesk.com (No, I am not affiliated with them in any way) --- Dean Collins <[EMAIL PROTECTED]> wrote: > Hi all, > > > > I was just on the phone with a B2C company in > Australia who are looking > to integrate an Asterisk solution with their > Salesforce.com CRM > pl

[Asterisk-Users] echo problem

I have installed Asterisk and when I hangup the zap channel Asterisk show this message: Feb 13 17:45:49 WARNING[1748]: chan_zap.c:6511 handle_init_event: Detected alarm on channel 4: Red AlarmFeb 13 17:45:49 WARNING[1748]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation

Re: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls

Voipjet should be fine however dont have all the calls go out at once. test your system first and see how many concurent calls you can have at once without loosing voice quality. I would also reccomend getting a dedicated server to do the calls as apposed to buying the equipment if you are doing a

[Asterisk-Users] [help] warning 4246

hi all, I have a problem with @ 1.2.4 on debian kernel 2.6.8-2-386.: -- Executing Dial("SIP/2003-bbae", "zap/2/03460816149|30|t") in new stack Feb 14 17:25:25 WARNING[4246]: channel.c:2535 ast_request: No channel type registered for 'zap' Feb 14 17:25:25 NOTICE[4246]: app_dial.c:1011 dial_exec_

[Asterisk-Users] Bristuff-0.3.0-PRE-1l and TDM400 with fxo ports

Hi Klaus, I have a problem with bristuff and analogue phones off FXO ports on a Digium TDM400 card. First I ran just the TDM400 cards with the phones on asterisk-1.2.4 and all OK. Then I changed to bristuff-0.3.0-PRE-1l (your latest version ) and then I ran into a problem, The analogue ph

[Asterisk-Users] Podget or Similar

While I'm thinking about it, is anyone else out there using podget or something similar to do news/weather playback? It's a neat idea, and I'd like to showcase it as a feature that the old Nortel just didn't even come close to doing... Thanks, Bob McDowell __

Re: [Asterisk-Users] Dial command to connect two channels and bypassasterisk server

Unless you use SIP ALG (Application Layer Gateway) like the module in netfilter to set the expectations? correct? RegardsOn 2/14/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: Wai Wu wrote:> If Asterisk is in the public network, it will work. The problem is when Asterisk is behind NAT and one of

Re: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls

Ron Senykoff <[EMAIL PROTECTED]> wrote: > I'm helping out with a political campaign and would like to use asterisk > to blast out about 200,000 calls with a short message from the candidate. Can you tell me which party this is for, so I can ensure I never vote for them? -- PGP key ID E85DC776 -

RE: [Asterisk-Users] consult about Digium Card

I am From Nicaragua. Regards From: "Alexander Lopez" <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: RE: [Asterisk-Users] consult about Digium Card Date: Tue, 14 Feb 2006 10:35:54 -0

RE: [Asterisk-Users] consult about Digium Card

I am From Nicaragua. Regards From: "Alexander Lopez" <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: RE: [Asterisk-Users] consult about Digium Card Date: Tue, 14 Feb 2006 10:35:54 -0

RE: [Asterisk-Users] consult about Digium Card

I am From Nicaragua. Regards From: "Alexander Lopez" <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: RE: [Asterisk-Users] consult about Digium Card Date: Tue, 14 Feb 2006 10:35:54 -0

Re: [Asterisk-Users] Rough Two Days

On Tuesday 14 February 2006 10:56, Sean Cook wrote: > we had originally purchased a tdm2400 with echo cancellation but > couldn't fit it in the chassis. > > Spent 3 - 4 hours tracking down the source of the echo to no avail, > enabled mark2 echo cancel and aggressive echo cancel to get the system >

Re: [Asterisk-Users] Use one sip account for multiple sipura

Reli Loin wrote: hello, I have one account i need using multiple sipura ata, for my account. it's possible in asterisk. No. Generally you never need multiple devices to use the same account information. This has been talked about in the archives. Personally I use the MAC address of the

Re: [Asterisk-Users] Call centre - * hang's up

> When agent tries to transfer a phone call (*2 - att transfer) he hangs up. > Why? When a phone call isn't from queue then att transfer works fine. > > In features conf I have *1 for recording, *2 for att transfer and #1 for > blind. In queue blind transfer works. For disconnect I have #0. > > I

Re: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls

How much time and hardware do you have to do this? VICIDIAL is capable of doing this(and we have done that with VICIDIAL in the past) with hundreds of concurrent calls if your servers can handle it, but it takes a few hours to setup and if you are only doing this once it may not be worth it. MATT

Re: [Asterisk-Users] ChanIsAvail

I was gonna say use a queue of sorts, throw the devices into the queue and tell it to ring all. I haven't played with it, but I would assume that if a line's in use, it won't ring that person. Aaron Joseph Tanner wrote: Perhaps I'm missing something here, but why not just have asterisk dial

Re: [Asterisk-Users] Nat, SIP, Realtime problem

Hall, Eric M. wrote: > Asterisk CVS-HEAD dated 2005-08-18 > WhitBox Linux respin 2 > mysql Ver 11.18 Distrib 3.23.58 > Cisco 7960G > > We are using the real-time drivers for sip and everything is working > great. > They have a few employees that use the phones from home on a RR or DSL > line. >

RE: [Asterisk-Users] odd 'digital' sound artifacts [1 card = no artifacts]

Ok, after 10+ minutes of mindnumbing hold music ;) no artifacts with only one TDM card installed, going to test with two cards on their own IRQ in a minute again.. Feb 14 09:59:31 [kernel] Zaptel Version: SVN-trunk-r941M Echo Canceller: MG2 Feb 14 09:59:31 [kernel] ACPI: PCI Interrupt Link [APC2]

RE: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls

Yeh my advice is don't, or if you must make sure you clean your dialling list first with the DNC list. Yes I know that political messages don't need to obey the DNC list but they should. This is why I don't donate t most political campaigns. Dean > -Original Message- > From: [EMAIL P

Re: [Asterisk-Users] Dial command to connect two channels and bypassasterisk server

Wai Wu wrote: > If Asterisk is in the public network, it will work. The problem is when > Asterisk is behind NAT and one of the client is also behind the same NAT. No, it won't. If one of the clients is behind a NAT firewall, you cannot tell that firewall to start accepting media directly from th

Re: [Asterisk-Users] about g729 license

Got it.. so, in this case, I am using 36 licenses, right?? Thank you very much Dov - Original Message - From: "Kevin P. Fleming" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, February 14, 2006 1:25 PM Subject: Re: [Asterisk-Users] about

Re: [Asterisk-Users] TE411P Really Bad Echo

On 2/13/06, Mike Pollitt <[EMAIL PROTECTED]> wrote: Hi Rob –Is it possible to disable the onboard echo canceller so that one may try the software cancellers instead? I have the TE110P and am experiencing the same bad echo problems that I can't seem to effect by fiddling

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