Re: [asterisk-users] VOICEMAIL : I've tried a lot but mailing through Asterisk is just not working...

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 jonas kellens wrote: | David, | | what is your SMTP-client then ? | | Did you change the mailcommand 'mailcmd' in voicemail.conf ?? Or is it | still /usr/sbin/sendmail ?? I don't have mailcmd in voicemail.conf, I was under the imp

Re: [asterisk-users] VOICEMAIL : I've tried a lot but mailing through Asterisk is just not working...

' 'phonetic/z_p' european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM [default] => ,David Abbott,x...@.net Thats all I have in there, asterisk will use my SMTP client without me doing anything. I am using asterisk 1.4 - -david - -- Powere

Re: [asterisk-users] Problems receiving some faxes in T.38

On Wed, May 20, 2009 at 6:58 AM, Santiago Gimeno wrote: > We have been working with the ReceiveFax application for some weeks now in > order to receive faxes in T.38 and it works fairly well, but there are some > faxes that for some reason we are not able to receive correctly. > > The asterisk ver

Re: [asterisk-users] Hang at 5:34 pm EST

It should be Tue May 19 18:39:18 EDT 2009 On Wed, May 20, 2009 at 3:56 AM, David @ULC wrote: > > Some at 5:34 pm EST DAILY, all my call get disconnect. > I tried RE-INSTALLATION, I tried Reinstallation on a virgin HDD, but its > same. > > I tried changing VOIP provide

Re: [asterisk-users] Hang at 5:34 pm EST

Is it that server disconnect all calls at 1am EDT as things are configured in that way ? On Wed, May 20, 2009 at 4:08 AM, David @ULC wrote: > > Current time is > Wed May 20 02:05:41 EDT 2009 on the server > > > > On Wed, May 20, 2009 at 3:56 AM, David @ULC wrote: > &

Re: [asterisk-users] Hang at 5:34 pm EST

Current time is Wed May 20 02:05:41 EDT 2009 on the server On Wed, May 20, 2009 at 3:56 AM, David @ULC wrote: > > Some at 5:34 pm EST DAILY, all my call get disconnect. > I tried RE-INSTALLATION, I tried Reinstallation on a virgin HDD, but its > same. > > I tried changi

[asterisk-users] Hang at 5:34 pm EST

Some at 5:34 pm EST DAILY, all my call get disconnect. I tried RE-INSTALLATION, I tried Reinstallation on a virgin HDD, but its same. I tried changing VOIP provider I tried changing Internet Provider..But no help.. What could be the reason ? Here are my enties of crontab : ### recording mixi

Re: [asterisk-users] Ghost ??

I am usIng VOIP. On Tue, May 19, 2009 at 11:46 PM, David @ULC wrote: > We are using asterisk and sometime when our guys are on call , they hear > some voice of person and amazingly that person is NOT from our center. > Any one faced this kind

Re: [asterisk-users] Ghost ??

No such pattern. It happens ad hoc. On Tue, May 19, 2009 at 11:46 PM, David @ULC wrote: > We are using asterisk and sometime when our guys are on call , they hear > some voice of person and amazingly that person is NOT from our center. > Any one faced this kind

[asterisk-users] Ghost ??

We are using asterisk and sometime when our guys are on call , they hear some voice of person and amazingly that person is NOT from our center. Any one faced this kind of thing ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- a

Re: [asterisk-users] Add Monitor application to call suppresses audio

On Mon, May 18, 2009 at 5:16 PM, Barry L. Kline wrote: >> So you say call 1 with recording made a file, and the call connected with >> voice. >> And call2 with recording made a file, but the customer didn't hear the voice? > > Yes.  In this case I'm using an outside Asterisk server to dial back i

Re: [asterisk-users] Manager API in PHP

On Mon, May 18, 2009 at 10:02 AM, Scott Gifford wrote: > Olivier writes: > > [...] > >> What is your opinion regarding PHP AMI API and Asterisk 1.6.1 ? >> I'm referring here to http://code.google.com/p/asterisk-php-api/. > > In my experience, asterisk-php-api works OK, but it's a bit slow.  It >

Re: [asterisk-users] Some direction for creating sound files for Asterisk

On Mon, May 18, 2009 at 1:48 PM, Jonathan Moore wrote: > First, my real problem is I don't totally understand what I'm doing... > which is why I'm making this post. > > What I'm trying to do is record various prompts for use with my > asterisk system.  I started out by just using some of the vario

[asterisk-users] Calls Declined

All my calls are getting DECLINED when I am trying from xlite : CLI shows : May 18 00:00:32 WARNING[4617]: channel.c:2781 ast_channel_make_compatible: No pa th to translate from SIP/cc101-b790c1d8(4) to SIP/sip19-090e87d8(256) May 18 00:00:32 WARNING[4617]: app_dial.c:1628 dial_exec_full:

Re: [asterisk-users] Capture "Server" header in SIP reply.

On Sun, May 17, 2009 at 9:04 AM, Chris Maciejewski wrote: > Unfortunately SIP_HEADER(FROM) is not an option for me. > > What I want to do is record in CDRs "User-Agent" header of calling > party (this can be easily done with ${CHANNEL(useragent)}), and SIP > "Server" header of called party (from 2

Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router

On Sat, May 16, 2009 at 10:22 AM, Timothy Smith wrote: > I have finally managed to get voice working. I both parties can hear > each other. The problem was nating. Our network is fairly big and > these machines are atleast 2 switches from each other. I just enabled > it (nat=route or nat=yes) and

Re: [asterisk-users] Capture "Server" header in SIP reply.

On Sun, May 17, 2009 at 6:43 AM, Chris Maciejewski wrote: > I am trying to capture "Server" header in a 200 OK reply message. > My idea was to use Dail(SIP/u...@domain,30,M(GetOtherPartyInfo)), > and inside of GetOtherPartyInfo macro use SIP_HEADER function. > unfortunately the above doesn't seem

[asterisk-users] Agent-Login/out in 1.6

eQueueMember(support,Local/queue${agentnumb...@ansqueue) exten => *20,n,Playback(agent-loggedoff) exten => *20,n,Verbose(2,Agent ${AgentNumber} removed) exten => *20,n,Hangup()" -- _ Mr. David Anthony O'Reilly, B.Sc Comp (Hons) M.Sc MOB Po

[asterisk-users] Agent-Login/out in 1.6

g the agentcallbacklogin command did.| I totally agree, I have never seen any example that makes it work. If somebody shows me how to do it without using Voicemail I will let you know. Thanks David -- _ Mr. David Anthony O'Reilly, B.Sc Comp (Hons)

Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router

On Sat, May 16, 2009 at 7:46 AM, Timothy Smith wrote: > I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9), > and also a dialpeer to forward on the router to forward calls to my > asterisk. It works properly but the problem is there is NO AUDIO! I > have tried to change codec but

[asterisk-users] Logging In / Out Agents on Asterisk 6 ???

ther calls outside of the queue cannot come in as all are blocked so not a great login/logout function. If anybody could help provide a sample of how they did it on 6 I would be extremely grateful and will create a WIKI page on it for others as I have been very unlucky trying to work this out. Many th

Re: [asterisk-users] help a bald guy

On Fri, May 15, 2009 at 11:17 AM, Danny Nicholas wrote: > AFAIK we are on PRI with no alarms. I'm thinking the common denominator with your arrangement is that call progress info isn't getting relayed properly, and / or there are line signaling issues. To me that means a problem with your DAHDI/

Re: [asterisk-users] help a bald guy

On Fri, May 15, 2009 at 10:33 AM, Danny Nicholas wrote: > It Mostly works for me;  Due to internal security concerns, we use POTS with > TDM400 and TDM410P cards for incoming/outgoing service.  Here is the > short-list of my most common concerns: > 1.  Some numbers (especially AT&T conferences) ne

Re: [asterisk-users] [asterisk-dev] Fax t38 capability

On Fri, May 15, 2009 at 5:06 AM, Khaled W. Chehab wrote: > Dears I installed digium fax and followed the instruction at > http://downloads.digium.com/pub/telephony/fax/README,And as  you can  see > above that t38 is loaded Step 1: This is an asterisk-users question. Step 2: This is not an asteris

Re: [asterisk-users] Add Monitor application to call suppresses audio

tempt at getting a recording.   It, and any > subsequent call, fails miserably. > > I'm open for any suggestions.  Thanks very much David. > > Barry In all cases, your dial argument to limit the call to 15 seconds seems to be working, and the calls are definitely getting hungup at 1

Re: [asterisk-users] Add Monitor application to call suppresses audio

On Wed, May 13, 2009 at 11:53 AM, Barry L. Kline wrote: > If I insert a Monitor() prior to dialing the outbound call, I get no > audio in the recording and the caller hears no audio.   Occasionally it > works (perhaps 1 out of 5 times) but most of the time the caller can't > hear the callee, and v

[asterisk-users] #-all.gsm

Recently I noted few recording link with # sign on it. like 223345#-all.gsm and all those voice files are NOT available for download. I tried changing the file name in the mysql db and removed # but still its not available. What could be the reason for # and why its NOT available for downlo

Re: [asterisk-users] Voicemail and remote directory with SSHFS

Tunnel samba or nfs through ssh, rather than using sshfs, then mount using once of those more ubiquitous standards. -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock Sent: Wednesday, May 13, 2009 1:09 PM To: Asterisk

Re: [asterisk-users] Asterisk 1.6 T.38 generation towards a Ciscovoice router

On Wed, May 13, 2009 at 9:21 AM, Jon Schøpzinsky wrote: > I used wireshark to debug the problem, and I can see that the cisco equipment > is correctly sending t.38 packets to asterisk, and the whole re-invite > process is successful. > The problem is, that Asterisk discards the t.38 packets with

Re: [asterisk-users] Proxying from one server to another

Redirect traffic with iptables like this: Host ~# iptables -t nat -I PREROUTING -d OLD_PUBLIC_IP -j DNAT --to NEW_PUBLIC_IP I'm not sure if this will work for SIP. You may need the proxy to change info in the sip messages between server and client. --Dave From: asterisk-users-boun...@lists.d

Re: [asterisk-users] Asterisk 1.6 T.38 generation towards a Cisco voice router

On Wed, May 13, 2009 at 3:30 AM, Jon Schøpzinsky wrote: > We are having some problems using t.38 together with a Cisco voice router at > one of our providers end. > > We are using the new digium asterisk fax module to generate the fax, and > when we use together with our internal Audiocodes Median

Re: [asterisk-users] Free Fax for asterisk

On Wed, May 13, 2009 at 7:39 AM, Markus Weiler wrote: > I wasn`t very conviced about Spandsp, after trying several versions it > worked, but not well. spandsp has been revised to the point that it's now at 0.0.6pre11, released this month. I've had quite the opposite experience, that newer version

Re: [asterisk-users] Free Fax for asterisk

On Wed, May 13, 2009 at 3:43 AM, Markus Weiler wrote: > I installed Digiums Free Fax for Asterisk and found out, that it > automatically retries failed faxes, is there a way to stop that? You already claimed that this isn't actually the case. I will tell you that the hardware fax appliances I hav

Re: [asterisk-users] Free Fax for asterisk

Hi All, Sorry to hijack this post but I am confused. What is the advantage of using this "Digium Fax For Asterisk" product when you can use Asterisks' 1.6.x module app_fax or Asterisks' 1.4.x agx-ast-addons with the app_txfax and app_rxfax modules? Regards David. -

Re: [asterisk-users] Is anyone keeping up with the versions?

On Tue, May 12, 2009 at 2:31 PM, Thermal Wetland wrote: > We are still using 1.4 and were going to start testing with 1.6.0, but then > 1.6.1 was released and now 1.6.2 is already in beta 2. > > That seems like a lot of independent releases to maintain.  I read about all > the regressions ans hurr

Re: [asterisk-users] DTMF received twice

hi there is a file wich describe the dtmf duration it depends on every country. you should look for that file and be sure you put you are on france it works for inband but look at it... Davod 2009/5/11 Administrator TOOTAI > David fire a écrit : > > out there is a file to change

Re: [asterisk-users] VoIP over satellite internet

Of course, that's assuming your satellite is in geosynchronous orbit. If It's in LEO... Singer, You are of course correct, low earth orbit will have lower latency. I was assuming that this user would be using a stationary link on the ground, not a portable sat phone or an aimable dish to make

Re: [asterisk-users] VoIP over satellite internet

...routing via satellite adds about a quarter second of latency to the path. Is that too much? Eric, I believe that you are mistaken. Routing via satellite adds about a quarter second of latency PER TRIP from earth to orbit. This is simply due to the distance a satellite is from the ground

Re: [asterisk-users] DTMF received twice

out there is a file to change the dtmf duration where are you? or from where is your cellphone? from other phones like lkand lines it works well? David 2009/5/11 Administrator TOOTAI > Hi all, > > I run an Asterisk 1.4.24.1 and face problem with DTMF. When calling from > my mobile

Re: [asterisk-users] Building a System.

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 John F. Ervin wrote: | So, people have recommended building a system from scratch, start with a | CentOS base and installing asterisk and all of the other utilities. | I've only used Trixbox for my business system. I'm wondering what | surprises I'd r

Re: [asterisk-users] Building a System.

ter-install Regards David. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John F. Ervin Sent: Monday, 11 May 2009 1:22 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Building a System. So, people have recommended

Re: [asterisk-users] How to get meetme participants in dialplan?

On Fri, May 8, 2009 at 4:54 PM, Steve Edwards wrote: > On Fri, 8 May 2009, David Backeberg wrote: >> You need a way to keep state. I use a database and AGI for that purpose. > > I thought about keeping state in the db, but then I'd have to run the AGI > when they leave the

Re: [asterisk-users] Record all calls

On Fri, May 8, 2009 at 5:31 PM, Michelle Dupuis wrote: > I'd like to setup a single extension for which all INBOUND and OUTBOUND > calls are recorded to a wav file.  I took a look at the wiki: > > http://www.voip-info.org/wiki/view/Asterisk+record+calls > > but it's not too helpful.  Can someone s

Re: [asterisk-users] AGI - Ways to create a call

>> -Original Message- >> From: tiagodura...@gmail.com >> Sent: Fri, 1 May 2009 11:02:58 -0400 >> To: asterisk-users@lists.digium.com >> Subject: [asterisk-users] AGI - Ways to create a call On Fri, May 1, 2009 at 11:18 AM, Jimmy Godbout wrote: >> Well, my question is: do you guys have any

Re: [asterisk-users] 1.6.1 app_fax: WARNING T.30 ECM carrier not found ??

On Mon, May 4, 2009 at 10:52 PM, sean darcy wrote: > Receiving a fax with 1.6.1: > >   == Spawn extension (incoming-pstn-line, fax, 1) exited non-zero on > 'DAHDI/4-1' >     -- Executing [...@incoming-pstn-line:1] NoOp("DAHDI/4-1", "Fax > Detected") in new stack >     -- Executing [...@incoming-ps

Re: [asterisk-users] precision of wait dialplan application

On Wed, May 6, 2009 at 8:17 AM, Johann Steinwendtner wrote: > But it seems the Wait(60) lasts longer than 60 seconds: > >     -- Executing [...@from_meridian:1] NoOp("DAHDI/29-1", "Test > wait") in new stack >     -- Executing [...@from_meridian:2] Answer("DAHDI/29-1", "") in > n

Re: [asterisk-users] How to get meetme participants in dialplan?

On Thu, May 7, 2009 at 11:44 AM, Steve Edwards wrote: > The meetmeadmin() dialplan function lets you specify a user to mute, > un-mute or kick. But how do you get a list of users in your dialplan? You need a way to keep state. I use a database and AGI for that purpose. > When a user joins a conf

Re: [asterisk-users] QoS & VPN

On Thu, May 7, 2009 at 3:54 PM, Brent Davidson wrote: > I've got multiple satellite office all linked back to the main office > via VPN.  Each office has their own asterisk server which registers back > to the main office's Asterisk server.  Each office also has a 1Mb > downstream / 384k - 768k up

Re: [asterisk-users] Cisco 7940 phones become unreachable over VPN after a time

On Wed, May 6, 2009 at 12:48 PM, i...@comtek.co.uk wrote: > Hi. > > I have a working internal Asterisk setup with 35 phones. Around 5-10 of > these phones are physically located in a remote office via a VPN. I am There are a number of other reasons you want a remote phone server at that other loc

Re: [asterisk-users] Cisco 7940 phones become unreachable over VPN after a time

On Wed, May 6, 2009 at 12:48 PM, i...@comtek.co.uk wrote: > Can anybody provide any suggestions to help debug this? If I'm unable to > isolate/resolve the problem then its likely we'll have to drop the > Asterisk solution and I've already grown rather attached to it. I have a number of ideas of w

Re: [asterisk-users] shut down a single PRI on a running Asterisk system?

On Wed, May 6, 2009 at 1:04 PM, James Van Vleet wrote: > We are wondering if there is any way to shut down a single PRI without > having to down Asterisk and/or interrupt other running PRI circuits. Somebody who knows Zaptel better could tell you whether this is a bad idea, but my first thought w

[asterisk-users] ConfBridge versus MeetMe

Formerly on a thread called [asterisk-dev] Where to find the code of application Bridge On Wed, May 6, 2009 at 7:38 AM, Tzafrir Cohen wrote: >> Can someone please tell me in which file the code for the application to >> be found? I was not able to find a file named app_bridge.c in the folder >> a

[asterisk-users] Bridge() and Goto() and dialplan contexts, oh my!

I may or may not be experiencing the behavior described in: http://bugs.digium.com/view.php?id=14241 I'm using asterisk-1.6.0.6, Bridge(), and I'm having a hangup context executed when the caller is still on the line. These channels are all SIP. I want a group of expert callers who can dial

Re: [asterisk-users] Cisco phone - can Call manager reflash automatically if we test in Asterisk with SIP?

CallManager which includes what firmware to load. It should then reload the SCCP firmware (if you are not using SIP) and reboot back to how it was. All of this is assuming that you have a standard CallManager environment of course. -Jonathan On Mon, May 4, 2009 at 3:14 PM, David Shauger wrote

Re: [asterisk-users] Cisco phone - can Call manager reflash automatically if we test in Asterisk with SIP?

David, Will it happen automatically when you reconnect it to Cisco Call Manager or does it require additional steps? Thanks! On May 4, 2009, at 4:14 PM, David Gibbons wrote: Yes, you can flash them back and forth as you require. From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Cisco phone - can Call manager reflash automatically if we test in Asterisk with SIP?

Yes, you can flash them back and forth as you require. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Shauger Sent: Monday, May 04, 2009 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users

[asterisk-users] Cisco phone - can Call manager reflash automatically if we test in Asterisk with SIP?

would be the case, but would like to be sure. Thanks! David Shauger Vice President Sollos Technology Solutions 678-317-9444 - voice 404-886-7603 - cell 772-679-5830 - fax d...@sollos.com http://www.sollos.com/ This

Re: [asterisk-users] advice on OrderlyStats (or other cc software)

On Mon, May 04, 2009 at 10:04:53PM +1000, Rob Hillis wrote: > Louis-David Mitterrand wrote: > > Hi, > > > > Is anyone here using OrderlyStats with asterisk in a call center > > setting? If so what what is your experience with it? Is that software > > really free

[asterisk-users] advice on OrderlyStats (or other cc software)

Hi, Is anyone here using OrderlyStats with asterisk in a call center setting? If so what what is your experience with it? Is that software really free for asterisk users? Or is there a better option out there? Thanks, ___ -- Bandwidth and Colocation P

Re: [asterisk-users] LoadAvg , Codec and Bandwidth Utilisation

Check this... http://prodsurvey.webng.com/top.jpg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] 2 phone extensions on a single conference room

How to solve If I see "2 phone extensions on a single conference room" which is causing the conference issue ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] LoadAvg , Codec and Bandwidth Utilisation

[root]# top top - 20:19:40 up 53 min, 1 user, load average: 9.54, 7.85, 6.44 Tasks: 224 total, 1 running, 223 sleeping, 0 stopped, 0 zombie Cpu(s): 7.5%us, 3.8%sy, 0.0%ni, 28.6%id, 59.1%wa, 0.2%hi, 0.8%si, 0.0%st PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 2

Re: [asterisk-users] LoadAvg , Codec and Bandwidth Utilisation

Box is working but sometimes cross connection issues Looks like my box is saving too many voice mails and due to which LoadAvg is going high. How to disable it ? Also, I am recording all calls.. 2009/5/1 David @ULC > > I have 2 MB of Lease line. > > This is What I se

Re: [asterisk-users] LoadAvg , Codec and Bandwidth Utilisation

: cumm: 6.47MB peak: 1.30Mb rates: 1.23Mb 1.26Mb 1.26Mb RX:6.45MB 1.28Mb 1.25Mb 1.26Mb 1.26Mb TOTAL: 12.9MB 2.58Mb 2.47Mb 2.51Mb 2.51Mb On Fri, May 1, 2009 at 3:27 PM, David @ULC wrote: > 1) If

[asterisk-users] LoadAvg , Codec and Bandwidth Utilisation

1) If I see the Loadavg more than 4 , whats the immediate solution to get it under 1 APART from restarting the server ? 2) I get too much of cross connections. Can Codec be the culprit ? I use g729. Can using GSM will solve the problem ? What could be the other reasons ? 3) Anyway to measure the

Re: [asterisk-users] 64bit: any problems with asterisk?

Is there something that you need to do so Asterisk will compile to 64bit or will Asterisk just function as 32bit on a 64bit platform? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Sunday, 26 A

Re: [asterisk-users] dahdi_dummy: Unable to register DAHDI rtc driver

On Wed, Apr 8, 2009 at 10:23 AM, Shaun Ruffell wrote: > David Backeberg wrote: >> Hello there: >> >> I think I have a silly kernel configuration problem. I'm running: >> * vanilla 2.6.27.10 kernel built from source >> * dahdi-2.1.0.4 built from source >&

Re: [asterisk-users] asterisk -C option not honored 100%

Tzafrir Cohen wrote: > On Tue, Apr 28, 2009 at 01:19:08PM -0500, David Siefert wrote: > >> Hello, >> >> I am trying to get a repeatable build setup for asterisk. Part of doing >> so involves using the -C option to specify the master config file. The >>

[asterisk-users] asterisk -C option not honored 100%

-1ubuntu3 on Ubuntu 8.10. Thanks, David ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Digium fax force T38?

On Sun, Apr 26, 2009 at 8:33 AM, Michael wrote: > Is it possible to force T38 for all invocations ReceiveFAX() ? If it's not T.38, it should instead be audio over G.711 or similar codec. Are your faxes going through as audio? If not, that's a strong indication that you have a SIP configuration is

Re: [asterisk-users] cheap CHEAP ata

Thanks for the info!!! 2009/4/24 Wilton Helm > >Have you checked ebay? > > Just beware that there are a lot of ATAs on Ebay that are locked to Vonage > or similar providers. While they are not impossible to unlock, it requires > considerable time and good Linux networking experience, as the pro

Re: [asterisk-users] listen to prompt before bridging call.

the message and then you bridge the calls using the brige app in asterisk 1.6. David 2009/4/24 Deepak > Hi, > Can someone please help to resolve the followinng issue: > > We would like an asterisk user to call a number and when the called party > picks up the phone, we play a mess

[asterisk-users] dahdi_tool reports that dahdi_dummy is UNCONFIGURED

Usually I used real Digium cards in asterisk systems, so I'm running into this for the first time. dahdi_tool reports that dahdi_dummy is in state UNCONFIGURED. This isn't super surprising, as it seems like the configuration files for DAHDI are really intended only for configuring real physical ca

[asterisk-users] Duplicating existing PBX function

hone rings instead of auto-answers. Any suggestions on how to do this? If there's another way, I'm open to that as well. TIA! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 d

Re: [asterisk-users] cheap CHEAP ata

thanks for your answer but the boards are too expensive. 4 fxs digium card is like 600. the 12 fxs open vox board (i have one) is like 40U$D per phone. (in the states). i need cheap!!! David 2009/4/24 Aryan Ameri > On Fri Apr 24 2009 14:03:14 GMT+1000 (EST) David fire > wrote: > &g

[asterisk-users] cheap CHEAP ata

hi i need many cheaps atas or some very cheap way to connect analogs phones to asterisk what do you recomend? i searches and only find solutions like 40 U$D (in the states, here in argentina is like 80 U$D) per phone any links or something? thanks! David -- (\__/) (='.'=)This is Bunny

Re: [asterisk-users] Asterisk and HUD server

Hi Marco, Try this: http://yum.trixbox.org/centos/4/RPMS/hudlite-server-1.4.32-1.i386.rpm Regards David. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Sambo Sent: Thursday, 23 April 2009 7:29 PM To: asterisk-users

[asterisk-users] how to know the channel from the iax phone side?

hi is there any way to know the channel from the phone side? (an iax phone) Thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. ___ -- Bandwidth and Colocat

Re: [asterisk-users] MOH always plays from the start

On Fri, Apr 17, 2009 at 11:39 AM, Mike wrote: > True, my mistake: I upgraded to 1.4.24.1, and the MoH file still always > start from the beginning. I believe I'm experiencing the same thing with my music on hold. I also would prefer a continuous play in the background, and I'm using asterisk-1.6.

Re: [asterisk-users] Peer 'iaxfax' is now UNREACHABLE! Time: 3

On Mon, Apr 20, 2009 at 6:02 PM, Marco wrote: > They are linked together through localhost. I've turned qualify on for the > iax peer. Periodically I've this message: > > [Apr 20 23:47:46] NOTICE[4641]: chan_iax2.c:9049 __iax2_poke_noanswer: > Peer 'iaxfax' is now UNREACHABLE! Time: 3 > [Apr 20 23

Re: [asterisk-users] T38 fax failing

On Mon, Apr 20, 2009 at 5:34 AM, Michael wrote: > Fax over T38 is failing, on the same system it worked with Callweaver. Some people claim great success with callweaver. If Callweaver is working great for you, why change what works? > What do I need to post to be get further assistance please?

Re: [asterisk-users] Asterisk 'outgoing' directory

On Mon, Apr 20, 2009 at 3:47 AM, Michael wrote: > Can this be used in the same way as Callweaver works, IE: to invoke Sendfax by > placing (using mv command) a job description file in it? Yes. Callfiles can be used to initiate faxes. ___ -- Bandwidth a

Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1

On Sat, Apr 18, 2009 at 10:27 AM, Thomas Kenyon wrote: > I will try this later, it looks straight forward enough. Does Asterisk > 1.6 SendFax command autonegotiate T.38 (in the way callweaver does)? Yes. Of course assuming that it's talking to a device that is T.38 capable. > Can I use a faxmach

Re: [asterisk-users] Here is Step by Step Example of Asterisk PBX System Install and configuration

On Fri, Apr 17, 2009 at 2:47 PM, Jimmy Ezell wrote: > Our small company is replacing Cisco CallManager with Asterisk (because we > are tired of sending them money) and I am documenting the process as I go on > my blog.  I am trying to make the notes as easy as possible in hopes that I > can eas

Re: [asterisk-users] Simultaneous Calls at a time

Xlite Btw, how to find out which codec a call is using when asterisk is dialing out ? On Thu, Apr 16, 2009 at 11:05 PM, David @ULC wrote: > > Which is the latest version of Asterisk ? > > > On Thu, Apr 16, 2009 at 11:04 PM, David @ULC wrote: > >> busy-level ? >>

Re: [asterisk-users] Simultaneous Calls at a time

Which is the latest version of Asterisk ? On Thu, Apr 16, 2009 at 11:04 PM, David @ULC wrote: > busy-level ? > > How to use it and whats the purpose ? > > > On Thu, Apr 16, 2009 at 10:43 PM, David @ULC wrote: > >> >> http://threebit.net/mail-archive/asteris

Re: [asterisk-users] Simultaneous Calls at a time

busy-level ? How to use it and whats the purpose ? On Thu, Apr 16, 2009 at 10:43 PM, David @ULC wrote: > > http://threebit.net/mail-archive/asterisk-users/msg07138.html > Remember that if you want to support attended transfers, you need at least > two > simultaneous calls. &

Re: [asterisk-users] Simultaneous Calls at a time

http://threebit.net/mail-archive/asterisk-users/msg07138.html Remember that if you want to support attended transfers, you need at least two simultaneous calls. So, its safe bet to keep call-limit=2. Advice ? On Thu, Apr 16, 2009 at 10:37 PM, David @ULC wrote: > My SIP config is be

Re: [asterisk-users] Simultaneous Calls at a time

solve my problem ? If yes. Great. Kindly advice. But will that allow 3 party conference ? On Thu, Apr 16, 2009 at 10:22 PM, David @ULC wrote: > "call-limit in sip.conf" > > Can you elaborate please and how to set that. > > Lets presume I have 10 agents and dial ratio i

Re: [asterisk-users] Simultaneous Calls at a time

"call-limit in sip.conf" Can you elaborate please and how to set that. Lets presume I have 10 agents and dial ratio is 4. On Thu, Apr 16, 2009 at 10:06 PM, David @ULC wrote: > > Even I thought so thats why I tried with 4 VOIP provider and things didn't > change.

Re: [asterisk-users] Simultaneous Calls at a time

Even I thought so thats why I tried with 4 VOIP provider and things didn't change. :-( On Thu, Apr 16, 2009 at 8:36 PM, David @ULC wrote: > > > Many time we face an issue where even if an agent is on Call, another call > comes in. > > Sometimes, even if agent hang up th

Re: [asterisk-users] Problem transferring calls between Cisco 7940 with SIP firmware

On Thu, Apr 16, 2009 at 10:08 AM, Massimiliano Stucchi wrote: > firmware.  The problem arises when transferring a call coming in from a > SIP account to another phone.  The call connects, but for the first 10 > seconds there is a situation with one-way audio, then it turns into a > fully working c

[asterisk-users] Simultaneous Calls at a time

Double , Triple and sometime 5 calls Many time we face an issue where even if an agent is on Call, another call comes i

Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1

On Wed, Apr 15, 2009 at 3:29 AM, Florian Hackenberger wrote: > Thanks for the explanation! Sounds all good. There is one remaining > question however. As you mentioned T.30, is app_fax capable of > terminating T.38? Yes although I'm speaking about 1.6. I can't say for certain what is required on

[asterisk-users] chan_mISDN with asterisk version 1.4.22 + codec negotiation patch

issues I need to be aware of? Thankyou, Regards, David Davoren Software Engineer LAKE Communications Ltd. Beech House, Greenhills Rd., Dublin 24, Ireland int. +353 1 4031084 fax +353 1 4520826 www.lakecommunications.com

[asterisk-users] Gxp 2000 softkey question

on to dial *1 during a call. I also have setup another key to monitor the status of recording (on/off) using the BLF function and the devstate function. (new in 1.6 back ported to 1.4) However I seem to be unable to combine these functions in to a single key. Can anyone offer any assistance? Thanks,

Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1

On Tue, Apr 14, 2009 at 10:46 AM, Lee Howard wrote: > David Backeberg wrote: >> It may be possible to use hylafax, but >> I don't know how or why you would. > > The reason *why* is generally due to support issues. What I was specifically getting at in the cont

Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1

On Tue, Apr 14, 2009 at 9:52 AM, Florian Hackenberger wrote: > With asterisk 1.6, is it possible to use hylafax, or would asterisk > terminate the fax calls itself? With app_fax integrated into asterisk-1.6, you have an 'infinite' modem pool that you control through the dial-plan. Using dialplan

Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1

On Tue, Apr 14, 2009 at 7:11 AM, Michael wrote: > On Tue, 14 Apr 2009 23:00:02 Florian Hackenberger wrote: > >> Can somone spot the problem? Is someone using t38modem with asterisk >> successfully? > The best advice I can offer is to give up now and use Callweaver otherwise you > can spend hours,

Re: [asterisk-users] .GSM -> .WAV (or ,MP3) Conversion

On Tue, Apr 14, 2009 at 7:39 AM, Tim Dobson wrote: > I'm trying to convert some call recordings from asterisk we have in .gsm Why not use sox for this purpose? sox mygsm.gsm -r 8000 -c 1 mywave.wav resample -ql Once it's a wav you can mp3 it with lame or your preferred encoder, but encoding and

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