Re: [asterisk-users] voicemail.conf locking problem

2006-11-29 Thread jezzzz .
Sounds like a good solution to me! Could potentially use externpass then use flock(), fcntl() et al. every once in a while to write all passwords back to voicemail.conf. Thanks for the insight. --- Michiel van Baak <[EMAIL PROTECTED]> wrote: > On 17:05, Wed 29 Nov 06, Scott Keagy wrote: > > If

Re: [asterisk-users] Voicemail, SQL & ODBC

2006-11-29 Thread RR
And as I wrote before, Asterisk <-> mySQl connection is already up and runnig (for CDR). So it just would have been quick and easy if Asterisk could have used the same path for audio data. O.K., lets invest some time in installing ODBC. NOrbert Norbert, mate, I don't know why you're havi

Re: [asterisk-users] How to park calls on a specific extension

2006-11-29 Thread Lacy Moore - Aspendora
On 11/29/06, Brian Capouch <[EMAIL PROTECTED]> wrote: Complaints are always considered, but calling the developers childish and repeating that complaint over and over in an email isn't likely to do much to advance the cause you've taken on. Sorry about the rant. I apologize for making the ch

Re: [asterisk-users] Voicemail, SQL & ODBC

2006-11-29 Thread Benjamin Jacob
AFAIK, ODBC helps you have any DB underneath, be it MySQL, PGSQL, etc., so why not go ahead with it? cheerz - Ben. Norbert Zawodsky wrote: Hi Peder, I asked the same question some time ago. Never got any answer... :-( Norbert Peder @ NetworkOblivion schrieb: Is the storage of actual

[asterisk-users] MeetMe announcements and SIP channels

2006-11-29 Thread Mike
Just curious if anyone knows of any hacks to enable announce entry/exit in MeetMe conferences with SIP (as opposed to ZAP) channels since the |i option will not work with SIP. Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- a

[asterisk-users] Return code - How to?

2006-11-29 Thread Doug Crompton
At voip-info.org they show the following example exten => s,1,Set(foo=${STAT(s,/var/t3)}) which I guess is suppose to work and make foo = size of t3 I did the following exten => 542,1,Set(s1=${STAT(e,"/var/lib/asterisk/t1")}) which should set s1 = 1 if the file exists and 0 if not. b

[asterisk-users] extension launch into AGI

2006-11-29 Thread Roy Kidder
I've got a simple set up with 1 fxo port and 1 fxs port in a Digium card connected to a POTS line and a phone set (physical extension). I've got all incoming calls launching directly into an AGI script. I'd like to do the same for the physical extension. In other words, when picking up the hand set

Re: [asterisk-users] SIP Port 5060

2006-11-29 Thread Leo Ann Boon
Brad Templeton wrote: My understanding was that the "port=" field on a particular SIP channel defines the port used at the remote end, ie. The user's phone will be talking on port X of their IP address, it does not alter what SIP port Asterisk is listening on on the Asterisk box. The host a

Re: [asterisk-users] How to park calls on a specific extension

2006-11-29 Thread Brian Capouch
Lacy Moore - Aspendora wrote: I think that's the problem with the Asterisk community right now. Anytime something is suggested, the response is either write it yourself or deal with what is there. Do you have experience with other big, complex Open Source projects? Do you know of any whe

Re: [asterisk-users] SIP Port 5060

2006-11-29 Thread lists
Agree. When I meant to say is that the step-by-step guide is for hooking up the PAP2 to the A/C line, network, and telephone set. This should give you a pretty good idea of the educational level of these people. - Daniel -Original Message- From: "Zeeshan Zakaria" <[EMAIL PROTECTED]> Sent

Re: [asterisk-users] SIP Port 5060

2006-11-29 Thread lists
That's how it's setup. However, setting up a VPN on the client's side is not done thru the provisioning :) - Daniel -Original Message- From: "Andrew Joakimsen" <[EMAIL PROTECTED]> Sent: Wed, November 29, 2006 8:39 pm To: [EMAIL PROTECTED], "Asterisk Users Mailing List - Non-Commercial Dis

Re: [asterisk-users] How to park calls on a specific extension

2006-11-29 Thread Lacy Moore - Aspendora
Either write what you want, or learn to use what we have and hope that SLA when it appears is better. Parking is not the best solution, I think that's the problem with the Asterisk community right now. Anytime something is suggested, the response is either write it yourself or deal with what

Re: [asterisk-users] SIP Port 5060

2006-11-29 Thread Brad Templeton
On Wed, Nov 29, 2006 at 04:49:38PM -0700, Joseph wrote: > What I have is that each device is listening on different port ex. > > [pstn-5665] ; incoming/outgoing calls on FXO port > type=friend > ... > port=5066 ; port on Pstn line > ... > > [318] ; incoming/outgoing calls on FXS Sipura-2002 > ty

[asterisk-users] register history

2006-11-29 Thread rilawich ango
Hi all, I noticed that when an user register asterisk, asterisk will update the corresponding record in db user table. However, if the user register failed, maybe wrong password. There is no record in database. How can I log those register records, including successful and failed login to the

Re: [asterisk-users] How to park calls on a specific extension

2006-11-29 Thread Brad Templeton
On Wed, Nov 29, 2006 at 06:05:31PM -0500, Steve Sobol wrote: > On Mon, 27 Nov 2006, Brad Templeton wrote: > > > On Mon, Nov 27, 2006 at 11:20:27PM -0500, Andrew Joakimsen wrote: > > > Can you explain how ValetParking and twenty minutes worth of "dialplan > > > creativitiy" can't do the same EXACT

RE: [asterisk-users] Siemens Gigaset C450 IP vs S450 IP

2006-11-29 Thread Cory Andrews
Andrew - I have been told they have no plans to introduce US distribution or availability on these products in the foreseeable future. I was told this by one of the channel managers from Siemens. I received some eval units of some of the Siemens SIP products from a reseller in EU, and they are qu

Re: [asterisk-users] Setting RTP ports for Asterisk?

2006-11-29 Thread Noah Miller
Hey Vincent - 1. What are the settings (in sip.conf?) to tell Asterisk to use specific ports for RTP? I guess you didn't see my reply earlier today - that setting is in rtp.conf 2. With this kind of setup, does Asterisk stay in the loop to forward RTP packets, or do X-Lite at home and the V

Re: [asterisk-users] Asterisk connection to a PBX

2006-11-29 Thread asterisk-robert
Inital setup for testing will be 2-4 channels in order to prove the concept. When successful we may include some PBX systems that do not have available T-1 slots. On Wed, 29 Nov 2006 19:18:50 -0800, "Tom Lynn" <[EMAIL PROTECTED]> wrote: > How many channels do you require? I'd favor T1 for a

RE: [asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread Ira
At 03:49 PM 11/29/2006, you wrote: That data is no easier to parse than the output generated from: Action: Command Command: sip show peer Note the colons used both as a field delimiter AND stuck in the regcontact. Actually, finding the data between a CR and a colon get's you the data label

Re: [asterisk-users] Asterisk connection to a PBX

2006-11-29 Thread Tom Lynn
How many channels do you require? I'd favor T1 for a few reasons. Higher port density means fewer cards per system, which will mean fewer interrupts. T1s won't require you to tune analog levels. Echo probability will be lower. On 11/29/06, asterisk-robert <[EMAIL PROTECTED]> wrote: We are

[asterisk-users] Trouble using 2 IAX2 DiDs provided by different ITSPs

2006-11-29 Thread hugolivude
Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both

[asterisk-users] Re: asterisk-users Digest, Vol 28, Issue 152

2006-11-29 Thread Ishanka Anuradha Ranasooriya
g beeping sounds at regular intervals no matter which phone we use. Does anyone know why? We are using a diqium tdm card. Thanks Kim -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061129/81c3ce7

[asterisk-users] Conferencing Issue please help

2006-11-29 Thread Ishanka Anuradha Ranasooriya
Hi All, I have a problem in configuring in asterisk. I configure asterisk meetme.conf and extension.conf, but when i transfer call to conference it give me this message and asterisk kill it self. Ouch ... error while writing audio data: : Broken pipe If any one knows abou

Re: [asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread Sig Lange
I didn't have a problem writing a client in C. There were a couple things to worry about I have teh code in svn on sourceforge and havn't done much w/ it anymore since i've moved jobs. Anyone want it? e-mail me directly. i'l freely hook it up. On 11/29/06, Richard Lyman <[EMAIL PROTECTED]>

Re: [asterisk-users] SIP Port 5060

2006-11-29 Thread Zeeshan Zakaria
http or tftp provisioning is what you do in this type of business. I can't imagine how can one do this business without using remote provisioning. PAP2 devices have excellent support for http, https and tftp. Using remote provisioning, you can easily control their ports, DIDs and all configuration

Re: [asterisk-users] Attn: DISA Experts(Strange problem with DISA)

2006-11-29 Thread Andrew Joakimsen
What if you used a better defined dialplan? On 11/28/06, Crazy Boy <[EMAIL PROTECTED]> wrote: Hi Friends, I am facing a strange problem with DISA. I have installed and configured Trixbox. I've created a secret extension i.e., 555 and called this extension in Digital Receptionist using custom e

Re: [asterisk-users] Siemens Gigaset C450 IP vs S450 IP

2006-11-29 Thread Andrew Joakimsen
Does anyone know where to source the Siemens Gigaset phones in North America? I called 1-800-SIEMENS and was told the Gigaset range is no longer marketed here since a few years ago. How far from being FCC compliant is the DECT standard? On 11/29/06, Eugen Leitl <[EMAIL PROTECTED]> wrote: I've

Re: [asterisk-users] SIP Port 5060

2006-11-29 Thread Andrew Joakimsen
Why don't you use the provisioning? it runs through http server... On 11/29/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: That's not possible. These are residential people who hardly know enough to hook up their PAP2 with detailed step-by-step instructions on hand and support on the phone :)

Re: [asterisk-users] How to park calls on a specific extension

2006-11-29 Thread Ira
At 03:08 PM 11/29/2006, you wrote: It'd actually be great, because the behavior would be almost identical to that of our old PBX. Either write what you want, or learn to use what we have and hope that SLA when it appears is better. Parking is not the best solution, but it works. If you have A

Re: [asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread Richard Lyman
TP'n to follow flow.. yes i understand the out is similar/same. i was attempting to get james to note that there already are manager commands. as a note for parsing the manager interface.. you will probably find it easier if you parse as a block, meaning *till* a blank line. then parse the

Re: [asterisk-users] Re: What's up with the Manager Interface?!?!

2006-11-29 Thread Richard Lyman
Tony Mountifield wrote: In article <[EMAIL PROTECTED]>, Richard Lyman <[EMAIL PROTECTED]> wrote: just wait till you get a 'hiccup' that causes a line to get cut off, drop a char, and continue on next line. (examples below) I've made heavy use of the Manager interface for over 2 years

[asterisk-users] Setting RTP ports for Asterisk?

2006-11-29 Thread Vincent Delporte
Hello When I make calls from home to the PSTN by going through the Net -> Asterisk -> the Net -> VoIP provider -> PSTN, I get no sound either way. I assume it's because I must tell Asterisk to use fixed ranges of UDP ports and map ports accordingly on the NAT firewall under which it is located

RE: [asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread Douglas Garstang
That data is no easier to parse than the output generated from: Action: Command Command: sip show peer Note the colons used both as a field delimiter AND stuck in the regcontact. CodecOrder: ulaw,g729,gsm Status: UNKNOWN SIP-Useragent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Reg-Contact : si

Re: [asterisk-users] SIP Port 5060

2006-11-29 Thread Joseph
What I have is that each device is listening on different port ex. [pstn-5665] ; incoming/outgoing calls on FXO port type=friend ... port=5066 ; port on Pstn line ... [318] ; incoming/outgoing calls on FXS Sipura-2002 type=friend ... port=5069 ; port on FXS line ... etc. Though I'm not sure if

Re: [asterisk-users] How to park calls on a specific extension

2006-11-29 Thread Steve Sobol
On Fri, 24 Nov 2006, Brad Templeton wrote: > As I was noting in an earlier message, the parking lot concept is to my > view not a thrilling interface at best, and I can't see many times one > would want it in a SOHO environment.It seems best for a large PBX > where people are moving to random

[asterisk-users] Call dropping

2006-11-29 Thread Ed Nuñez
Can you please tell me what am I missing? If I don't make a choice before the last prompt starts playing the call is falling through. I want the menu to be replayed if a choice is not made, but the call gets disconnected and I see the following message in the Asterisk BE colsole. == Pars

RE: [asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread Douglas Garstang
> -Original Message- > From: James Texter [mailto:[EMAIL PROTECTED] > Sent: Wednesday, November 29, 2006 3:03 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] What's up with the Manager Interface?!?! > > > Doug, > Your issue isn't with t

Re: [asterisk-users] How to park calls on a specific extension

2006-11-29 Thread Steve Sobol
On Mon, 27 Nov 2006, Brad Templeton wrote: > On Mon, Nov 27, 2006 at 11:20:27PM -0500, Andrew Joakimsen wrote: > > Can you explain how ValetParking and twenty minutes worth of "dialplan > > creativitiy" can't do the same EXACT thing you are describing? Sometimes the > > simplest answer is never th

Re: [asterisk-users] voicemail.conf locking problem

2006-11-29 Thread Tzafrir Cohen
On Wed, Nov 29, 2006 at 05:05:13PM -0500, Scott Keagy wrote: > If you have enough users where this comes up as a real issue, I'd > recommend migrating to Asterisk Realtime voicemail, [ snip ] If you have expected a different type of reply, consider discussion this in asterisk-dev . Yes. The c

[asterisk-users] Call recording with Asterisk BE

2006-11-29 Thread Ed Nuñez
ecuting Set("SIP/1656-b7d10740", "AGENTFILENAME=1656-20061129-183350-") in new stack -- Executing Monitor("SIP/1656-b7d10740", "wav|1656-20061129-183350-|m") in new stack -- Executing Queue("SIP/1656-b7d10740", "NOC") in new stack Ed Nuñez

Re: [asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread Richard Lyman
James Texter wrote: Doug, Your issue isn't with the manager. It's with the CLI output you are trying to hijack via manager :D If you run "sip show peer 2944093" in the CLI, you'll see a blank line, followed by a line that is "* Name". It appears what you really want is a manager Action to

Re: [asterisk-users] voicemail.conf locking problem

2006-11-29 Thread Michiel van Baak
On 17:05, Wed 29 Nov 06, Scott Keagy wrote: > If you have enough users where this comes up as a real issue, I'd recommend > migrating to Asterisk Realtime voicemail, then can have row-level locking > etc. if you use the right kind of storage engine... I've found problems using > the dial-by-name

[asterisk-users] Cisco 7940 Firmware 8.2

2006-11-29 Thread James R. Stevens
Greetings, I am cutting my teeth with SIP phones and my first issue is getting a Cisco 7940 to Authenticate with my VoIP provider (BBTelsys). I did read some notes on the vo-ip website about 7.5 being the better firmware version. Has anyone had trouble with 8.2 and SIP registering? Should I just

Re: [asterisk-users] Manage Users in LDAP

2006-11-29 Thread Gary Richardson
phpldapadmin is pretty nice. I was using 2-3 different ldap clients to get the job done until I got over my php bias and installed it. It lets me do everything I want, without crashing. On 11/27/06, Steven Baker <[EMAIL PROTECTED]> wrote: Hello All, we are using asterisk+openldap. Do is there a

RE: [asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread Douglas Garstang
> -Original Message- > From: James Texter [mailto:[EMAIL PROTECTED] > Sent: Wednesday, November 29, 2006 3:03 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] What's up with the Manager Interface?!?! > > > Doug, > Your issue isn't with t

[asterisk-users] Modprobe Zaptel

2006-11-29 Thread Julian Varanini
Hi all For some dumb reason I decided to upgrade from Mandriva 2006 to 2007, thinking I could install asterisk all over again. Anyway I did install asterisk, zaptel and libpri. After install I ran modprobe zaptel which said "zaptel not found". Thanks to help on this mailing list I had a fix

[asterisk-users] Re: What's up with the Manager Interface?!?!

2006-11-29 Thread Tony Mountifield
In article <[EMAIL PROTECTED]>, Richard Lyman <[EMAIL PROTECTED]> wrote: > just wait till you get a 'hiccup' that causes a line to get cut off, > drop a char, and continue on next line. > (examples below) I've made heavy use of the Manager interface for over 2 years now, and have never seen the

[asterisk-users] g726 voice prompts

2006-11-29 Thread Eric Bishop
Anyone know if it posible to make voice promps native g726 or g711 format? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asteris

RE: [asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread Douglas Garstang
> -Original Message- > From: Steve Edwards [mailto:[EMAIL PROTECTED] > Sent: Wednesday, November 29, 2006 2:55 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] What's up with the Manager Interface?!?! > > > On Wed, 29 Nov 2006, Douglas Garst

[asterisk-users] beeping noise in background

2006-11-29 Thread Kim Jones
I have asterisk 1.2.12.1 running with several client phone options. Our echo cancellation is finally working great. The only problem I seem to be having is there is background noise including beeping sounds at regular intervals no matter which phone we use. Does anyone know why? We are using a d

Re: [asterisk-users] Billing software with reseller accounts

2006-11-29 Thread Dovid B
I have been using Enswitch. Has some bugs but over all works great. It's not open source but worth the money. - Original Message - From: "Guillermo Salas M." <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, November 29, 2006 3:12 AM S

RE: [asterisk-users] voicemail.conf locking problem

2006-11-29 Thread Scott Keagy
If you have enough users where this comes up as a real issue, I'd recommend migrating to Asterisk Realtime voicemail, then can have row-level locking etc. if you use the right kind of storage engine... I've found problems using the dial-by-name directory with realtime voicemail, but it seems you

Re: [asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread James Texter
Doug, Your issue isn't with the manager. It's with the CLI output you are trying to hijack via manager :D If you run "sip show peer 2944093" in the CLI, you'll see a blank line, followed by a line that is "* Name". It appears what you really want is a manager Action to show a sip peer, in wh

RE: [asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread Steve Edwards
On Wed, 29 Nov 2006, Douglas Garstang wrote: G. Here's another example... Action: Command Command: sip show peer 2944093 Response: Follows Privilege: Command * Name : 2944093 Secret : MD5Secret: Context : 180o_CallStart Subscr.Cont. : 180o_WatchBLF Why the HE

[asterisk-users] voicemail.conf locking problem

2006-11-29 Thread jezzzz .
I'm wondering if anyone is having problems when multiple users concurrently change their voicemail passwords. Consider the following scenario (based on vm_change_password() in app_voicemail.c): - user1 wishes to change his password so voicemail.conf is opened and read into a buffer - user1 change

Re: [asterisk-users] Re: Spandsp rxfax txtax fails no errors

2006-11-29 Thread daveasterisk
Thanks for the response!!! I enabled debuging in the menuselect configuration for compiling asterisk 1.4 beta3. In logging.conf enabled debug loggin to the /var/log/asterisk/debug file and to the console. Restarted (not just reload) asterisk and there is plenty of general debugging info in the

RE: [asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread Douglas Garstang
> -Original Message- > From: Douglas Garstang > Sent: Wednesday, November 29, 2006 12:26 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [asterisk-users] What's up with the Manager Interface?!?! > > > > -Original Message- > > From: Michael Collin

[asterisk-users] Polycom 601 Second Incoming Call

2006-11-29 Thread Dovid B
Hi List, I have a Polycom 601 that when the user is on the phone they only hear one beep and the CID of the second incoming call is not shown. Is there a way to have the CID show up for the second call ? And a way to configure the phone to beep more often if there is another call coming in. The

[asterisk-users] Playing streaming MOH in Asterisk

2006-11-29 Thread Matt
I thought I sent this out.. but don't see it so apologies if it went already. I am trying to get streaming MOH working but haven't been able to.. I am running 1.2.x Based on people's suggestions in other e-mails I've tried: [scanner] mode=custom application=/usr/local/bin/mpg123 -s --mono -y

RE: Spam? Re: [asterisk-users] Getting app_cepstral to work withAsterisk 1.4.0-beta3

2006-11-29 Thread Hall, Eric M.
I get an error when I do a make install [EMAIL PROTECTED] app_swift-0.9.5]# make install gcc -g -Wall -D_REENTRANT -D_GNU_SOURCE -fPIC -DCHANNEL_HAS_CID -DNEW_CONFIG -I/opt/swift/include -c -o app_swift.o app_swift.c app_swift.c:49: warning: type defaults to `int' in declaration of `STANDARD_

Re: [asterisk-users] Voicemail, SQL & ODBC

2006-11-29 Thread Norbert Zawodsky
Hi Noah, Noah Miller wrote: > Hi Norbert - > > Just a thought: You could go the other way - share a volume on a > separate webserver, and have the asterisk box connect to the webserver > via NFS as a client, and store the voicemail on the NFS share. While > I don't have any exact numbers, it seem

Re: [asterisk-users] mISDN

2006-11-29 Thread Patrick
On Wed, 2006-11-29 at 16:38 +0100, Timothy Parez wrote: > I get the following with debug on: > > P[ 3] I IND :SETUP oad:497978546 dad:50556010 pid:10 state:none > P[ 3] --> channel:1 mode:TE cause:16 ocause:16 rad: cad: > P[ 3] --> info_dad: onumplan:2 dnumplan:2 rnumplan: cpnnumplan:0 > P[ 3]

Re: [asterisk-users] No sound: X-Lite -> Asterisk -> VoIP Provider -> Cellphone

2006-11-29 Thread Noah Miller
Hi Vincent - Here's what I did on the X-Lite at home in the Topology section: IP address : Discover global address STUN server : Discover server Port used on local computer : Manually specify range 8000-8019 Here are the ports that I forwarded from my NAT router at home: UDP 5060 UDP 3478 (STUN

RE: [asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread Michael Collins
> Here's a good example. I'm trying to get SIP blf. I managed to split my > result into a list of lines by splitting on ANY of \r\n, \n or \r. I was > going use the column headings from the third line as my keys for my > dictionary/hash, rather than hard coding them. Notice anything? The 'Call > I

Re: [asterisk-users] Voicemail, SQL & ODBC

2006-11-29 Thread Noah Miller
Hi Norbert - >> I want to store all of my voicemail stuff in a database so that I can >> give users web access to it, but I don't want to run web services on my >> * server itself. If it is all in a DB, I can have a web box and a >> separate SQL box and none of it should affect *. > > Yes, you

RE: [asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread Douglas Garstang
> -Original Message- > From: Matt Florell [mailto:[EMAIL PROTECTED] > Sent: Wednesday, November 29, 2006 12:52 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] What's up with the Manager Interface?!?! > > > On 11/29/06, Douglas Garstang <[EM

Re: [asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread Matt Florell
On 11/29/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. You would need a ton of rope and a few hundred horses for that :) The Manager API code is distributed across dozens of source files in the

[asterisk-users] NEED ASTERISK DEVELPER : OH323-asterisk driver and openh323

2006-11-29 Thread Oliver Vermeulen
Dear List, I'm looking for a coder/developer that can modify oh323 return codes on asterisk Example on based on SIP and h323. Right now we are receiving : Call Rejected (code 21) Network Out of Order (code 38) Need to able to replace dose codes with -> No Circuit/Channel Available (code 34) P

[asterisk-users] Asterisk connection to a PBX

2006-11-29 Thread asterisk-robert
We are thinking of setting up an Asterisk system to route calls between 2 of our factories. Our idea is to connect an Asterisk box to each PBX and then use SIP(or IAX) to truck between the 2 systems on our internal network. I would be interested in any ideas regarding the connection points:

RE: [asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread Douglas Garstang
> -Original Message- > From: Michael Collins [mailto:[EMAIL PROTECTED] > Sent: Wednesday, November 29, 2006 11:20 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] What's up with the Manager Interface?!?! > > > > Sometimes the data comes back

Re: [asterisk-users] Getting app_cepstral to work with Asterisk 1.4.0-beta3

2006-11-29 Thread Bob Chiodini
Eric, It looks like the definition for PTHREAD_MUTEX_RECURSIVE is within an #ifdef __USE_UNIX98 (on Fedora Core 6, anyway). You could try defining it within the Makefile. Similar to the _GNU_SOURCE definition in the app_cepstral.so: app_cepstral.c stanza. Bob... On Wed, 2006-11-29 at 13:38 -05

Re: [asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread Richard Lyman
Douglas Garstang wrote: The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it

[asterisk-users] I am unable to find any included rpms with hudlite...

2006-11-29 Thread Jordan Novak
I am installling on a scratch asterisk running white box linux (fedora) Does anyone know where to find them after the rpm runs. I am looking for ircd and the perl dependancies. The instructions make a ton of assumptions, so I am not sure what is happening here. Jordan Novak Senior Telecommunicati

Re: [asterisk-users] Getting app_cepstral to work with Asterisk 1.4.0-beta3

2006-11-29 Thread Earle Clubb
Hall, Eric M. wrote: Using this link http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt   This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3   I get the following errors on make install   Any help would be GREAT!   Thanks  

[asterisk-users] Call Recording and Call Transfers

2006-11-29 Thread Stephen Kratzer
Howdy, Anybody have any ideas on how to record to a different file each time a call is transferred by means of the transfer button on Polycom phones? I basically need to be able to execute StopMonitor and an AGI script each time a call is transferred without using features.conf for transfers. T

RE: [asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread Michael Collins
> Sometimes the data comes back separated by \r\n, and sometimes it's > separated by \n. > The whole thing is completely inconsistent, and trying to write any kind > of API for it is -GHASTLY- Doug, What language(s) are you using? Just curious. I've been tinkering with Perl, POE, and POE::Compo

Re: [asterisk-users] SIP Port 5060

2006-11-29 Thread lists
Ok. So, it's not that I can make Asterisk listen on multiple ports for any SIP friend, but I could override the port on an individual SIP friend. So, instead of having something like: bindport=5060,5080,5081,5082 in the general section of sip.conf, I need to just have bindport=5060 in the gene

Re: [asterisk-users] SIP Port 5060

2006-11-29 Thread lists
Wow. I didn't know you could do that. So, I could have something like this in sip.conf: bindport=5060,5080,5081,5082 and it will make Asterisk listen on all those 4 ports? - Daniel -Original Message- From: "Joseph" <[EMAIL PROTECTED]> Sent: Wed, November 29, 2006 1:31 am To: [EMAIL PRO

Re: [asterisk-users] SIP Port 5060

2006-11-29 Thread lists
That's not possible. These are residential people who hardly know enough to hook up their PAP2 with detailed step-by-step instructions on hand and support on the phone :) Thanks, Daniel -Original Message- From: "Tom Lynn" <[EMAIL PROTECTED]> Sent: Wed, November 29, 2006 12:28 am To: "Aste

[asterisk-users] What's up with the Manager Interface?!?!

2006-11-29 Thread Douglas Garstang
The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug. _

Re: [asterisk-users] AGI PHP Issues (Not new to Asterisk but new to AGI)

2006-11-29 Thread Time Bandit
I am attempting my first go at a simple AGI application using PHP (Getting Asterisk to SAY PHONETIC ABC). I have dabbled with PHP but I am by no means a professional standard developer. Can't really say what is wrong with your code since I never did an AGI in PHP without this class : http://phpa

Re: [asterisk-users] g729 registered

2006-11-29 Thread Ralph Liebessohn
On 11/20/06, Ralph Liebessohn <[EMAIL PROTECTED]> wrote: On 11/20/06, Alex Robar <[EMAIL PROTECTED]> wrote: > Hi Ralph, > > Have you setup your PAP2 to allow the 729 codec? I believe you actually > have to tell it that it's allowed to use that codec before it will work. > > Cheers, > Alex > >

Re: [asterisk-users] Sipura phone does not ring

2006-11-29 Thread Larry Alkoff
Fran when you say "specify the next hop" do you mean the S0 line be an extension in sip.conf or a context in extensions.conf? Or should the line simply be tacked on to my [default] context? Larry Fran Oliveira wrote: I think it is wrong. You should specify the next hop with some like this S0<

[asterisk-users] Loosing IAX connection between offices

2006-11-29 Thread DM
Setup: Office A: router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv Asterisk: v.1.2.4 static IP Office B: router: Linksys WRT54GL running Linksys firmware v4.30.2 Asterisk: v.1.2.7.1 dynamic IP (using dyndns name) Office A is set up with refresh dns and cron job for iax2 reload e

Re: [asterisk-users] Asterisk + Avaya S8700

2006-11-29 Thread Michel R Vaillancourt
Tomer Horn wrote: Hello list, I am curious here if anybody here got an experience connecting Avaya to Asterisk using H323 / T1. I am completely lack of experience with Avaya and I wanna know if anybody here has connected Avaya to Asterisk using H323 and managed to stabilize it. Google provide

[asterisk-users] Getting app_cepstral to work with Asterisk 1.4.0-beta3

2006-11-29 Thread Hall, Eric M.
Using this link http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3 I get the following errors on make install Any help would be GREAT! Thanks [CC] app_cepstral.c -> app_cepstral.o In file

Re: [asterisk-users] Re: IAX access to FWD broken?

2006-11-29 Thread Zed
I've got to the point with FWD and IAX that I just connect directly via SIP to IPKall, using my Asterisk box's address as the proxy. It simply works better and eliminates another point of failure at FWD. I also find it helps keep things a little more organized since I can assign my own internal S

[asterisk-users] b410p hangup detection - Portugal

2006-11-29 Thread Nuno Pais Fernandes
Hi, I've setup a trixbox 1.2.2 instalation with digium b410p. It uses misdn driver. I'm able to place calls and receive calls using this interface. One thing that happens is when i dial an a number from a sip client to an number that is routed through b410p and the called party rejects the call

Re: [asterisk-users] mISDN

2006-11-29 Thread Timothy Parez
Hi, Sorry, uncomenting that actually worked. Now I need to filter on the last two numbers, that shoulnd't be to hard I guess. Tim. Giorgio Incantalupo schreef: Hi Patrick, it seems like Asteirsk cannot match any number, try uncomment the two following lines: ;exten => _X.,1,Dial(SIP/tim

Re: [asterisk-users] IAX access to FWD broken?

2006-11-29 Thread Derek Whitten
jason wrote: > last I had heard, pretty much all FWD accounts that were created in the > past year or so no longer work with IAX. Still don't know why. > > Timothy Parez wrote: >> I've got the same problem here. >> It can't register anymore --> timeout >> >> >> Brian Capouch schreef: >>> I hadn't

Re: [asterisk-users] mISDN

2006-11-29 Thread Timothy Parez
I get the following with debug on: P[ 3] I IND :SETUP oad:497978546 dad:50556010 pid:10 state:none P[ 3] --> channel:1 mode:TE cause:16 ocause:16 rad: cad: P[ 3] --> info_dad: onumplan:2 dnumplan:2 rnumplan: cpnnumplan:0 P[ 3] --> Bearer: Speech P[ 3] --> Codec: Alaw P[ 0] --> * NEW CHANNEL

[asterisk-users] Asterisk + Avaya S8700

2006-11-29 Thread Tomer Horn
Hello list, I am curious here if anybody here got an experience connecting Avaya to Asterisk using H323 / T1. I am completely lack of experience with Avaya and I wanna know if anybody here has connected Avaya to Asterisk using H323 and managed to stabilize it. Google provides mixed comments r

Re: [asterisk-users] mISDN

2006-11-29 Thread Timothy Parez
Hi, I did, that was my first try, but it didn't work. Giorgio Incantalupo schreef: Hi Patrick, it seems like Asteirsk cannot match any number, try uncomment the two following lines: ;exten => _X.,1,Dial(SIP/timothy,30,r) ;exten => _X.,2,Hangup() Giorgio Incantalupo On Wed, 2006

Re: [asterisk-users] Re: IAX access to FWD broken?

2006-11-29 Thread Timothy Parez
Still doesn't work for me. Still get timeout Michael Graves schreef: I'm travelling today but I was just able to use Firefly to login to FWD via IAX2. I called the echo test with no problems other than the lousy network in this hotel. My Astlinux server Also reports that it's registered with

[asterisk-users] Blind transfer # not working for forwarded or picked calls

2006-11-29 Thread Roger Lewau
Hello list We have a situation where calls need to be transfered to another extension. We are using # to accomplish this but we found this is only working for calls answered at the original called extension. If the call has been forwarded to another extension or if the call has been picked up by

Re: [asterisk-users] Voicemail, SQL & ODBC

2006-11-29 Thread Norbert Zawodsky
Noah Miller wrote: > Hi Peder - > >> Is the storage of actual voicemail messages in a database still limited >> to ODBC? If so, why? > > Yes. Why? Nobody has developed a voicemail solution that directly > connects to a *SQL database for message storage. A clear answer :-) Although a sad one :-( B

Re: [asterisk-users] Voicemail, SQL & ODBC

2006-11-29 Thread Norbert Zawodsky
Derek Whitten wrote: > Norbert Zawodsky wrote: > >> RR wrote: >> >> >>> Mate, I can't say it with authority but I'm almost certain that the >>> only DB that a specific driver was written for is MySQL. I think if >>> you use res_mysql.o you should be able to talk to mySql directly >>> witho

Re: [asterisk-users] Cisco 7970 SIP upgrade issues

2006-11-29 Thread Paul A Brown
Thanks for all the help guys. I cannot load the new SIP image straight on as the SCCP image is very old. i read the FAQs posted on the lists and it tells me I need to upgrade the SCCP image to at least 7 before I can load the SIP image. This is the problem I am having. I cannot load SIP until I

Re: [asterisk-users] Re: IAX access to FWD broken?

2006-11-29 Thread Michael Graves
I'm travelling today but I was just able to use Firefly to login to FWD via IAX2. I called the echo test with no problems other than the lousy network in this hotel. My Astlinux server Also reports that it's registered with FWD via IAX2. My account is a couple years old. Michael On Wed, 29 N

Re: [asterisk-users] mISDN

2006-11-29 Thread Giorgio Incantalupo
Hi Patrick, it seems like Asteirsk cannot match any number, try uncomment the two following lines: > ;exten => _X.,1,Dial(SIP/timothy,30,r) > ;exten => _X.,2,Hangup() Giorgio Incantalupo > On Wed, 2006-11-29 at 13:26 +0100, Timothy Parez wrote: >> Hi, >> >> I'm able to place outgoing calls using

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