Re: [asterisk-users] Change Codec

Here is some more details about my setup: Customer <-> PRI <-> Server A with G.729 <-> IAX Trunk G729 <-> Server B no G729 (pass through) <-> Snom Phone with G729 with incoming call there is no problem with when I try to make outbound and want to play some prompt on server b Im not able. in ser

[asterisk-users] make iso image

Dears I am using centos 4.4 wirh asterisk 1.2.17 What I want is to make a customized packages to be installed from cd with no need every time to install packages and my personalized web interface , Regards * No employee or agent is authorize

Re: [asterisk-users] Cisco 7940 no outgoing audio

Hi Salvatore The firmware is PS03-08-2-00. Unfortunately I can only packet capture on the asterisk server itself, but I am seeing: P 172.16.8.22 > 172.16.8.1: ICMP 172.16.8.22 udp port 2224 unreachable, length 36 IP 172.16.8.20 > 172.16.8.1: ICMP 172.16.8.20 udp port 17099 unreachable, length 36

Re: [asterisk-users] ADSL routers with integrated SIP QoS for otherdevices

Hi, I can agree for smaller installation/home offices the Linksys WRT series is pretty good (I'm using this at home). I'm using the dd-wrt Firmware (www.dd-wrt.com) which is also available for plenty other routers. With QoS values set right I always have clear audio even under "rough" conditions

Re: [asterisk-users] OT: Capture Asterisk traffic

I think you want: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp dst portrange 5060-65534 Thanks tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst portrange 5060-35000 tcpdump: unknown host 'portrange' tcpdump version 3.8 libpcap version 0.8.3 man tcpdump indicates tha

Re: [asterisk-users] OT: Capture Asterisk traffic

Well, the first thing I notice is that your first tcpdump example is listening on eth0, and the second is listening on eth1. What happens when you do tcpdump -i eth1 -s 0 -w /tmp/tcpdump.1 Do you see the RTP traffic then? Thanks That was a typo. Should have read: The following works: tcpdum

Re Re: [asterisk-users] TC400B

Dear Andres; How much it cost the 4 licenses of G729 and from where I have to buy them? Also, what if I need to do IP Trunk between Asterisk and another IP PBX in another side (in case I need 30 ports for this IP Trunk, and I need to use G729 or G723 codec), then also I need to buy a license for

[asterisk-users] Re: RE: Digital Phones (Dean Collins)

Dear Collins; But what the cards that I can use it for these digital phones (if available)? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

[asterisk-users] Asterisk Integration of XMPP/Jingle

Hi, Anybody here able to get Jingle work with Asterisk? Demuel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Newbie Question about E1

Hi PaulH, thanks for your answer! Now, another question.. Every E1 card has support for pri_net/pri_cpe or only some of them has? Can you tell me at least one card that can do that? Thanks again. -- Luar Roji On Fri, Apr 20, 2007 at 03:32:45PM +1000, Paul Hales wrote: > > Your best bet is a

[asterisk-users] VPN between Asterisk server and phone client

Hi, I wish to make a secure tunnel between the asterisk server and the softphone. I have seen the feature X-tunnel in x-lite. It will be a great help if somebody can guide me to confure a tunnel between the asterisk server and X-lite or any other phone. Regards, biju __

RE: [asterisk-users] How many users can be supported simultaneously?

Thanks again but my "scenario" is being developed for academic purposes so I afford limited resources..Furthermore the server has not a free PCI slot so I can not add any second interface..SIP packets are of higher priority comparing to the other data Από: [EMA

RE: [asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help

Hi Remco Post, Thank you for the tip. I have verified that the permissions are correct for the table and procedure. However, I think I may have got to the bottom of the issue now. What look like was happening is that asterisk was trying to delete any matching row prior to an insert operation. So

Re: [asterisk-users] OT: Capture Asterisk traffic

On Wed, 2 May 2007, CSB wrote: Well, the first thing I notice is that your first tcpdump example is listening on eth0, and the second is listening on eth1. What happens when you do tcpdump -i eth1 -s 0 -w /tmp/tcpdump.1 Do you see the RTP traffic then? Thanks That was a typo. Should have

Re: [asterisk-users] How many users can be supported simultaneously?

Hi, there are some interesting figures on http://www.thrallingpenguin.com/articles/asterisk-solaris.htm. Knud Antonopoulos Angelos wrote: Thanks again but my "scenario" is being developed for academic purposes so I afford limited resources..Furthermore the server has not a free PCI slot so

Re: [asterisk-users] Applet?

Dean : you know well there are other Click-to-Talk at this price range. http://www.bicomsystems.com/callnow/ coming in at £1000, $2000, €1500 and to include minor customisations to provide your look and feel. Should anyone want more info please contact me offline. Steve steve 'at' bicomsystems

Re: [asterisk-users] OT: Capture Asterisk traffic

. . . > man tcpdump indicates that I should be able to use >= syntax but it > doesn't > work as expected. Any further advice appreciated. > > Cameron When interested in packets, I usually use ethereal and a 4 port hub, plugging the ethereal and asterisk boxs into the hub and uplink the hub to

RE: [asterisk-users] Applet?

Take it to the Biz list. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Wingfield Sent: Wednesday, May 02, 2007 6:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [aster

RE: [asterisk-users] OT: Capture Asterisk traffic

> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Joe acquisto > Sent: Wednesday, May 02, 2007 6:08 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] OT: Capture Asterisk traffic > > . . .

RE: [asterisk-users] VPN between Asterisk server and phone client

> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Biju > Sent: Wednesday, May 02, 2007 5:38 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [asterisk-users] VPN between Asterisk server and phone client >

Re: [asterisk-users] Improving Asterisk's DNS support

Brian Capouch wrote: > Kristian is exactly right: there needs to be an architectural fix that > will stop EVERYTHING in the server from hanging when lookups are done > (and then hang) for a given IAX/SIP peer. It seems to me the best approach would be to report and document the problem. Has anyo

Re: [asterisk-users] headsets for linksys/sipura phones?

Wilson Pickett wrote: >> I have 4-5 different Nokias, none have a 2.5mm jack. Nothing that >> even remotely resembles a jack. > > The older ones did have 2.5 jacks Interesting - I wonder if this is due to different models being marketed in different places? I have a Nokia 7110 from around 1999

Re: [asterisk-users] Funky BIND/named errors

Brett Crapser wrote: > I have been getting these for awhile now in my log files. > > Apr 24 11:02:38 asterisk named[1072]: > lame server resolving 'pbx_loopback.so' (in'so'?): 205.166.226.38#53 > Apr 24 11:02:38 asterisk named[1072]: > lame server resolving 'pbx_dundi.so' (in 'so'?): 205.166.226.

RE: [asterisk-users] Digital Phones

> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Stephen Bosch > Sent: Wednesday, May 02, 2007 1:36 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Digital Phones > > Salvatore Giudice

[asterisk-users] MySQL ** DBI connect failed : Too many connections **

Hey everybody, I'm hoping someone can look over the below statement and tell me if I've got something wrong. Yesterday, after running the MySQL addon version 1.2.4 under Asterisk 1.2.17 for a couple months, I started to get connection failures for one of my systems. Logging into the webmin

Re: [asterisk-users] make iso image

Mondo/Mindi Backup will work On 5/2/07, Khaled Chehab <[EMAIL PROTECTED]> wrote: Dears I am using centos 4.4 wirh asterisk 1.2.17 What I want is to make a customized packages to be installed from cd with no need every time to install packages and my personalized web interface , Reg

[asterisk-users] voice mail format

Hi folks, my goal is to access voicemail (there were some posts about this) but not by dialing numbers. As asterisk sends voicemails in e-mail, it's cheaper for us to read e-mails on our cell phone (3g, gprs), and the message is attached there. i've looking around in voicemail.conf and foun

Re: [asterisk-users] make iso image

2007/5/2, Khaled Chehab <[EMAIL PROTECTED]>: Dears I am using centos 4.4 wirh asterisk 1.2.17 What I want is to make a customized packages to be installed from cd with no need every time to install packages and my personalized web interface , Regards -- ***

Re: [asterisk-users] make iso image

On Wed, May 02, 2007 at 11:07:43AM -0700, Khaled Chehab wrote: > Dears > > I am using centos 4.4 wirh asterisk 1.2.17 > > What I want is to make a customized packages to be installed from cd with no > need every time to install packages and my personalized web interface , > You're asking in th

Re: [asterisk-users] OT: Capture Asterisk traffic

On Wed, May 02, 2007 at 08:52:42PM +1200, CSB wrote: > > > >Well, the first thing I notice is that your first tcpdump example is > >listening on eth0, and the second is listening on eth1. > > > >What happens when you do > > > >tcpdump -i eth1 -s 0 -w /tmp/tcpdump.1 > > > >Do you see the RTP traffic

RE: [asterisk-users] MySQL ** DBI connect failed : Too many connections**

http://dev.mysql.com/doc/refman/5.0/en/too-many-connections.html Max connections is set to 100+1 by default. Maybe you just need to raise that number. Otherwise, figure out why connections are staying open if they are. Thanks, Steve Totaro http://www.asteriskhelpdesk.com/component/option,com_wr

Re: RE : [asterisk-users] How do I do this in Asterisk?

Hi Francoies, Many thanks for your reply will give it a try. Best regards and thanks, Christian On Tue, 2007-05-01 at 20:09 +0200, [EMAIL PROTECTED] wrote: > Hi Christian, > > Increase a variable in the menu loop, or exactly in the "t" and "i" > extensions like this : > > exten => s,1,Wait(3)

[asterisk-users] Volume (gain?) on VoIP-only system.

Hi, all. I've got a customer who's complaining of low volume, especially for conference calls. If this were a Zap system, I'd just bump up txgain in their zaptel.conf file... but it isn't. Should I crank the volume of the phones (they're Polycoms), or is there some other, more graceful, system-w

[asterisk-users] ZAP Error: Unable to create channel of type 'Zap'

Hi Group, I have a problem with my FXO interface. It seems that asterisk cannot see any configured channels, only pseudo. Pls see the error and my config below. Rgrds Markus - May 2 15:09:00 NOTICE[17327]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 -

Re: [asterisk-users] Improving Asterisk's DNS support

Per Jessen wrote: Brian Capouch wrote: Kristian is exactly right: there needs to be an architectural fix that will stop EVERYTHING in the server from hanging when lookups are done (and then hang) for a given IAX/SIP peer. It seems to me the best approach would be to report and documen

Re: [asterisk-users] ADSL routers with integrated SIP QoS for otherdevices

On 5/2/07, Tim Koehler <[EMAIL PROTECTED]> wrote: Hi, I can agree for smaller installation/home offices the Linksys WRT series is pretty good (I'm using this at home). I'm using the dd-wrt Firmware (www.dd-wrt.com ) which is also available for plenty other routers. With QoS values set right I a

Re: [asterisk-users] Digital Phones

Steve Totaro wrote: > Citel makes SIP to Digital gateways. I have had poor experience with > them and doubt I would try it again without seeing many improvements > listed in their firmware releases. > > Just to clarify, I had loud bursts of static when first picking up or > originating a call, ph

Re: [asterisk-users] Volume (gain?) on VoIP-only system.

Ken D'Ambrosio wrote: > Hi, all. I've got a customer who's complaining of low volume, especially > for conference calls. If this were a Zap system, I'd just bump up txgain > in their zaptel.conf file... but it isn't. Should I crank the volume of > the phones (they're Polycoms), or is there some

[asterisk-users] Testing Asterisk and Zaptel

Hello Asterisk-Users, My organization is putting together a VoIP setup with Asterisk for a call center, and we currently have two identical machines for redundancy. How do you test your redundant machines? How do you test for load problems? Do you have a strategy or regular plan? I'm most concern

[asterisk-users] Reinvite after DTMF?

Is there a way to do the following scenario? 1) my asterisk box receives an incoming call from a toll free number provider such as nufone, voicepulse, etc. 2) It then dials a number via SIP and outputs a DTMF sequence. ok, that part we do every day. 3) After DTMF though, is it possible to ge

[asterisk-users] SIPGetHeader in Asterisk 1.4

Hi to all, I recently tried to upgrade my Asterisk 1.2 to 1.4. I use quite extensively SIPGetHeader cmd in my Dialplan. But this application is not found in 1.4.2, and I do not see it in 1.4.4code either ??? I could find indeed SIPAddHeader in code. BUT Where did SIPGetHeader go ? any new cmd re

RE: [asterisk-users] Applet?

Steve, yes I know you offer Jiax but you charge for the installation and customization (and he does a great job guys – I’ve used it works well). Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph

RE: Re Re: [asterisk-users] TC400B

The g729 licenses are US$10 a pop and you can buy them directly from www.Digium.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Wednesday, May 02, 2007 5:10 AM To: asterisk-users@lists.digium.com Subject: Re Re: [asterisk-users] T

Re: [asterisk-users] SIPGetHeader in Asterisk 1.4

Hi, You can use function SIP_HEADER instead. See http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header Manu Mehta A R I C E N T Plot-17, Sector 18, Gurgaon 122015, Haryana, India Main +91.124.4095888 x3274 Fax +91.124.4095912 "Jean-Marc Salsa" <[EMAIL PROTECT

Re: [asterisk-users] ADSL routers with integrated SIP QoS for otherdevices

Hi Mike, I'm using a Linksys WRT54GS (older model, I believe V1.1 or 2.0). For the office (exhibitions, testing purposes) I ordered 2 Buffallo routers (directly from the DD-WRT page, preconfigured with DD-WRT). I like to support this guys work and pay a couple Euros more for the router. The B

Re: [asterisk-users] SIPGetHeader in Asterisk 1.4

Thanks ! On 5/2/07, Manu Mehta <[EMAIL PROTECTED]> wrote: Hi, You can use function SIP_HEADER instead. See http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header *Manu Mehta* * * *A R I C E N T* Plot-17, Sector 18, Gurgaon 122015, Haryana, India Main +91.124.4095888

Re: [asterisk-users] MySQL ** DBI connect failed : Too many connections**

Steve Totaro wrote: http://dev.mysql.com/doc/refman/5.0/en/too-many-connections.html Thanks for the link, I now have a few commands that I can try if it happens again. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deser

Re: [asterisk-users] Runaway MOH/mp3123 process?

Alex Balashov wrote: Has anyone noticed a problem with runaway mpg123 processes for music-on-hold eating up ~100% CPU and driving the load on the machine way up? Alex, I recommend dropping mpg123 for native music-on-hold. There are multiple benefits to this, including avoiding all of the pro

[asterisk-users] cdr on channel lacks end, not posted

hello, I'm getting a lot of messages in the log file like : May 2 17:12:07 NOTICE[15877] cdr.c: CDR on channel ' Local/[EMAIL PROTECTED],2' not posted May 2 17:12:07 NOTICE[15877] cdr.c: CDR on channel ' Local/[EMAIL PROTECTED],2' lacks end May 2 17:12:07 WARNING[15901] func_db.c: DB requires

RE: [asterisk-users] Applet?

I have never 'managed' a hotel PBX. That is completely inaccurate. You have no idea what you are talking about. Yes, Estara was $2k dollars according to the client. If there were other provisions in their contract with Estara, I am not aware of them. They have made deals in the past that involved

[asterisk-users] Returning different SIP Hangup Cause

Hi, I would like to return different values/cause to another SIP Server with Hangup cmd. I tried to put different values in Hangup(xx) ... but it always returns the same value ! How can I send back different error cause ? Thanks, Jean-Marc ___ --Band

[asterisk-users] Asterisk-1.4 with agent snmp

Hi, I`m trying to use the agent snmp buit in the asterisk-1.4, but I can`t do this... I used this link to do it: http://www.voipphreak.ca/archives/382 but I can't... Does somebody know how to do this or knows a how-to to do this?? Thank's _

Re: [asterisk-users] using Playback() to play a random sound file

Unless there is some native rand() function available in Asterisk, I'd look into writing a simple AGI using Perl, PHP or Python to return back a random file to Playback(). More information here: http://www.voip-info.org/wiki-Asterisk+AGI HTH - sf Jay Austad wrote: > I've got a directory under /

Re: [asterisk-users] Calls in ulaw, not gsm as desired

So I reloaded things and had just gsm set for 2 of my polycom 501 phones. However, the logs say "No codec found", which leads me to believe that polycom 501 phones can't use gsm. Does anyone have something like this working? If not gsm, is there a better option with these phones over a low bandwidt

Re: [asterisk-users] Asterisk-1.4 with agent snmp

Hi Everton, Which portion of my howto are you having trouble with? Make sure you have compiled the res_snmp when you compiled asterisk. If you want to take this offlist email me at [EMAIL PROTECTED] Thanks, Matt G On 02/05/07, Everton Goularth <[EMAIL PROTECTED]> wrote: Hi, I`m trying to use

RE: [asterisk-users] OT: Capture Asterisk traffic

Sounds like you have an old libpcap. Try using this: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 'udp[2:2] >= 5060 and udp[2:2] <= 65534' This works on one of my machine that has a libpcap that doesn't support portrange. I guess you can't use macros to define the port range. So, you'll hav

RE: [asterisk-users] Digital Phones

Yeah, they still sell the phones: http://products.nortel.com/go/product_content.jsp?segId=0&catId=null&parId=0 &prod_id=8593&locale=en-US I looked around and I can't find that kind of multiplexer. You'd looks like you would need a small digital PBX since you need to be able to define stations and

[asterisk-users] Re: headsets for linksys/sipura phones?

>> I have 4-5 different Nokias, none have a 2.5mm jack. Nothing that >> even remotely resembles a jack. > > The older ones did have 2.5 jacks Interesting - I wonder if this is due to different models being marketed in different places? I have a Nokia 7110 from around 1999 - no jacks whatsoever.

RE: [asterisk-users] CDR changes in 1.4.3?

> I will, in the coming days, look at some of the extraneous CDR's that > are generated, and see what I can do to get rid of them. It's > not always > that simple. > If we ring a phone, for instance, and no-one answers it, is > that truly, > really, something that no-one will ever be, could ever

[asterisk-users] delay in switching between contexts

Hi, I am facing this issue, where I get a delay of aroud five seconds when switching between contexts (through extension.conf) . This is how my extensions looks like. [salesivr] exten => _X.,1,NoOp(Incoming call from user ${EXTEN} and caller id ${CALLERID}) exten => _X.,2,Playback(emptyy) exten

Re: [asterisk-users] Returning different SIP Hangup Cause

On 5/2/07, Jean-Marc Salsa <[EMAIL PROTECTED]> wrote: Hi, I would like to return different values/cause to another SIP Server with Hangup cmd. I tried to put different values in Hangup(xx) ... but it always returns the same value ! How can I send back different error cause ? Thanks, Jean-Marc

Re: [asterisk-users] headsets for linksys/sipura phones?

Interesting - I wonder if this is due to different models being marketed in different places? I have a Nokia 7110 from around 1999 - no jacks whatsoever. I think it'd ude to Nokia and others realizing they could make money grow on trees by not allowing standard headsets to be connected but inst

Re: [asterisk-users] Re: headsets for linksys/sipura phones?

I bought one of the Plantronics 120 or 130, not sure which, and the acoustic conduction within the device causes such an echo that it is unusable with my 842. Even the speakerphone on the 842 is better, and that isn't saying much. John Novack JR Richardson wrote: >> I have 4-5 different Nok

Re: [asterisk-users] Re: headsets for linksys/sipura phones?

JR Richardson wrote: > These work fine with the SPA-942's > Thanks, that's good to know. > http://brandcell.stores.yahoo.net/planm1headha.html > Plantronics M175, we get them for $23, but this site is cheaper. Yeah, that's cheap - I've just ordered two M175s at USD40/each. /Per Jessen, Züri

Re: [asterisk-users] delay in switching between contexts

Danish Samad wrote: Hi, I am facing this issue, where I get a delay of aroud five seconds when switching between contexts (through extension.conf) . This is how my extensions looks like. [salesivr] exten => _X.,1,NoOp(Incoming call from user ${EXTEN} and caller id ${CALLERID}) exten => _X.,2,Pl

[asterisk-users] Re: OT: Capture Asterisk traffic

> "CSB" == CSB <[EMAIL PROTECTED]> writes: CSB> But I want to be a bit more selective: tcpdump -C 100 -W 10 -w CSB> /tmp/tcpdump -i eth1 -s 0 udp and dst port >= 5060 >= redirects stdout to a file named "=". Possibly not what you want. /Benny _

Re: [asterisk-users] delay in switching between contexts

On Wednesday 02 May 2007 11:49 am, Danish Samad wrote: > [salesivr] > exten => _X.,1,NoOp(Incoming call from user ${EXTEN} and caller id > ${CALLERID}) > exten => _X.,2,Playback(emptyy) > exten => _X.,3,Background(Main_Sales) > exten => _X.,4,WaitExten(2) > When I press a digit in _X,3 or _X,4 it

[asterisk-users] Re: How many users can be supported simultaneously?

> "KM" == Knud Müller <[EMAIL PROTECTED]> writes: KM> Hi, there are some interesting figures on KM> http://www.thrallingpenguin.com/articles/asterisk-solaris.htm. It's hard to take them as more than a lower bound on that particular hardware. No attempt is made at figuring out what actually li

RE: [asterisk-users] Re: How many users can be supported simultaneously?

CPU becomes more important if there is a lot of codec conversion. Memory becomes more important in supporting the overall volume of calls. Network resources are obviously limited by bandwidth. A 2.4ghz xeon with 1gb ram can easily handle 80+ calls is there is no codec conversion or the call ter

Re: [asterisk-users] Calls in ulaw, not gsm as desired

Try ilbc if the phone supports (free) or g729 ( better but your asterisk will need license if you want to transcode calls from g729 to other codecs or want to record calls ) . Also check your phones config if its support multiple codecs . . On 02/05/07, Rob Schall <[EMAIL PROTECTED]> wrote:

[asterisk-users] 1.4 memory leak?

Is there a memory leak in asterisk 1.4? The other day with asterisk 1.4.0 I noticed that top was reporting a RES of 106 meg for the asterisk process. Restarting the process brought it down to more like 4 meg, but it grew over time to be 20+. So yesterday morning I upgraded to 1.4.4 in case

[asterisk-users] Call In queue stucks

Hello All, I have a queue with only one agent logged in al the time, but if for some reason the agent cannot pick up the call for 2 full ring, the phone does not ring the 3rd time and all the call in the queue get stuck. Below is my agents.conf [general] persistentagents=yes multiplelogin=no

Re: [asterisk-users] 1.4 memory leak?

With all due respect, I believe you might be a bit paranoid. 10-11M is quite normal for the linux kernel to allocate for asterisk. It's not necessarily what the process is using, but that's just how memory management works within the kernel. What's 10-11M of RAM these days anyway? - sf Adam Mof

Re: [asterisk-users] Calls in ulaw, not gsm as desired

A simple glance at their website will tell you this about the 501 " G.711 μ/A and G.729A (Annex B) configuration " On May 2, 2007, at 12:22 PM, Jaswinder Singh wrote: Try ilbc if the phone supports (free) or g729 ( better but your asterisk will need license if you want to transcode calls f

Re: [asterisk-users] Returning different SIP Hangup Cause

From: "Kristian Kielhofner" <[EMAIL PROTECTED]> Date: Wed, 2 May 2007 11:55:06 -0400 On 5/2/07, Jean-Marc Salsa <[EMAIL PROTECTED]> wrote: Hi, I would like to return different values/cause to another SIP Server with Hangup cmd. I tried to put different values in Hangup(xx) ... but it always ret

[asterisk-users] Daemontools and holidays macro

Hi I've recently released the daemontools scripts I use to run both Asterisk and Flash Operator Panel, and a macro to tell whether today is a holiday or not and jump to different dialplan places accordingly. They are here: daemontools scripts: http://www.bisente.com/blog/2007/04/27/spanish-asteri

[asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation

Hello everyone, As the most established implementer of Asterisk in Singapore, we often have a need to provision hundreds of IP phones at a time. Provisioning each IP phone by editing each xml file individually was very time consuming so we developed a tool internally called IP Phone Provisionin

[asterisk-users] Asterisk locked up

SOFTWARE FreePBX 2.1.3 CentOS 4.4 Asterisk 1.2.13 Zaptel 1.2.10 Sangoma Wanpipe 2.3.4.5 I had an Asterisk server lock up on me today after 95 days of up time. Had to manually kill the Asterisk process and then restart. Nothing out of the ordinary in terms of memory use as far as I could te

RE: [asterisk-users] Reinvite after DTMF?

From: "Wilson Pickett" <[EMAIL PROTECTED]> Date: Wed, 2 May 2007 15:30:21 +0200 Is there a way to do the following scenario? 1) my asterisk box receives an incoming call from a toll free number provider such as nufone, voicepulse, etc. 2) It then dials a number via SIP and outputs a DTMF sequ

[asterisk-users] Large dial plans and variables

I have a large dial plan here with over 3000 lines, and several dozen macros. As it grew, it became apparent that there was some problems. 1. When you pass arguments to a macro in the form of $ARG1, $ARG2 etc, if that macro calls another macro, and passes arguments like this as well, you lose

[asterisk-users] PRI T1 Problems

Sorry for disturbing you, but we have some problems with an installation with multiple (84) T1s from Quest. Now, our Problem is disconnected numbers are reported by sending in- band channel alert message and the B-Channel will have the tri tone and respective message but the line is never "picked

[asterisk-users] allowing call to my pabx every 15 minutes

Hello all, I have a set up that answer my customer. and its working well, however, the number of call to technical dept is what i want to reduce. I want all call to get to voice prompt except that that enter when minutes is 15, 30, 45, 60(in multiples of 15 minutes). how can i achieve this and

[asterisk-users] allowing call every 15mins

Hello all, I have a set up that answer my customer. and its working well, however, the number of call to technical dept is what i want to reduce. I want all call to get to voice prompt except that that enter when minutes is 15, 30, 45, 60(in multiples of 15 minutes). how can i achieve this and

[asterisk-users] Answer A Ringing Queue By Dialing An Extension

Hello I am sure someone has done this and I know I have read about answering extensions before. I just can't find where I read it. What I would like to do is simple in theory. If one of my queues is ringing I would like to be able to dial an extension which immediately connects me to the ne

Re: [asterisk-users] Re: How many users can be supported simultaneously?

Benny Amorsen wrote: "KM" == Knud Müller <[EMAIL PROTECTED]> writes: KM> Hi, there are some interesting figures on KM> http://www.thrallingpenguin.com/articles/asterisk-solaris.htm. It's hard to take them as more than a lower bound on that particular hardware. No attempt is made

Re: [asterisk-users] Large dial plans and variables

Doug Garstang wrote: > I have a large dial plan here with over 3000 lines, and several dozen > macros. As it grew, it became apparent that there was some problems. > > 1. When you pass arguments to a macro in the form of $ARG1, $ARG2 etc, > if that macro calls another macro, and passes argument

Re: [asterisk-users] VPN between Asterisk server and phone client

Concur with Steve: OpenVPN is your friend. At one time, I used "VPN on Demand"-type functionality in my dial plan to trunk a certain subset of calls to a different * server via OpenVPN. This is what that dialplan looked like: [trunkfreecallsviaoffsite] exten => _X.,1,NoOp exten => _X.,n,Playback(

Re: [asterisk-users] Large dial plans and variables

Doug Garstang wrote: > I have a large dial plan here with over 3000 lines, and several dozen > macros. As it grew, it became apparent that there was some problems. > > 1. When you pass arguments to a macro in the form of $ARG1, $ARG2 etc, > if that macro calls another macro, and passes argument

Re: [asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help

Bruce McAlister wrote: > Hi Remco Post, > Having read your patch I suggest you report this bug at bugs.digium.com, it seems to be legit... > However, I think I may have got to the bottom of the issue now. > > What look like was happening is that asterisk was trying to delete any > matching row

Re: [asterisk-users] allowing call to my pabx every 15 minutes

Goke Aruna wrote: > Hello all, > > I have a set up that answer my customer. and its working well, > > however, the number of call to technical dept is what i want to reduce. > > I want all call to get to voice prompt except that that enter when > minutes is 15, 30, 45, 60(in multiples of 15 minu

Re: [asterisk-users] PRI T1 Problems

Unless you're in China, you should advertise you telephone number as: +1 863 248 1195 Is your provider "Quest" or Qwest? Have you removed the disconnected numbers from your dial plan? [EMAIL PROTECTED] wrote: Sorry for disturbing you, but we have some problems with an installation with multi

Re: [asterisk-users] Returning different SIP Hangup Cause

On 5/2/07, Yuan LIU <[EMAIL PROTECTED]> wrote: >From: "Kristian Kielhofner" <[EMAIL PROTECTED]> >Date: Wed, 2 May 2007 11:55:06 -0400 > >On 5/2/07, Jean-Marc Salsa <[EMAIL PROTECTED]> wrote: >>Hi, >> >>I would like to return different values/cause to another SIP Server with >>Hangup cmd. >>I trie

Re: [asterisk-users] allowing call every 15mins

On Wednesday 02 May 2007 3:04 pm, Goke Aruna wrote: > I have a set up that answer my customer. and its working well, > however, the number of call to technical dept is what i want to reduce. > I want all call to get to voice prompt except that that enter when > minutes is 15, 30, 45, 60(in multiple

RE: [asterisk-users] PRI T1 Problems

> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] > Sent: Wednesday, May 02, 2007 2:57 PM > To: asterisk-users@lists.digium.com > Cc: [EMAIL PROTECTED] > Subject: [asterisk-users] PRI T1 Problems > > Sorry for distu

RE: [asterisk-users] allowing call every 15mins

http://www.voip-info.org/wiki/view/Asterisk+tips+openhours Thanks, Steve Totaro http://www.asteriskhelpdesk.com/component/option,com_wrapper/Itemid,37/ KB3OPB > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Goke Aruna > Sent: Wed

Re: [asterisk-users] VPN between Asterisk server and phone client

Kai-Uwe Jensen wrote: Concur with Steve: OpenVPN is your friend. At one time, I used "VPN on Demand"-type functionality in my dial plan to trunk a certain subset of calls to a different * server via OpenVPN. This is what that dialplan looked like: [trunkfreecallsviaoffsite] exten => _X.,1,NoOp e

Re: [asterisk-users] Daemontools and holidays macro

Vicente Aguilar wrote: Hi I've recently released the daemontools scripts I use to run both Asterisk and Flash Operator Panel, and a macro to tell whether today is a holiday or not and jump to different dialplan places accordingly. They are here: daemontools scripts: http://www.bisente.com/blog/

Re: [asterisk-users] Large dial plans and variables

Philipp Kempgen wrote: Doug Garstang wrote: I have a large dial plan here with over 3000 lines, and several dozen macros. As it grew, it became apparent that there was some problems. 1. When you pass arguments to a macro in the form of $ARG1, $ARG2 etc, if that macro calls another macro,

RE: [asterisk-users] allowing call every 15mins

Wow... Now that is customer service... I love it. It's the old "maybe the customer will stop calling if we stop answering the phone" approach. I love it. Anyway, I think you could do it with an AGI call to a script that tracks callerid and last call time. The script could basically decide whether

RE: [asterisk-users] VPN between Asterisk server and phone client

SSL VPN's can be a bit sketchy when it comes to QoS. Usually IPSEC is recommended for udp streaming media. However, people have shown some decent success with SSL VPN's and VoIP. Free S/WAN is a good option if you want to try IPSEC. It should be much more UDP friendly. The following aren't VPN's

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