[Asterisk-Users] Callerid
Hi all, How i can change the CallerId format in plan id? for the example i can see the value of CALLERID variable like lateef 110001 I want to let asterisk do in plain id like lateef any idea? -- Thank You, Code Lover ___ --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] Aastra firmware 1.3.x (Far-End sound level issue)
Hello Robert, I have a similar issue with the Aastra 9133i and recorded .wav voicemail files. The recorded wav is too soft. I need to find a way to boost the volume level. Does anyone have any solutions or ideas? I know this is done by setting the handset tx gain: and handset sidetone gain: parameters via a config file from a tftp server. I was looking for working example values though, as I seem to be unable to get it right. See below message which has surfaced in this list (and I received the exact message when complaining at Aastra): v The audio issue you mention is a known issue as a result of some audio adjustments we've added in 1.3. The audio properties have been adjusted slightly in the 1.3 firmware to reduce side-tone and echo on the local and far-end equipment. In line with these adjustments we have added some configurable parameters that allow users to configure their own audio settings to best suit their needs. These parameters are only configurable via the .cfg file via a TFTP server and use the following syntax: headset tx gain: headset sidetone gain: handset tx gain: handset sidetone gain: handsfree tx gain: Each of these parameters can be adjusted by + / - 10 db Example 1: headset tx gain: -5 (reduce the headset transmit gain by 5 db) Example 2: handset tx gain: 10 (increase the handset transmit gain by 10 db) Please try adjustiing these parameter and let me know if this resolves the problem. ^^ Please let me know if you get the far-end soundlevel to a decent volume. With regards, Taco Scargo ___ --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] New To Asterisk/POTS - Hardware Setup Question
On Dec 21, 2005, at 10:26 AM, [EMAIL PROTECTED] wrote: Regards to All, I recently setup an Asterisk system ([EMAIL PROTECTED]) and it works like a charm so far. It is in a SOHO behind another Linux iptable NAT firewall with no problems. Hopefully this isn't too dumb a question, and its the right place to ask it. The situation is that at this time I have only one incoming PSTN line which I have not yet hooked up (I have a single port FXO wildcard arriving soon for test purposes) which I would like to have available whether the server is available or not. I'm thinking that a Sipura or Grandstream analog adapter with PSTN passthrough is the solution, but I'm not sure, as I'm new to the whole PBX/POTS system. Everything I've seen with passthrough is also a router/gateway. Is that necesary and will it work or is there a better solution? For example, we have regular power outages here at my location lasting anywhere from 1 minute to two hours and if the system is down I would like to still have access to local 911 as well as other local numbers. The obvious thing to do is just unplug one of the phones and plug it directly into the POTS line, but I'm hoping there is a product available that will work with both Asterisk and allow passthrough that will not only transparent, but be less expensive than setting up a UPS system that will hold the server up for an hour or so. A UPS to hold up the adapting device and phone for an extended period would be far cheaper, I think. The grandstream HT-488 has a single port FXO and a single FXS, and also a fail over. So when the power goes off, the phone immediately is hooked up the old fashioned way, and when the power is on it can be used as a VOIP handset. I personally only use the WAN side of this device and ignore the router and LAN side. Seems to work OK so far, although sometimes it seems wiggly too... Time will tell Marty ___ --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] Matching SIP users and peers
On Fri, Dec 23, 2005 at 01:28:14PM -0600, Kevin P. Fleming wrote: C F wrote: Kevin, are you saying that in 1.2 a peer can make calls to asterisk as well, so there is a reason to set the context? If so what is the difference between friend and peer? Yes. All configuration options supported under 'type=user' are also supported under 'type=peer'. interesting... it looks like we can nuke a lot of duplicated code in chan_sip.c then :) cheers luigi ___ --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] tdm400 fxo problem
2005/12/23, Rich Adamson [EMAIL PROTECTED]: I'm really puzzled and I don't know why it is behaving this way. Any hints?Are you sure the phones and system at work are not electronic/digital sets?Yes, I'm sure phones and lines are both analogic (POTS). ___ --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] tdm400 fxo problem
2005/12/24, James B. MacLean [EMAIL PROTECTED]: A problem I had, although different than this, was caused by having/etc/zapata.conf as koolstart when they should have been loopstart.Might be something else to try :).JESI also tried with loopstart in zaptel.conf and zapata.conf, but without success :/ ___ --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] tdm400 fxo problem
2005/12/23, Kerry Garrison [EMAIL PROTECTED]: Possible polarity problem with the jack. -KerrySo I should to check I have +48VDC on the phone line (red wire is +). Right? ___ --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] ISDN/CAPI outgoing calls - weirdness with ringing
Armin,
I changed the dial-string to include flags 'ob' as you mentioned (below)
and now I get the following when I dial a BT phone number
- dial number, get:
Proceeding (in 100) briefly
- after a second or so:
Ringng Destination (in 180)
- double ringing tone:
BT style ringing generated by the exhange
Cisco phone US-style ringing (generated by the phone)
these are overlaid on each other (mixed together)
My hunch is that there's something not right with the call set up sequence
and CAPI handling.
I'll send you some protocol traces off list.
Regards
Mike
- Original Message -
From: Armin Schindler [EMAIL PROTECTED]
To: Michael J. Tubby B.Sc (Hons) G8TIC [EMAIL PROTECTED]
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, December 18, 2005 3:12 PM
Subject: Re: [Asterisk-Users] ISDN/CAPI outgoing calls - weirdness with
ringing
On Fri, 16 Dec 2005, Michael J. Tubby B.Sc (Hons) G8TIC wrote:
All,
I have the following set up:
Fedora Core 4 box (yum updated to current)
Asterisk 1.2.1 + Chan_Capi-cm-0.6.1
AVM C4 card
2 x ISDN2e lines bonded with switchboard number, fax number and 10 x DDI
numbers from British Telecom
14 x Cisco 7960 phones with SIP 7.5
The ISDN lines work in P2P mode and calls are presented with the last 4
digits
only - I land them in a context and branch out from there - everything to
do
with incoming calls works just fine!
I have a problem with outgoing calls that are routed over the BT network
and
the way in which 'ringing' is presented... depending on the called party
number (hence phone provider) I get different results. For example:
a) if I dial another BT number I get a fraction of a second's ring
followed by
silence until the called party answers. The Cisco phone displays:
Proceeding (in 100)
very briefly and is almost immediately over-written by:
Session Progress (in 183)
until the called party answers - at no point is Ringing Destination (in
180)
displayed
b) if I dial an Orange or O2 mobile number I get a second or two's worrth
of
silence [while the Orange network locates the mobile] then the mobile
rings in
the normal way and the Cisco phone plays out US style ringing. When the
number
is dialled the phone displays:
Proceeding (in 100)
when the mobile starts to ring the Cisco phone displays:
Ringng Destination (in 180)
c) if I dial a Bulldog phone number then I get three messages:
Proceeding (in 100) - for a second or so
Session Progress (in 183) - for a couple of seconds
Ringng Destination (in 180) - while the called party's phone rings
d) and the really weird one - if I dial *some* international numbers I
get
both UK (BT) ringing tone overlaid with Asterisk/VoIP (US) ringing tone
I have two ways of dialling out:
1. with an explicit 9 for an outside line -- get dialtone from BT and
then
dial rest of the digits - like a legacy PBX
2. dialing just based on the fact that the extension starts with a zero
so its
an outside call via BT
I have tried all combinations of early B3 connect 'always', 'on success'
and
'never' and it doesn't appear to change things... the relevant part of
extensions.conf is below for completness.
Before I dive in to the next level down:
- is this a known issue?
- is there a solutiuon/workaround/patch/fix
- do I need to get down and dirty with CAPI and SIP debug?
Have you tried CAPI-Dial option 'o' ? Together with 'b' it should give
you progress in any case.
Armin
Mike
;
; external-routes: this is where we get to dial out
;
[external-routes]
;
; outgoing via main ISDN line using explicit 9 for an outside line
; and ISDN eqarly B3 connect (overlap sending) to drop us to the
; BT provided dialtone and work like a normal/legacy phone system -
; we force the caller ID to our exchange number so that DDI's dont
; leak out
;
exten = 9,1,NoOp(ISDN: Pickup outside line (early B3 connect) for:
${CALLERIDNUM})
exten = 9,2,SetCallerId(${THORCOM_MAIN})
exten = 9,3,Dial(CAPI/g1//b)
exten = 9,4,Hangup
;
; implicit trunked call - here we could/should do an ENUM look
; up to see if we can place the call via IP and fall back to BT
; if not... just for now this isn't implemented and we always call
; out via BT!!
;
exten = _0.,1,Dial(CAPI/g1/${EXTEN}/b); early B3
connect
always
;exten = _0.,1,Dial(CAPI/g1/${EXTEN}/B) ; early B3
connect
on success
;exten = _0.,1,Dial(CAPI/g1/${EXTEN}) ; no special
options
exten = _0.,2,Hangup
___
--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
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
[Asterisk-Users] Laptop PCMCIA ISDN card
Hi, I try to install asterisk on my laptop. I have two options for the ISDN card: AVM B1 PCMCIA and Eicon Diva Mobile V90. Has anyone any experiences using one of that cards? Kind regards and merry christmas, Sascha -- [EMAIL PROTECTED] ___ --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] ISDN/CAPI outgoing calls - weirdness with ringing
On Sat, 24 Dec 2005, Michael J. Tubby G8TIC wrote:
I changed the dial-string to include flags 'ob' as you mentioned (below)
and now I get the following when I dial a BT phone number
- dial number, get:
Proceeding (in 100) briefly
- after a second or so:
Ringng Destination (in 180)
- double ringing tone:
BT style ringing generated by the exhange
Cisco phone US-style ringing (generated by the phone)
these are overlaid on each other (mixed together)
My hunch is that there's something not right with the call set up sequence
and CAPI handling.
This is not a problem of CAPI. When you specify 'b' for early-b3, you will
get the tones from the switch. If your phone adds its own tone, even when it
receives progress tones, then it is incorrect (maybe wrong setup).
Armin
I'll send you some protocol traces off list.
Regards
Mike
- Original Message - From: Armin Schindler [EMAIL PROTECTED]
To: Michael J. Tubby B.Sc (Hons) G8TIC [EMAIL PROTECTED]
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, December 18, 2005 3:12 PM
Subject: Re: [Asterisk-Users] ISDN/CAPI outgoing calls - weirdness with
ringing
On Fri, 16 Dec 2005, Michael J. Tubby B.Sc (Hons) G8TIC wrote:
All,
I have the following set up:
Fedora Core 4 box (yum updated to current)
Asterisk 1.2.1 + Chan_Capi-cm-0.6.1
AVM C4 card
2 x ISDN2e lines bonded with switchboard number, fax number and 10 x
DDI
numbers from British Telecom
14 x Cisco 7960 phones with SIP 7.5
The ISDN lines work in P2P mode and calls are presented with the last
4 digits
only - I land them in a context and branch out from there -
everything to do
with incoming calls works just fine!
I have a problem with outgoing calls that are routed over the BT
network and
the way in which 'ringing' is presented... depending on the called
party
number (hence phone provider) I get different results. For example:
a) if I dial another BT number I get a fraction of a second's ring
followed by
silence until the called party answers. The Cisco phone displays:
Proceeding (in 100)
very briefly and is almost immediately over-written by:
Session Progress (in 183)
until the called party answers - at no point is Ringing Destination
(in 180)
displayed
b) if I dial an Orange or O2 mobile number I get a second or two's
worrth of
silence [while the Orange network locates the mobile] then the mobile
rings in
the normal way and the Cisco phone plays out US style ringing. When
the number
is dialled the phone displays:
Proceeding (in 100)
when the mobile starts to ring the Cisco phone displays:
Ringng Destination (in 180)
c) if I dial a Bulldog phone number then I get three messages:
Proceeding (in 100) - for a second or so
Session Progress (in 183) - for a couple of seconds
Ringng Destination (in 180) - while the called party's phone rings
d) and the really weird one - if I dial *some* international numbers
I get
both UK (BT) ringing tone overlaid with Asterisk/VoIP (US) ringing
tone
I have two ways of dialling out:
1. with an explicit 9 for an outside line -- get dialtone from BT
and then
dial rest of the digits - like a legacy PBX
2. dialing just based on the fact that the extension starts with a
zero so its
an outside call via BT
I have tried all combinations of early B3 connect 'always', 'on
success' and
'never' and it doesn't appear to change things... the relevant part
of
extensions.conf is below for completness.
Before I dive in to the next level down:
- is this a known issue?
- is there a solutiuon/workaround/patch/fix
- do I need to get down and dirty with CAPI and SIP debug?
Have you tried CAPI-Dial option 'o' ? Together with 'b' it should give
you progress in any case.
Armin
Mike
;
; external-routes: this is where we get to dial out
;
[external-routes]
;
; outgoing via main ISDN line using explicit 9 for an outside
; line
; and ISDN eqarly B3 connect (overlap sending) to drop us to
; the
; BT provided dialtone and work like a normal/legacy phone
; system -
; we force the caller ID to our exchange number so that DDI's
; dont
; leak out
;
exten = 9,1,NoOp(ISDN: Pickup outside line (early B3 connect) for:
${CALLERIDNUM})
exten = 9,2,SetCallerId(${THORCOM_MAIN})
exten = 9,3,Dial(CAPI/g1//b)
exten = 9,4,Hangup
;
; implicit trunked call - here we could/should do an ENUM look
; up to see if we can place the call via IP and fall back to BT
; if not... just for now this isn't implemented and we always
; call
; out via BT!!
;
exten =
[Asterisk-Users] CAPI and *
Hi, I got the newest asterisk (SVN-trunk-r7413) that compiled fine without any errors or warnings. I got chan_capi 0.4 PRE1 and modified the sources together with a friend so ina way that no error or warning occurs. When I try to load chan_capi the following error is printed and asterisk quits: ,[ capi ]- | [app_capiCD.so]Dec 24 12:57:16 WARNING[9060]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: ast_capi_MessageNumber | Dec 24 12:57:16 WARNING[9060]: loader.c:554 load_modules: Loading module app_capiCD.so failed! ` (Sorry for the long lines, I don't want to break the messaged). I'm not sure where to ask how to solve this, so I'm just asking here. Any help appreciated and have a nice christmas, Sascha -- [EMAIL PROTECTED] ___ --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] CAPI and *
On Sat, 24 Dec 2005, Sascha Andres wrote: Hi, I got the newest asterisk (SVN-trunk-r7413) that compiled fine without any errors or warnings. I got chan_capi 0.4 PRE1 and modified the sources together with a friend so ina way that no error or warning occurs. When I try to load chan_capi the following error is printed and asterisk quits: ,[ capi ]- | [app_capiCD.so]Dec 24 12:57:16 WARNING[9060]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: ast_capi_MessageNumber | Dec 24 12:57:16 WARNING[9060]: loader.c:554 load_modules: Loading module app_capiCD.so failed! ` (Sorry for the long lines, I don't want to break the messaged). I'm not sure where to ask how to solve this, so I'm just asking here. Any help appreciated and have a nice christmas, I suggest you use chan_capi-cm from sourceforge.net instead of old 0.3.5/0.4.0. And when installing a new version, remove old files from installation like app_capi* Armin ___ --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
[Asterisk-Users] Best Voip provider
Hi list: i have a bad experience with voip providers , Any body knows a voip provider i can depend on and to trust with good rates and quality? __ Yahoo! for Good - Make a difference this year. http://brand.yahoo.com/cybergivingweek2005/ ___ --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
[Asterisk-Users] Broken sound Music on hold, , voice prompts good
Hi, When I call to my asterisk server, voice prompts play ok but when it goes to music on hold, sound is all broken. Why is that, is there some ports which Music on hold uses which are not configured properly, or there is some other reason. Zeeshan A Zakaria ___ --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] AMP stuff via CLI?
On Fri, Dec 23, 2005 at 05:18:01PM -0500, Ken D'Ambrosio wrote: Hi, all. I like AMP a lot -- I think it's a nifty program, and it makes a lot of tasks very easy to do. However, as with any GUI, it's hard to automate what it does. So: are there any CLI equivalents for the stuff AMP does? Yeah, yeah, I could hand-edit the files, but AMP uses a MySQL backend, and I'm not even sure what-all goes into there. So if someone's got a) AMP-equivalent CLI stuff, or b) CLI commands that work, that don't have anything to do with AMP, but would still allow for easy automation, then I'd be very, very interested to hear about it. Define what AMP does. AMP is written without any thought whatsoever of seperation of logic from interface. so if you want to create an alternative interface you'll either have to rewrite parts of it or reimplement parts of it in your CLI. Alternatively, consider using other interfaces that do keep this separation. One such interface is http://destar.berlios.de . However if all you need is the CLI interface, simply write a few scripts and be done with that. So again, what do you need from the CLI interface? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] Best Voip provider
On Sat, 2005-12-24 at 04:31 -0800, jonny hashem wrote: Hi list: i have a bad experience with voip providers , Any body knows a voip provider i can depend on and to trust with good rates and quality? best is relative.. There is no single 'best'. The 'best' provider for you may be different for me. Think about the following factors: Good rates to where? You need to find one that has the rates that you want to the places that you actually call. Good rates for what services? You may want specific services, such as a DID in a specific place, ability to connect asterisk directly (as opposed to using an ATA with an FXO), voicemail, etc. Good rates for what volume? Residential calling plans are largely cheaper than business class service. Reliability and quality is also subjective. Depending on where you are located on the internet a provider may be reliable for you that I find unreliable. I may find one highly reliable but you cant connect to it reasonably. Generally this means that you need to get a list of providers and traceroute/ping their servers to see how long it will travel there and back again. this will give you an indication of how fast the server is in relation to you. You may want to give a little more information on the type of service that you want so that recommendations can be tailored to you specifically as opposed to giving you effectively a list of most voip providers. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --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
[Asterisk-Users] Dialling out with clone X100P board
Hi all :
I need a little help please.
I have a clone X100P board. I have it all set up and working (just
testing so far) for incoming calls from PSTN.
For outgoing to PSTN I have a strange problem.
I dial out OK, the Zap channel answers the SIP channel ok, (But I do not
see a Call bridged message, and the call has some strange charateristics.
If I call 123, I can connect to and hear the time clock provided by BT
(I'm in the UK) Is this 'audio before answer'?)
If I call any other external number, eg my cellphone, it never rings,
and after 30 secs or so the Zap channel hangs up.
I have been testing this with a very simple Dial(ZAP/1/${EXTEN}) command.
What should I be looking for in my setup?
Many thanks, and happy Christmas to all.
Roger
___
--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] Dialling out with clone X100P board
I had the same problem at first. Try adding a w or two before the
${EXTEN}. That makes it wait a little bit before sending the DTMF numbers.
Here is the dial() I'm using:
Dial(ZAP/1/ww${EXTEN})
Try it out and see. Let us know if it works.
Ryan
Hi all :
I need a little help please.
I have a clone X100P board. I have it all set up and working (just
testing so far) for incoming calls from PSTN.
For outgoing to PSTN I have a strange problem.
I dial out OK, the Zap channel answers the SIP channel ok, (But I do not
see a Call bridged message, and the call has some strange charateristics.
If I call 123, I can connect to and hear the time clock provided by BT
(I'm in the UK) Is this 'audio before answer'?)
If I call any other external number, eg my cellphone, it never rings,
and after 30 secs or so the Zap channel hangs up.
I have been testing this with a very simple Dial(ZAP/1/${EXTEN}) command.
What should I be looking for in my setup?
Many thanks, and happy Christmas to all.
Roger
___
--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
___
--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] Best Voip provider
hi: Iam using voip providers to get international calls,I provide Callshops with international calls ,my prefered destinations are Germany,Srilanka,Iraq,Egypt,Kuwait , i send calls 700 to 1000 minutes daily. My big problem is bad voice quality that i have experience it with many voip providers. --- trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Sat, 2005-12-24 at 04:31 -0800, jonny hashem wrote: Hi list: i have a bad experience with voip providers , Any body knows a voip provider i can depend on and to trust with good rates and quality? best is relative.. There is no single 'best'. The 'best' provider for you may be different for me. Think about the following factors: Good rates to where? You need to find one that has the rates that you want to the places that you actually call. Good rates for what services? You may want specific services, such as a DID in a specific place, ability to connect asterisk directly (as opposed to using an ATA with an FXO), voicemail, etc. Good rates for what volume? Residential calling plans are largely cheaper than business class service. Reliability and quality is also subjective. Depending on where you are located on the internet a provider may be reliable for you that I find unreliable. I may find one highly reliable but you cant connect to it reasonably. Generally this means that you need to get a list of providers and traceroute/ping their servers to see how long it will travel there and back again. this will give you an indication of how fast the server is in relation to you. You may want to give a little more information on the type of service that you want so that recommendations can be tailored to you specifically as opposed to giving you effectively a list of most voip providers. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group ___ --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 __ Yahoo! for Good - Make a difference this year. http://brand.yahoo.com/cybergivingweek2005/ ___ --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] Best Voip provider
On Sat, 2005-12-24 at 09:21 -0800, chawki hammoud wrote: hi: Iam using voip providers to get international calls,I provide Callshops with international calls ,my prefered destinations are Germany,Srilanka,Iraq,Egypt,Kuwait , i send calls 700 to 1000 minutes daily. My big problem is bad voice quality that i have experience it with many voip providers. I forgot to add this before, you may have better luck posting to asterisk-biz... -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --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] recording queue calls
Dov Bigio wrote:
Hi,
When I set monitor-format=wav49 on file queues.conf for a queue,
Asterisk records calls at /var/spool/asterisk/monitor. But the file
names it users are the call-ids of the calls.
Is there a way to change that, and use information such as date, time,
agent and queue to build the filename?
It would make the localization of such files much more easy.
Other useful that I miss is the capability to to allow the files to be
stored in different directories, such as
/var/spool/asterisk/monitor/queue1, /var/spool/asterisk/monitor/queue2,
and so on, based on the queuename. Is this possible by any means?
Hi,
Yes. All you need to do is use the following in your extension.conf at
the point before you call the queue
SetVar(MONITOR_FILENAME=foo)
or, if you are using 1.2.x
Set(MONITOR_FILENAME=foo)
For example, I have:
Set(MONITOR_FILENAME=${TIMESTAMP}-${CALLERIDNUM}-to-SALES-${UNIQUEID})
and then a little later on:
Queue(salesqueue|t|||60)
in my extensions.conf
Which sets the monitor filename to start with a timestamp, then the CID
of the caller, then the to-SALES is what I use to differentiate
between queues (I'd have a different Set command for a different queue).
I then add the UNIQUEID as a just in case to make absolutely sure
there's no way I'd ever have two files of the same name.
I hope this helps,
Faris.
___
--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] recording queue calls
Faris,
Is there a way to have * send save these in an off-server location? Or
have * e-mail them via smtp and then delete them from the server
automatically?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faris
Raouf
Sent: Saturday, December 24, 2005 10:18 AM
To: Dov Bigio; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] recording queue calls
Dov Bigio wrote:
Hi,
When I set monitor-format=wav49 on file queues.conf for a queue,
Asterisk records calls at /var/spool/asterisk/monitor. But the file
names it users are the call-ids of the calls.
Is there a way to change that, and use information such as date, time,
agent and queue to build the filename?
It would make the localization of such files much more easy.
Other useful that I miss is the capability to to allow the files to be
stored in different directories, such as
/var/spool/asterisk/monitor/queue1,
/var/spool/asterisk/monitor/queue2,
and so on, based on the queuename. Is this possible by any means?
Hi,
Yes. All you need to do is use the following in your extension.conf at
the point before you call the queue
SetVar(MONITOR_FILENAME=foo)
or, if you are using 1.2.x
Set(MONITOR_FILENAME=foo)
For example, I have:
Set(MONITOR_FILENAME=${TIMESTAMP}-${CALLERIDNUM}-to-SALES-${UNIQUEID})
and then a little later on:
Queue(salesqueue|t|||60)
in my extensions.conf
Which sets the monitor filename to start with a timestamp, then the CID
of the caller, then the to-SALES is what I use to differentiate
between queues (I'd have a different Set command for a different queue).
I then add the UNIQUEID as a just in case to make absolutely sure
there's no way I'd ever have two files of the same name.
I hope this helps,
Faris.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.14.7/214 - Release Date:
12/23/2005
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.14.7/214 - Release Date:
12/23/2005
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] in and out recorded audio mixing in queues
Way back I was still on Asterisk 1.0.7, I configured my systems to mix
the incoming and outgoing audio call recordings into one file per call
for both normal calls and queued calls using:
exten =
_9.,1,Monitor(wav49,${TIMESTAMP}-${CALLERIDNUM}-to-${EXTEN:1}-${UNIQUEID},m)
; m option merges audio into one file and deletes the parts
in extensions.conf
and
monitor-join = yes
in queues.conf
As far as I remember this worked perfectly.
But I was only on 1.0.7 for a very short while, and quickly updated. One
system is on 1.0.9 Stable and the others are on the very latest SVN HEAD.
I recently decided to have a bit of a spring clean on the audio files,
and to my horror found that only the very first few files on all
machines were mixed (I presume those were the ones when I was still on
1.0.7) while the rest were all still there as separate -in and -out
files. This is despite the console showing that soxmix was being called
to join the two files and remove the individual parts each time a call
was made or received, and with no errors. Soxmix is installed, and does
work - I can copy and paste the command from the console output at the
command line and the single audio file is then properly created and the
individual in and out parts deleted.
I've just tried changing to using MixMonitor in extensions.conf on the
1.2x machines and this works perfectly for normal outgoing or incoming
calls that don't involve queues.
But this obviously doesn't solve my queues mixing problem on the 1.2
machines, nor any of the mixing problems on the 1.0.9 machine.
Has anyone else come across this issue? Any pointers please?
Thanks,
Faris.
___
--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] recording queue calls
Tom Lynn wrote: Faris, Is there a way to have * send save these in an off-server location? Or have * e-mail them via smtp and then delete them from the server automatically? I'm sure there is a very technical way of doing it. For example if I remember correctly you can set your own script to run to join the two sides of an audio recording (something I tried using to solve the problem I'm having with joining two sides of a conversation, but with no luck). You could add a mail command to the script to do what you want. I'm afraid I don't remember the exact details of how this is done, but I think I came across it when searching for asterisk call recording on Google. There was a full script for an alternative mixing solution. Or you could use rsync, running every hour or every day as a cron job, to synchronise the /var/spool/asterisk/monitor directory on the machine tasking the calls with a second server. e.g. rsync -e ssh -avz /var/spool/asterisk/monitor/ [EMAIL PROTECTED]:~/monitorbackup You'd need to set up a passwordless private/public key combination for this to work automatically though. There may also be issues with the rsync job using too much bandwidth and causing audio quality problems. Hmm... Well, I'm sure someone who know more than me on this topic will pipe up on this! Faris. ___ --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
[Asterisk-Users] PRI outgoing caller ID stopped working
For the last 2.5 years I've been using Asterisk with a Bell Canada PRI with no issues. In the last month or so something has changed; I cannot send *any* caller ID. Incoming works great, and if I place a call through a VOIP provider the caller ID I'm sending shows up. I have not changed any configuration values, and I know that Bell Canada allows me to set Caller ID, as I've been doing it for the last 2.5 years. Looking at the q.931 logs it certainly looks like I'm sending it out, but I'm not fluent enough to tell if it's right. Perhaps someone can assist? The log clearly shows that I'm not blocking outgoing CID. It's also interesting to note that at the same time that this happened, I lost the ability to dial internationally through my PRI. I get a hangupcause of 2: No Route to Transit Network. Again, I have not changed any configurations around that time, but I did move from CVS HEAD to SVN TRUNK. The debug log for an example call is inline. Merry Christmas, everyone. :-) -A. Dec 24 14:01:02 DEBUG[26872] chan_zap.c: Using channel 1 Dec 24 14:01:02 DEBUG[26872] chan_zap.c: -- Making new call for cr 32772 Dec 24 14:01:02 DEBUG[26872] chan_zap.c: Protocol Discriminator: Q.931 (8) len=60 Dec 24 14:01:02 DEBUG[26872] chan_zap.c: Call Ref: len= 2 (reference 4/0x4) (Originator) Dec 24 14:01:02 DEBUG[26872] chan_zap.c: Message type: SETUP (5) Dec 24 14:01:02 DEBUG[26872] chan_zap.c: [Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 04Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 03Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 80Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 90Dec 24 14:01:02 DEBUG[26872] chan_zap.c: a2Dec 24 14:01:02 DEBUG[26872] chan_zap.c: ] Dec 24 14:01:02 DEBUG[26872] chan_zap.c: Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Dec 24 14:01:02 DEBUG[26872] chan_zap.c: Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Dec 24 14:01:02 DEBUG[26872] chan_zap.c: Ext: 1 User information layer 1: u-Law (34) Dec 24 14:01:02 DEBUG[26872] chan_zap.c: [Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 18Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 03Dec 24 14:01:02 DEBUG[26872] chan_zap.c: a9Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 83Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 81Dec 24 14:01:02 DEBUG[26872] chan_zap.c: ] Dec 24 14:01:02 DEBUG[26872] chan_zap.c: Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 Dec 24 14:01:02 DEBUG[26872] chan_zap.c: ChanSel: Reserved Dec 24 14:01:02 DEBUG[26872] chan_zap.c:Ext: 1 Coding: 0 Number Specified Channel Type: 3 Dec 24 14:01:02 DEBUG[26872] chan_zap.c:Ext: 1 Channel: 1 ] Dec 24 14:01:02 DEBUG[26872] chan_zap.c: [Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 1eDec 24 14:01:02 DEBUG[26872] chan_zap.c: 02Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 80Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 83Dec 24 14:01:02 DEBUG[26872] chan_zap.c: ] Dec 24 14:01:02 DEBUG[26872] chan_zap.c: Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Dec 24 14:01:02 DEBUG[26872] chan_zap.c:Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] Dec 24 14:01:02 DEBUG[26872] chan_zap.c: [Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 28Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 0fDec 24 14:01:02 DEBUG[26872] chan_zap.c: b1Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 42Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 45Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 4eDec 24 14:01:02 DEBUG[26872] chan_zap.c: 53Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 48Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 41Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 57Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 20Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 43Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 41Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 4eDec 24 14:01:02 DEBUG[26872] chan_zap.c: 41Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 44Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 41Dec 24 14:01:02 DEBUG[26872] chan_zap.c: ] Dec 24 14:01:02 DEBUG[26872] chan_zap.c: Display (len=15) Charset: 31 [ BENSHAW CANADA ] Dec 24 14:01:02 DEBUG[26872] chan_zap.c: [Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 6cDec 24 14:01:02 DEBUG[26872] chan_zap.c: 0cDec 24 14:01:02 DEBUG[26872] chan_zap.c: 21Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 81Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 35Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 31Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 39Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 32Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 39Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 31Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 35Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 31Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 31Dec 24 14:01:02 DEBUG[26872] chan_zap.c: 32Dec 24 14:01:02 DEBUG[26872] chan_zap.c: ] Dec 24 14:01:02
RE: [Asterisk-Users] recording queue calls
Rsync could happen overnight, but I'm really looking for a solution that removes the recording from the system so as not to kill my limited storage. I'll be running astlinux from a 256mg Compact Flash card and 256meg of USB keydisk space for configs and recordings. I need to move 'em off fast. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Faris Raouf Sent: Saturday, December 24, 2005 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] recording queue calls Tom Lynn wrote: Faris, Is there a way to have * send save these in an off-server location? Or have * e-mail them via smtp and then delete them from the server automatically? I'm sure there is a very technical way of doing it. For example if I remember correctly you can set your own script to run to join the two sides of an audio recording (something I tried using to solve the problem I'm having with joining two sides of a conversation, but with no luck). You could add a mail command to the script to do what you want. I'm afraid I don't remember the exact details of how this is done, but I think I came across it when searching for asterisk call recording on Google. There was a full script for an alternative mixing solution. Or you could use rsync, running every hour or every day as a cron job, to synchronise the /var/spool/asterisk/monitor directory on the machine tasking the calls with a second server. e.g. rsync -e ssh -avz /var/spool/asterisk/monitor/ [EMAIL PROTECTED]:~/monitorbackup You'd need to set up a passwordless private/public key combination for this to work automatically though. There may also be issues with the rsync job using too much bandwidth and causing audio quality problems. Hmm... Well, I'm sure someone who know more than me on this topic will pipe up on this! Faris. ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.7/214 - Release Date: 12/23/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.7/214 - Release Date: 12/23/2005 ___ --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] Broken sound Music on hold, , voice prompts good
sound is all broken? WTF is that meant to mean. Does it play or doesn't it? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Zeeshan wrote: Hi, When I call to my asterisk server, voice prompts play ok but when it goes to music on hold, sound is all broken. Why is that, is there some ports which Music on hold uses which are not configured properly, or there is some other reason. Zeeshan A Zakaria ___ --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 ___ --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] Callerid
Assuming its a SIP based device [110001] user=something allow=whatever callerid= lateef Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Code Lover wrote: Hi all, How i can change the CallerId format in plan id? for the example i can see the value of CALLERID variable like lateef 110001 I want to let asterisk do in plain id like lateef any idea? -- Thank You, Code Lover ___ --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 ___ --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] Broken sound Music on hold, , voice prompts good
It means that Music on Hold works but listener listens it in bits and pieces. Zeeshan A Zakaria -Original Message- From: Mark Phillips [mailto:[EMAIL PROTECTED] Sent: Saturday, December 24, 2005 2:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Broken sound Music on hold, , voice prompts good sound is all broken? WTF is that meant to mean. Does it play or doesn't it? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Zeeshan wrote: Hi, When I call to my asterisk server, voice prompts play ok but when it goes to music on hold, sound is all broken. Why is that, is there some ports which Music on hold uses which are not configured properly, or there is some other reason. Zeeshan A Zakaria ___ --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 ___ --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 ___ --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] Broken sound Music on hold, , voice prompts good
WTF is that meant to mean. Does it play or doesn't it? Calm down. It probably means that it's breaking up while it is playing. But let the OP explain... no need to discourage him like that. ___ --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] PRI outgoing caller ID stopped working
Andrew Kohlsmith wrote: For the last 2.5 years I've been using Asterisk with a Bell Canada PRI with no issues. In the last month or so something has changed; I cannot send *any* caller ID. Incoming works great, and if I place a call through a VOIP provider the caller ID I'm sending shows up. Interestingly, some systems I manage also began exhibiting this behavior in the past ten days or so. I have been working with the telco and they too show the Calling Number being received as expected over the PRI, but yet the far end receives 'Unknown' or 'Out of Area' depending on their CLID display device. I will continue to try to debug it, but I can't back down the code on that box to an older version for comparison of the PRI traffic; if you can do so, that would be most helpful. ___ --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
[Asterisk-Users] System(...) but how to pass parameters?
How to pass some parameters to shell script, invoked in CLI through application system(...)? I want to do some logging of incoming CID-s to file. Is there some other method to do this? ___ --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
[Asterisk-Users] System(...) but how to pass parameters?
Not in CLI, Invoked in extensions.conf:
exten = s,1,system(/usr/bin/logscript) ;and how to pass some parameters
here?
if I do somenhing like:
exten = s,1,system(/usr/bin/logscript,${CALLERID},pstn)
then I get error.
- Original Message -
From: Pisac [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: ned 25. dec 2005 1:08
Subject: [Asterisk-Users] System(...) but how to pass parameters?
How to pass some parameters to shell script, invoked in CLI through
application system(...)?
I want to do some logging of incoming CID-s to file. Is there some
other
method to do this?
___
--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
___
--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] Merry Xmas to everybody!
Greetings from Lima Peru Carlos Rojas On 12/23/05, Guillermo Salas M [EMAIL PROTECTED] wrote: Xmas is tomorrow at my country.. Merry Xmast to all :)Greetings from Ecuador - South America ;) On Fri, 2005-12-23 at 19:20 -0500, tracinet wrote: Nothing wrong at all - this is the Merry Christmas thread.Feel free to start a Happy Kwanza thread :) Merry Christmas to all !! On 12/23/05, Mark Phillips [EMAIL PROTECTED] wrote: What's wrong with us that celebrate Kwanza? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Dmitry Ivanov wrote: On Friday 23 December 2005 10:22, Mauro Zanin wrote: Hi everybody, no issues this time. Only stopped to say: Merry Christmas and Happy New Year. Yes, Merry Christmas, Happy New Year and Hanukkah :) Just received nice postcard from Digium :) ___ --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 ___ --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 ___ --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 --Guillermo Salas M.Telconet S.A. MantaCalle 15 y Av. 24 Esq.Phone : 593 5 262 8071Mobile: 593 9 985 5138SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED]www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902Soporte en Linea en http://www.manta.telconet.netPlease avoid sending me Word or PowerPoint attachments.See http://www.fsf.org/philosophy/no-word-attachments.html___--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 ___ --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] System(...) but how to pass parameters?
On Sun, 2005-12-25 at 01:08 +0100, Pisac wrote: How to pass some parameters to shell script, invoked in CLI through application system(...)? I want to do some logging of incoming CID-s to file. Is there some other method to do this? Other than through system? Or did you want information on using system to do this? You could do system(echo some parameters/some/logfile) if in ael odds are you will need to escape the / marks with \ (ie \/some \/logfile) If you want another way to log specific data and only that data to a file, you can call an AGI to do it, you can write a logging module for asterisk, there may be other ways as well but that is the first that comes to mind. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --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] System(...) but how to pass parameters?
On Sun, 2005-12-25 at 01:22 +0100, Pisac wrote:
Not in CLI, Invoked in extensions.conf:
exten = s,1,system(/usr/bin/logscript) ;and how to pass some parameters
here?
if I do somenhing like:
exten = s,1,system(/usr/bin/logscript,${CALLERID},pstn)
then I get error.
The singlke argument to system() is what you would type at the command
line (ie your shell prompt). You get errors because you are using
commas instead of spaces.
try system(/usr/bin/logscript ${CALLERID} pstn)
--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605 Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Latest Source
Instal subversion package, in your linux to be abale to use svn. Regards Carlos Rojas On 12/20/05, Douglas Garstang [EMAIL PROTECTED] wrote: No idea on this. Can't find it on digium's web site. How do we download the latest source for Asterisk? Looks like they switched to SVN from CVS? Never used it... is there a Linux client for it? Doug. -Original Message-From: Rehan Ahmed [mailto: [EMAIL PROTECTED]]Sent: Tuesday, December 20, 2005 4:57 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Affordable IP Phones for Asterisk Hello Dakota, I have a few that i can ship you from vida21.com for 70$ each they with me in the US. The client is now using cisco i have 70 pcs with me Rehan On 12/20/05, Dakota [EMAIL PROTECTED] wrote: Are there any IP Phones that can work with Asterisk, that cost less than $60?if so, what's the model/manufacturer? Dakota___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Rehan Ahmed AllahWala http://www.SuperTec.com - Tommrow's Technology, Today.http://www.didx.net - DID Number Exchange and Peering Service. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
[Asterisk-Users] Busy signal for incoming calls from broadvoice
When someone calls me via BroadVoice, they get a busy signal. My * box is behind a NAT firewall. I have enabled port forwarding of UDP 5060 and 1:2 to the * box. I added nat=yes externalip and localnet to the sip.conf under [general]. It still doesn't work. I just want * to be able to answer the phone and send the caller to voicemail directly. What could be the problem? ___ --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] Busy signal for incoming calls from broadvoice
On Sat, 2005-12-24 at 20:17 -0500, Robert La Ferla wrote: When someone calls me via BroadVoice, they get a busy signal. My * box is behind a NAT firewall. I have enabled port forwarding of UDP 5060 and 1:2 to the * box. I added nat=yes externalip and localnet to the sip.conf under [general]. It still doesn't work. I just want * to be able to answer the phone and send the caller to voicemail directly. What could be the problem? Did this start today or so? Rumor has it that BV is broken right now and others are having problems completing calls. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --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] Busy signal for incoming calls from broadvoice
trixter aka Bret McDanel wrote: On Sat, 2005-12-24 at 20:17 -0500, Robert La Ferla wrote: When someone calls me via BroadVoice, they get a busy signal. My * box is behind a NAT firewall. I have enabled port forwarding of UDP 5060 and 1:2 to the * box. I added nat=yes externalip and localnet to the sip.conf under [general]. It still doesn't work. I just want * to be able to answer the phone and send the caller to voicemail directly. What could be the problem? Did this start today or so? Rumor has it that BV is broken right now and others are having problems completing calls. My bv number works at the moment ___ --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
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 148
Need help Install asterisk-oh323hi everybodyi have just installed asterisk 1.2.1 and added asterisk-oh323-0.7.3installed also pwlib1.5.2 and openh323_1.12.2 (the Mimas patches 2)i did followed all instructions but when i it created this problem. hope you could help me.thanks so much in advance.[EMAIL PROTECTED]:~/voip/asterisk-oh323-0.7.3# makefor x in wrapper asterisk-driver; do make -C $x build || exit 1 ; donemake[1]: Entering directory `/root/voip/asterisk- oh323-0.7.3/wrapper'./check_ver /root/pwlib pwlib./check_ver /root/voip/openh323_Mimas_patch2 openh323ar rc liboh323wrap_s.a wrapper_misc.o asteriskaudio.o wrapconnection.o wrapendpoint.o wrapper.o wrapcaps.o wrapgkserver.omake[1]: Leaving directory `/root/voip/asterisk-oh323-0.7.3/wrapper'make[1]: Entering directory `/root/voip/asterisk-oh323-0.7.3/asterisk-driver'gcc -Wall -march=i686 -pipe -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -fPIC -g -O6 -fomit-frame-pointer -I/usr/src/asterisk/include -I../wrapper -c -o chan_oh323.o chan_oh323.c chan_oh323.c:49:22: asterisk.h: No such file or directorychan_oh323.c: In function `oh323_show_channels':chan_oh323.c:1107: warning: implicit declaration of function `snprintf'chan_oh323.c: In function `oh323_exception': chan_oh323.c:1521: warning: implicit declaration of function `printf'chan_oh323.c: In function `oh323_codecset2str':chan_oh323.c:3113: warning: implicit declaration of function `sprintf'chan_oh323.c: In function `reload_config': chan_oh323.c:4675: warning: implicit declaration of function `sscanf'chan_oh323.c: At top level:chan_oh323.c:3242: warning: `update_call_ids' defined but not usedmake[1]: *** [chan_oh323.o] Error 1make[1]: Leaving directory `/root/voip/asterisk- oh323-0.7.3/asterisk-driver'make: *** [subdirs_build] Error 1[EMAIL PROTECTED]:~/voip/asterisk-oh323-0.7.3# ___ --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] Busy signal for incoming calls from broadvoice
trixter aka Bret McDanel wrote: On Sat, 2005-12-24 at 20:17 -0500, Robert La Ferla wrote: When someone calls me via BroadVoice, they get a busy signal. My * box is behind a NAT firewall. I have enabled port forwarding of UDP 5060 and 1:2 to the * box. I added nat=yes externalip and localnet to the sip.conf under [general]. It still doesn't work. I just want * to be able to answer the phone and send the caller to voicemail directly. What could be the problem? Did this start today or so? Rumor has it that BV is broken right now and others are having problems completing calls. I haven't been with BroadVoice long enough for my data to be relevent. i.e. less than 24 hrs. I tried calling their tech support and wasn't able to reach a live person and my call got dropped a few times. Haven't had problems otherwise. ___ --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] Callerid
Hi, I am using SIPS softphoe. and i tested with another SIP Gatekeeper and i can see callerid in plain format. But when i am trying using Asterisk it is apearing callerid, username. So i don't think this is from client side or softphone. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com ___ --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] Best Voip provider
hello, You can check this compnay. http://www.hatif.com Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! for Good - Make a difference this year. http://brand.yahoo.com/cybergivingweek2005/ ___ --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] Virtual Memory Usage
On Fri, Dec 23, 2005 at 06:59:50AM -0600, Rich Adamson wrote: This is another thing: Linux tends to use the availble free memory for IO buffers, disk cache and such. So in the output of 'free', look at the second line. I'm not the OP, but for those of us that are not considered strong sys admin's (but have been around and using linux since early 90's), could you provide us with a short list of what we should be looking at to monitor/manage memory? (More of an educational thingie.) When I run 'free' on a small 15 user system as an example, I see: total used free sharedbuffers cached Mem:451172 449788 1384 0 121740 63036 -/+ buffers/cache: 265012 186160 Swap: 917496320 917176 which kind of implies that I should probably add mem to this box. Am I approaching this backasswards or drawing an incorrect conclusion? You currently have just 320kB of memory swapped out. That can easily be parts of precesses that will mostly be unused be used (e.g: the console login processes). Your system has lots of free memory, as you can see in the second line. However the OS does not let that free memory lie still: it is used for IO buffers and caching. No need to get extra memory. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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
