[asterisk-users] not succeeding to hide callerid with outbound calls
Hi All, In my dialplan and standard asterisk CLI logging i see that i am able to restrict the callerid when dialing out with asterisk. however, on the receiving phone, the callerid is still displayed. When i increment the logging of the pri with pri set debug on span 1 on the CLI i also get the lower level debugging info from the pri. From here it looks like the SET CALLERPRES()=prohib is not working as expected... we see: Presentation: Presentation permitted We use asterisk 1.6 with DAHDI and a PRI ISDN 30 line in the Netherlands. Can anyone help me sorting out this issue?? Thanks in advance! -- Executing [...@macro-transfer:25] NoOp(SIP/joostkuif-0003, geheim) in new stack -- Executing [...@macro-transfer:26] NoOp(SIP/joostkuif-0003, voor de SET CALLERPRES() = allowed_not_screened) in new stack -- Executing [...@macro-transfer:27] NoOp(SIP/joostkuif-0003, CALLINGPRES = 0) in new stack -- Executing [...@macro-transfer:28] Set(SIP/joostkuif-0003, CALLERPRES()=prohib) in new stack -- Executing [...@macro-transfer:29] NoOp(SIP/joostkuif-0003, na de SET CALLERPRES() = prohib) in new stack -- Executing [...@macro-transfer:30] NoOp(SIP/joostkuif-0003, CALLINGPRES is nu = 35) in new stack -- Executing [...@macro-transfer:31] Goto(SIP/joostkuif-0003, dial) in new stack -- Goto (macro-transfer,s,36) -- Executing [...@macro-transfer:36] NoOp(SIP/joostkuif-0003, dial) in new stack -- Executing [...@macro-transfer:37] Dial(SIP/joostkuif-0003, DAHDI/g1/003164616,10) in new stack -- Making new call for cref 32772 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=45 TEI=0 Call Ref: len= 2 (reference 4/0x4) (Sent from originator) Message Type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) User information layer 1: A-Law (35) [18 03 a1 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Preferred Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 Type: CPE] [6c 0b 21 81 36 34 36 31 36 30 35 39 30] Calling Number (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '64616' ] [70 0e 80 30 30 33 31 36 34 36 31 36 30 35 39 30] Called Number (len=16) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '003164616' ] [a1] Sending Complete (len= 1) q931.c:4877 q931_setup: Call 32772 enters state 1 (Call Initiated). Hold state: Idle -- Called g1/003164616 Protocol Discriminator: Q.931 (8) len=10 TEI=0 Call Ref: len= 2 (reference 4/0x4) (Sent to originator) Message Type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Exclusive Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 Type: CPE] Received message for call 0x2c4e93f0 on 0xbead2a0 TEI/SAPI 0/0, call-pri is 0xbead2a0 TEI/SAPI 0/0 -- Processing IE 24 (cs0, Channel Identification) q931.c:6916 post_handle_q931_message: Call 32772 enters state 3 (Outgoing Call Proceeding). Hold state: Idle -- DAHDI/1-1 is proceeding passing it to SIP/joostkuif-0003 Protocol Discriminator: Q.931 (8) len=9 TEI=0 Call Ref: len= 2 (reference 4/0x4) (Sent to originator) Message Type: ALERTING (1) [1e 02 84 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] Received message for call 0x2c4e93f0 on 0xbead2a0 TEI/SAPI 0/0, call-pri is 0xbead2a0 TEI/SAPI 0/0 -- Processing IE 30 (cs0, Progress Indicator) q931.c:6806 post_handle_q931_message: Call 32772 enters state 4 (Call Delivered). Hold state: Idle -- DAHDI/1-1 is ringing -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to finish an AGI
Any particular reason you don't want to put the logic of the macro in your
AGI?
Yes...i've no idea how to do it...it's a PERL script, i'm already checking
how to do this...but it will be a little complicated :(
2010/9/3 Steve Edwards asterisk@sedwards.com
On Thu, 2 Sep 2010, Danny Dias wrote:
Sorry for my poor explanation...what i'm trying to do is to invoke a Macro
from my AGI, like this:
$agi-exec(Macro,check-call-limit);
If the Macro checks that the group_name is bigger than a number specified
for every peer with setvar it should Hangup the call (frobidden,1 in the
Gotoif...) but this
is not happening, the AGI always continue with is process and it doesn´t
play attention to the Hangup in the macro, the macro is here:
[macro-check-call-limit]
exten = s,1,Set(group_name=out_calls_user_${SIPCHANINFO(peername)})
exten = s,n,Set(GROUP()=${group_name})
exten = s,n,GotoIf($[${GROUP_COUNT(${group_name})}
${MAX_OUT_CALLS_PER_USER}] forbidden,1)
; EXITO:
exten = s,n,MacroExit
; FRACASO:
exten = forbidden,1,NoOp(*** llamada saliente bloqueada: el usuario
${SIPCHANINFO(peername)} tiene actualmente
${MATH(${GROUP_COUNT(${group_name})})-1,int)} llamadas
salientes)
exten = forbidden,n,Hangup(21) ; ISUP 21 = SIP 403 (Forbidden)
The concept of calling a macro from within an AGI seem convoluted, but may
work. I've never tried it.
Any particular reason you don't want to put the logic of the macro in your
AGI?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Wanted: UK-specific hardware recommendations (FXO and FXS)
I have a pair of Asterisk servers which are happily routeing VoIP calls. I want to hook one of them to the PSTN. Given that I am in the UK, what is a reasonably easily-available device to provide an FXO interface from a Linux box, with a minimum of faffing around with drivers? Just one line is needed, though in theory two might eventually be useful. My usual white-box hardware suppliers don't seem to play in this field. Also: I've heard good things about the PAP2T for getting analogue handsets to talk to a VoIP server. But all the ones I can see on eBay are PAP2T-NA models. Will these work with British handsets? (Obviously with a plug adaptor to put the BT jack into an RJ11 socket, but that's relatively easy to arrange.) Roger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanted: UK-specific hardware recommendations (FXO and FXS)
On 3 Sep 2010, at 10:07, Roger Burton West wrote: Also: I've heard good things about the PAP2T for getting analogue handsets to talk to a VoIP server. But all the ones I can see on eBay are PAP2T-NA models. Will these work with British handsets? (Obviously with a plug adaptor to put the BT jack into an RJ11 socket, but that's relatively easy to arrange.) PAP2 was discontinued a long time ago. Use a 2102. There are fake PAP2's out there so avoid. 2102 works fine with a UK handset for me. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] openvz
Can i run asterisk on a openvz vps or do i need a kernel? I dont use dadi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] openvz
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias Subject: [asterisk-users] openvz Can i run asterisk on a openvz vps or do i need a kernel? I dont use dadi Blind Answer - you should be able to; Asterisk doesn't rebuild the kernel. You might have to get some kernel source using ZYPPER (in caps so Outlook express doesn't change it to zipper). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to use MYSQL(Set timeout x)
I use Asterisk 1.6.2.11 and this is my dialplan:
[test]
exten = ,1,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten = ,n,Answer()
exten = ,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten = ,n,PlayBack(hello-world)
exten = ,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten = ,n,MYSQL(Set timeout 2)
exten = ,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten = ,n,MYSQL(Connect connid localhost user pass asterisk)
exten = ,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten = ,n,MYSQL(Query resultid ${connid} SELECT SLEEP(10))
exten = ,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten = ,n,MYSQL(Fetch fetchid ${resultid} RESULT)
exten = ,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten = ,n,MYSQL(Clear ${resultid})
exten = ,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten = ,n,MYSQL(Disconnect ${connid})
exten = ,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten = ,n,NoOp(Result: ${RESULT})
exten = ,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten = ,n,Hangup()
When i call to this is the CLI output:
Connected to Asterisk 1.6.2.11 currently running on Asterisk (pid = 2092)
Verbosity is at least 2147483647
Asterisk*CLI
== Using SIP RTP CoS mark 5
-- Executing [9...@test:1] NoOp(SIP/test-, 2010-09-03 14:57:35)
in new stack
-- Executing [9...@test:2] Answer(SIP/test-, ) in new stack
-- Executing [9...@test:3] NoOp(SIP/test-, 2010-09-03 14:57:35)
in new stack
-- Executing [9...@test:4] Playback(SIP/test-, hello-world) in
new stack
-- SIP/test- Playing 'hello-world.gsm' (language 'en')
-- Executing [9...@test:5] NoOp(SIP/test-, 2010-09-03 14:57:36)
in new stack
-- Executing [9...@test:6] MYSQL(SIP/test-, Set timeout 2) in
new stack
-- Executing [9...@test:7] NoOp(SIP/test-, 2010-09-03 14:57:36)
in new stack
-- Executing [9...@test:8] MYSQL(SIP/test-, Connect connid
localhost user pass asterisk) in new stack
-- Executing [9...@test:9] NoOp(SIP/test-, 2010-09-03 14:57:37)
in new stack
-- Executing [9...@test:10] MYSQL(SIP/test-, Query resultid 1
SELECT SLEEP(10)) in new stack
-- Executing [9...@test:11] NoOp(SIP/test-, 2010-09-03
14:57:47)
in new stack
-- Executing [9...@test:12] MYSQL(SIP/test-, Fetch fetchid 2
RESULT) in new stack
-- Executing [9...@test:13] NoOp(SIP/test-, 2010-09-03
14:57:47)
in new stack
-- Executing [9...@test:14] MYSQL(SIP/test-, Clear 2) in new
stack
-- Executing [9...@test:15] NoOp(SIP/test-, 2010-09-03
14:57:47)
in new stack
-- Executing [9...@test:16] MYSQL(SIP/test-, Disconnect 1) in
new stack
-- Executing [9...@test:17] NoOp(SIP/test-, 2010-09-03
14:57:47)
in new stack
-- Executing [9...@test:18] NoOp(SIP/test-, Result: 0) in new
stack
-- Executing [9...@test:19] NoOp(SIP/test-, 2010-09-03
14:57:47)
in new stack
-- Executing [9...@test:20] Hangup(SIP/test-, ) in new stack
== Spawn extension (test, , 20) exited non-zero on 'SIP/test-'
Asterisk*CLI
According to Asterisk*CLI core show application MYSQL:
MYSQL(Set timeout num)
Set the connection timeout, in seconds.
As you see the SELECT SLEEP(10) query took 10 seconds and the MySQL timeout
was set to 2 seconds.
I think the timeout should have ended the execution of the query.
Does anyone know why it didn't?
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Re: [asterisk-users] openvz
On Fri, Sep 03, 2010 at 03:11:39PM +0200, mattias wrote: Can i run asterisk on a openvz vps or do i need a kernel? I dont use dadi I don't expect any problem. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [draft] DAHDI-linux DAHDI-tools 2.4.0 Release Announcement
The Asterisk Development Team is pleased to announce the release of DAHDI-Linux and DAHDI-Tools version 2.4.0. DAHDI-Linux 2.4.0, DAHDI-Tools 2.4.0, and DAHDI-Linux-Complete are available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete In addition to several bug fixes, the most significant changes from the 2.3.0 release are: General DAHDI Changes: * Added DAHDI_MAINT_ALARM_SIM maintenance mode for drivers that support alarm simulation (wct4xxp). This is only used by dahdi_maint and doesn't change the ABI. * Span callbacks are moved out of the dahdi_span structure potentially saving memory when a single driver implements multiple spans. Updated Drivers: * wctdm24xxp, wcte12xp: Fix bug when moving to memory mapped registers where the interrupt handler was run twice for every interrupt. * wctdm24xxp, wcte12xp: Processing moved back to interrupt handler. (Closes issue #17289 Reported by alecdavis) * wctdm24xxp, wcte12xp: Update VPMADT032 firmware to 1.25. Contains improvements to prevent loss of convergence when signal levels go over a certain threshold and for handling line condition changes. * wctdm24xxp: Fix race conditions/improvements in FXS line feed register handling. (Closes issues #17724 and #17764. Reported and patched by alecdavis) * wctdm24xxp: Added companding module parameter to replace alawoverride. When BRI modules are installed on a Hx8 board alaw is the default companding so change the semantics to just allow the companding to be forced as opposed to overriding a default. The default is auto which means alaw if there are BRI modules, otherwise ulaw. * wctdm24xxp: Set 'spantype' for digital spans so that they can be displayed with dahdi_scan. * wcte12xp: dahdi_cfg does not need to be called twice when using RBS signalling. * wcte12xp: Loopback module parameter removed since 'dahdi_maint' can now put the spans in digital loopback. * wct4xxp: Add 'latency', 'max_latency', and 'ms_per_irq' module parameters to set expected latency conditions when using Gen5 firmware. * wct4xxp: Added support for network loopback modes via dahdi_maint. * wct4xxp: Which span is providing card timing is now exported via sysfs. * wcb4xxp: Fixed pulse mask for improved TBR3 compliance. * wcb4xxp: Added pci-ids for Junghanns PCI-E cards. * wcb4xxp: Added 'companding' module parameter. * wcb4xxp: Fixed bug when using automatic timing sync. * wcb4xxp: Which span is providing card timing is now exported via sysfs. * wctdm: Added configurable debounce to support old rotary phones. (Closes issue #16339. Reported by alecdavis patch by tilghman.) * xpp: FXS: support VMWI config from Asterisk = 1.6.1 PRI: - PRI Astribanks always sync AB (and independent) - don't send duplicates in E1 as in D4 - Reduce noise at E1 startup. - T1 CAS fixes. PIC 4 rev. 7381: fix T1 returning signaling register in non-CAS Changes to dahdi-tools: * dahdi_maint: Added support for simulating alarm conditions. * dahdi_scan: Report more detailed alarm information. * xpp_fxloader: - Load firmware in the background - Support 1163 twinstar devices - A delay loop for older kernels (e.g. 2.6.18) * astribank_is_starting does not depend on libusb. * Allow using CONNECTOR/LABEL in genconf_parameters for pri_termtype For a full list of changes in these releases, please see the ChangeLogs at http://svn.asterisk.org/svn/dahdi/linux/tags/2.4.0/ChangeLog and http://svn.asterisk.org/svn/dahdi/tools/tags/2.4.0/ChangeLog Issues found in these releases can be reported at http://issues.asterisk.org Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Faxes
We have asterisk connected to the PSTN VIA TE410P (ANSI SS7 to T1 PRI) cards. I am having trouble completing faxes. Carrier send calls to me using SIP. Any recommendation to have some success with Fax. We trying using T.38 pass through and using G711U codec. Asterisk Version 1.6.1.1 Thanks, Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] openvz
On Fri, Sep 03, 2010 at 03:11:39PM +0200, mattias wrote: Can i run asterisk on a openvz vps or do i need a kernel? I dont use dadi I don't expect any problem. Absolutely right: 1.6.x works fine with OpenVZ and Virtuozzo out of the box as long as you don't need any hardware interfaces. You don't need any kernel sources, though these are available. You basically just install exactly as you would on any other system. There is no problem with timing (e.g. MOH) - at least none that I've ever come across and at least not for 1.6.x which is what I've used under Virtuozzo for some time now. If you want to install Digium g.729 licenses you need to do some small configuration changes but these are easy to do and full instructions are on the OpenVZ Wiki somewhere. Faris. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [SOLVED ]How to finish an AGI
I've done it ;)
This is what i did:
In the Macro:
[macro-check-call-limit-mercurios]
exten = s,1,Set(group_name=out_calls_user_${SIPCHANINFO(peername)})
exten = s,n,Set(GROUP()=${group_name})
exten = s,n,GotoIf($[${GROUP_COUNT(${group_name})}
${MAX_OUT_CALLS_PER_USER}]?forbidden,1)
; EXITO:
exten = s,n,MacroExit
; FRACASO:
exten = forbidden,1,NoOp(*** llamada saliente bloqueada: el usuario
${SIPCHANINFO(peername)} tiene actualmente
${MATH(${GROUP_COUNT(${group_name})})-1,int)} llamadas salientes)
exten = forbidden,n,Set(toagi=1);en este caso la llamada no la cuelga la
macro como en endpoints la cuelga el agi.
exten = forbidden,n,Hangup()
Then, in the AGI:
AGI:
$AGI-exec(Macro,check-call-limit-mercurios);
$limitada = $AGI-get_variable('toagi');
if ($limitada eq '1'){
$AGI-verbose(Superado el limite de llamadas salientes out.agi tira
el canal);
$AGI-hangup($chann);
}
#
#
By the way, is it necessary to Hangup the Macro if the AGI is already doing
this?
BR ;)
2010/9/3 Steve Edwards asterisk@sedwards.com
On Thu, 2 Sep 2010, Danny Dias wrote:
Sorry for my poor explanation...what i'm trying to do is to invoke a Macro
from my AGI, like this:
$agi-exec(Macro,check-call-limit);
If the Macro checks that the group_name is bigger than a number specified
for every peer with setvar it should Hangup the call (frobidden,1 in the
Gotoif...) but this
is not happening, the AGI always continue with is process and it doesn´t
play attention to the Hangup in the macro, the macro is here:
[macro-check-call-limit]
exten = s,1,Set(group_name=out_calls_user_${SIPCHANINFO(peername)})
exten = s,n,Set(GROUP()=${group_name})
exten = s,n,GotoIf($[${GROUP_COUNT(${group_name})}
${MAX_OUT_CALLS_PER_USER}] forbidden,1)
; EXITO:
exten = s,n,MacroExit
; FRACASO:
exten = forbidden,1,NoOp(*** llamada saliente bloqueada: el usuario
${SIPCHANINFO(peername)} tiene actualmente
${MATH(${GROUP_COUNT(${group_name})})-1,int)} llamadas
salientes)
exten = forbidden,n,Hangup(21) ; ISUP 21 = SIP 403 (Forbidden)
The concept of calling a macro from within an AGI seem convoluted, but may
work. I've never tried it.
Any particular reason you don't want to put the logic of the macro in your
AGI?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Faxes
On Fri, Sep 3, 2010 at 10:49 AM, dave george dgeo...@teletoneinc.com wrote: We have asterisk connected to the PSTN VIA TE410P (ANSI SS7 to T1 PRI) cards. I am having trouble completing faxes. Carrier send calls to me using SIP. Any recommendation to have some success with Fax. We trying using T.38 pass through and using G711U codec. Asterisk Version 1.6.1.1 Thanks, Dave Dave, T.38 in some fashion. But you don't really explain your call flow or what you are trying to do. You say you have PSTN and then talk about SIP. Are you just trying to pass the calls to physical FAX machines, or a server to handle faxing? Elaborate a bit and I am sure someone can offer some advice. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanted: UK-specific hardware recommendations (FXOand FXS)
Roger Burton West wrote: I want to hook one of them to the PSTN. Given that I am in the UK, what is a reasonably easily-available device to provide an FXO interface from a Linux box, with a minimum of faffing around with drivers? Just one line is needed, though in theory two might eventually be useful. My usual white-box hardware suppliers don't seem to play in this field. I've had good experiences with an OpenVox A400P, once you've done the Dahdi dance, it settles down to be very reliable. Reasonable price, too. I bought mine from Voipon, although I'm sure a bit of shopping around will find other vendors. Cheers, Ade. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the PSTN. The carrier sending the calls wants me to be able to pass faxes to physical fax machines on the PSTN. So far they are failing. We just want ot be able to pass faxes using g711u or t.38 pass through. Thanks, Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Friday, September 03, 2010 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Faxes On Fri, Sep 3, 2010 at 10:49 AM, dave george dgeo...@teletoneinc.com wrote: We have asterisk connected to the PSTN VIA TE410P (ANSI SS7 to T1 PRI) cards. I am having trouble completing faxes. Carrier send calls to me using SIP. Any recommendation to have some success with Fax. We trying using T.38 pass through and using G711U codec. Asterisk Version 1.6.1.1 Thanks, Dave Dave, T.38 in some fashion. But you don't really explain your call flow or what you are trying to do. You say you have PSTN and then talk about SIP. Are you just trying to pass the calls to physical FAX machines, or a server to handle faxing? Elaborate a bit and I am sure someone can offer some advice. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] openvz
Outlook? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, September 03, 2010 3:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] openvz From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias Subject: [asterisk-users] openvz Can i run asterisk on a openvz vps or do i need a kernel? I dont use dadi Blind Answer - you should be able to; Asterisk doesn't rebuild the kernel. You might have to get some kernel source using ZYPPER (in caps so Outlook express doesn't change it to zipper). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya - What are their current cost?
You can come and take my 3 Ayaya switches and all the associated cost$! It'a all * for me. and to answer your ?---they cost too much$ =) On Thu, Sep 2, 2010 at 12:32 PM, bruce bruce bruceb...@gmail.com wrote: I am not interested in open source solutions. I want to know how much the propriety systems cost in terms of licensing. Specially Avaya now a days per extension. Exclusive or Inclusive of the hardware for 10 agents as noted. Thanks On Thu, Sep 2, 2010 at 8:18 AM, Muhammad Shomail Haider msh0...@gmail.com wrote: Hi Bruce, It all depends what exactly you are in need of. A basic call center solution will only cost $500 exclusive of hardware, depending on your need you will have to decide what type of servers you need or weather you would have handsets or softphones, type of headgears you want, kind of workstation you will need. I work for a company that provides open source call center solution. You can visit the website www.crystalconsulting.pk or if you want more detail you can email the detail of the requirements on sa...@crystalconsulting.pk or you can email me and I can revert back to you with detail. Regards, Shomail On Fri, Aug 27, 2010 at 2:03 PM, justmun...@gmail.com wrote: Hi Everyone, Just a quick estimate of what Call Center Software/Hardware providers charge now a days for a 10 seat and 20 seat with upfront costs and monthly licensing cost? Thanks, -- Muhammad Shomail Haider www.shomail.blogspot.com www.facebook.com/shomail www.twitter.com/shomail www.linkedin.com/in/shomail -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] openvz
1- I am interested in this as well. Looking into Proxmox as it provides a nice interface (do you guys know of any other good one?) 2- Would the conference calls be fine as well? I understanding Asterisk 1.6.x uses a kernel timing source now a days so that ztdummy is not needed anymore? 3- Would installing from yum repository be just fine? Thanks On Fri, Sep 3, 2010 at 10:31 AM, mattias m...@mjw.se wrote: Outlook? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, September 03, 2010 3:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] openvz From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias Subject: [asterisk-users] openvz Can i run asterisk on a openvz vps or do i need a kernel? I dont use dadi Blind Answer - you should be able to; Asterisk doesn't rebuild the kernel. You might have to get some kernel source using ZYPPER (in caps so Outlook express doesn't change it to zipper). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] openvz
Blind Answer - you should be able to; Asterisk doesn't rebuild the kernel. You might have to get some kernel source using ZYPPER (in caps so Outlook express doesn't change it to zipper). El 03/09/10 09:31, mattias escribió: Outlook? Outlook Express is a total PITA. Should I recommend you to use Mozilla Thunderbird... Sorry for the offtopic. And I agree, you should have no problems with asterisk using it inside an openVZ VPS. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to create a coredump for Asterisk
On Thu, Sep 2, 2010 at 2:26 PM, Thorolf Godawa nos...@godawa.de wrote: Any idea what is going wrong here? Read doc/backtrace.txt If you cannot get Asterisk to coredump, try running it under gdb to see what is happening. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
On Fri, Sep 3, 2010 at 11:50 AM, dave george dgeo...@teletoneinc.com wrote: The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the PSTN. You don't say the percentage that are failing. However, people who have worked with SIP on asterisk have been known to do: exten = s,1,Playback(silence/1) exten = s,n,Whatever(is_next) And I don't know why, but this seems to make things better. If you're doing an Answer and then a receive_Fax, try putting a playback silence in between and see if that helps anything. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
All my attempts are failing. Thanks Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Friday, September 03, 2010 1:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Faxes On Fri, Sep 3, 2010 at 11:50 AM, dave george dgeo...@teletoneinc.com wrote: The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the PSTN. You don't say the percentage that are failing. However, people who have worked with SIP on asterisk have been known to do: exten = s,1,Playback(silence/1) exten = s,n,Whatever(is_next) And I don't know why, but this seems to make things better. If you're doing an Answer and then a receive_Fax, try putting a playback silence in between and see if that helps anything. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
Can you post the dialplan snippet you are using? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
On 09/03/2010 10:50 AM, dave george wrote: The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the PSTN. The carrier sending the calls wants me to be able to pass faxes to physical fax machines on the PSTN. So far they are failing. We just want ot be able to pass faxes using g711u or t.38 pass through. As I told you on the asterisk-ss7 list, you can't 'pass through' T.38, because the PSTN does not speak T.38. If one side of the call is SIP, and the other side is TDM, then you have only two choices: pass the call through in audio mode (FAX over VoIP), or act as a T.38 gateway (FAX over T.38). At this time, the only option without patching Asterisk is to pass the call through in audio mode, but there are many, many problems with doing FAX over VoIP (Steve Underwood's page on the soft-switch.org site explains them very well). There are patches in the issue tracker at issues.asterisk.org to add T.38 gateway functionality to various releases of Asterisk, and they work well for quite a few people. If you added that, you'd be able to act as a T.38 gateway, which would dramatically increase your chances of success. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
Thanks Kevin, I tried passing it over VOIP using g711U codecs with no success. I will try using the patches that you mentioned and post the results. Thanks, Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, September 03, 2010 2:17 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Faxes On 09/03/2010 10:50 AM, dave george wrote: The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the PSTN. The carrier sending the calls wants me to be able to pass faxes to physical fax machines on the PSTN. So far they are failing. We just want ot be able to pass faxes using g711u or t.38 pass through. As I told you on the asterisk-ss7 list, you can't 'pass through' T.38, because the PSTN does not speak T.38. If one side of the call is SIP, and the other side is TDM, then you have only two choices: pass the call through in audio mode (FAX over VoIP), or act as a T.38 gateway (FAX over T.38). At this time, the only option without patching Asterisk is to pass the call through in audio mode, but there are many, many problems with doing FAX over VoIP (Steve Underwood's page on the soft-switch.org site explains them very well). There are patches in the issue tracker at issues.asterisk.org to add T.38 gateway functionality to various releases of Asterisk, and they work well for quite a few people. If you added that, you'd be able to act as a T.38 gateway, which would dramatically increase your chances of success. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to tell if there is a transfer from CDR?
Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
g711 across a network without perfect jitter/delay characteristics will not work. You cannot do g711 faxing across the internet - at all. It's not a perfect solution even in an office on a dedicated LAN environment (you'll still get failed faxes). On Fri, Sep 3, 2010 at 12:32 PM, dave george dgeo...@teletoneinc.comwrote: Thanks Kevin, I tried passing it over VOIP using g711U codecs with no success. I will try using the patches that you mentioned and post the results. Thanks, Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, September 03, 2010 2:17 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Faxes On 09/03/2010 10:50 AM, dave george wrote: The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the PSTN. The carrier sending the calls wants me to be able to pass faxes to physical fax machines on the PSTN. So far they are failing. We just want ot be able to pass faxes using g711u or t.38 pass through. As I told you on the asterisk-ss7 list, you can't 'pass through' T.38, because the PSTN does not speak T.38. If one side of the call is SIP, and the other side is TDM, then you have only two choices: pass the call through in audio mode (FAX over VoIP), or act as a T.38 gateway (FAX over T.38). At this time, the only option without patching Asterisk is to pass the call through in audio mode, but there are many, many problems with doing FAX over VoIP (Steve Underwood's page on the soft-switch.org site explains them very well). There are patches in the issue tracker at issues.asterisk.org to add T.38 gateway functionality to various releases of Asterisk, and they work well for quite a few people. If you added that, you'd be able to act as a T.38 gateway, which would dramatically increase your chances of success. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] openvz
Some days ago in my lab I setup Proxmox, installed a CentOS 5.2 appliance on OpenVZ, installed all asterisk related stuff (except dahdi), including php, mysql, munin, other tools, set it up with a dialplan and it worked just fine. Then manually made multiple copies of the folder where all this installation was stored. This gave me multiple instances of CentOS/asterisk, which I configured with unique IP addresses. I have been using this setup for a few days now and all seems good. I plan to test conferencing next week. So far seems like a good and stable setup. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-03 1:22 PM, Miguel Molina mmol...@millenium.com.co wrote: Blind Answer - you should be able to; Asterisk doesn't rebuild the kernel. You might have to ge... El 03/09/10 09:31, mattias escribió: Outlook? Outlook Express is a total PITA. Should I recommend you to use Mozilla Thunderbird... Sorry for the offtopic. And I agree, you should have no problems with asterisk using it inside an openVZ VPS. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocat... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
Try open souce solution ICTFAX for T.38 faxing developed by us available at http://www.sourceforge.net/projects/ictfax Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ On Sat, Sep 4, 2010 at 3:03 AM, Joel Maslak jmas...@antelope.net wrote: g711 across a network without perfect jitter/delay characteristics will not work. You cannot do g711 faxing across the internet - at all. It's not a perfect solution even in an office on a dedicated LAN environment (you'll still get failed faxes). On Fri, Sep 3, 2010 at 12:32 PM, dave george dgeo...@teletoneinc.comwrote: Thanks Kevin, I tried passing it over VOIP using g711U codecs with no success. I will try using the patches that you mentioned and post the results. Thanks, Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, September 03, 2010 2:17 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Faxes On 09/03/2010 10:50 AM, dave george wrote: The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the PSTN. The carrier sending the calls wants me to be able to pass faxes to physical fax machines on the PSTN. So far they are failing. We just want ot be able to pass faxes using g711u or t.38 pass through. As I told you on the asterisk-ss7 list, you can't 'pass through' T.38, because the PSTN does not speak T.38. If one side of the call is SIP, and the other side is TDM, then you have only two choices: pass the call through in audio mode (FAX over VoIP), or act as a T.38 gateway (FAX over T.38). At this time, the only option without patching Asterisk is to pass the call through in audio mode, but there are many, many problems with doing FAX over VoIP (Steve Underwood's page on the soft-switch.org site explains them very well). There are patches in the issue tracker at issues.asterisk.org to add T.38 gateway functionality to various releases of Asterisk, and they work well for quite a few people. If you added that, you'd be able to act as a T.38 gateway, which would dramatically increase your chances of success. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
