RE: [Asterisk-Users] Conference solution for 100+ users
We have found a new thing called 'the pub' It even provides beverages. Trust me, you can't find a program that can do that! PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergio Veltri Sent: Wednesday, 20 April 2005 3:37 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Conference solution for 100+ users Hi List, I am looking for some advice. I need to come up with a conference solution that will allow users to join mainly to listen to a guy talk about a product for an hour. My main concern is the client side. I need people from within firewalls to be able to join the conference with speakers built-in their laptops or computers. All I know is that Skype works in most of the customers this guy will be addressing. I am considering the following options: 1-Skype-like softphone for *. is there any? 2-Just do audio streaming and have the customers use windows media player. (I dont know how to do this) 3-Use some kind of Softphone with VPN... 4- Do Softphone---Port 80--- SER---Asterisk w/meetme. Whatever solution I come up with MUST allow anybody to listen in assuming nobody can change firewalls. Any one has already done this? Any feedback will be much appreciated. Thanks, Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RealTime ignoring switch = Realtime/context@realtime_ext
OK, been messing with RealTime like a week off and on, I can safely say it's killing me! I have dug and dug and dug to find what I am missing, no dice. I am running the latest version of * from CVS as of about a week ago. Call comes in from a PRI into the todd_test_1 extension, if I uncomment the lines for the _888 number directly in the extensions.conf file the call is answered without a problem. If I comment the lines and just leave the switch in place it's suppose to lookup the extensions from the mysql table from what I understand. All I get when calling in from the PRI is this: -- Extension '8885551212' in context 'todd_test_1' from '2145551212' does not exist. Rejecting call on channel 0/1, span 1 It appears that the switch command is totally being ignored. I also checked the MySQL logs to see if Asterisk/RealTime was even hitting it but I see nothing in the MySQL logs at all that would indicate Asterisk is talking to it. My phone numbers/passwords etc. have been changed but most everything else in my configs are as is. Any help would be appreciated, I am sure I am just missing something really simple and I am gonna smack myself in the head when it's brought to my attention. ### extensions.conf# [todd_test_1] switch = Realtime/[EMAIL PROTECTED] ;## New stuff for new system ## ;exten = _888NXX,1,Answer ;exten = _888NXX,2,Wait(1) ;exten = _888NXX,3,Playback(cannot-complete-as-dialed) ;exten = _888NXX,4,Playback(check-number-dial-again) ;exten = _888NXX,5,Hangup # --- ## extconfig.conf# realtime_ext = mysql,mydbname,extensions_table ## ## res_mysql.conf # [general] dbhost = my.dbserver.com dbname = mydbname dbuser = mydbusername dbpass = mydbpass dbport = mydbport dbsock = /tmp/mysql.sock ## - # DB Schema # FieldTypeNullDefault id int(11) No context varchar(20) No exten varchar(20) No priority tinyint(4) No 0 app varchar(20) No appdata varchar(128) No 1;todd_test_2;_888NXX;1;Wait;2 2;todd_test_2;_888NXX;2;SayNumber;102 3;todd_test_2;_888NXX;1;Playback;pbx-invalid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NAT and only been able to have 1 SIP phone behind
Guys. Ive read on the wiki that a common problem with nat is that you can only have 1 sip phone behind, how do you get around this issue? Having a sip enabled router behind the nat like the GS 488 489 or 486? Or how have you done it without having any kind of linux box (SER or *) behind the nat. The idea is to have 2 or more sip clients behind some NAT and been able to connect to a remote asterisk box. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Text Messages
Is there any way of sending text messages to SIP ATA's or phones? Like SMS but for SIP IAX2 ATAs or phones. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Using voicemail independently from Asterisk PBX
How have you done it for */* combinations? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Martes, 19 de Abril de 2005 07:35 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Using voicemail independently from Asterisk PBX Sure. You need to decide how you will interconnect to * vm from CM. H323? SIP? MGCP? Then, you'll set your dialplan so that when calls come into *, instead of going to a station first, it goes immediately to the Voicemail app. MWI is probably the biggest unknown. I'm not sure if anyone has figured out how to get MWI to work between CM and *. I know several folks have figured it out on the list for SER/* and */* combinations. On 4/19/05, John Riek [EMAIL PROTECTED] wrote: I would like to use Asterisk as a standalone voicemail server and integrate it with a Cisco Call Manager PBX. I need to know how to run the voicemail system independently. Does anybody know how to do this? __ Do you Yahoo!? Plan great trips with Yahoo! Travel: Now over 17,000 guides! http://travel.yahoo.com/p-travelguide ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX realtime HELP
I have been looking and have tried many many things but not have been able to get it working I am running Connected to Asterisk CVS-HEAD-04/14/05-11:56:07 currently running on localhost (pid = 1927) Regards Paul Dracevich Wireless Technology Consultant Wayby Group Mobile +64 29 638 9675 Phone +64 9 623 2143 Fax +64 9 623 1380 email [EMAIL PROTECTED] website www.vnet.cc the freedom to communicate is the right of every individual in the 21st century Intellectual Property protection is the key to the Knowledge Economy This email was sent to you via YOUtopia ... it's all about YOU. The information contained in this email and any attachments is confidential and may be legally privileged. If you are not the intended recipient then you must not use, disseminate, distribute or copy any information contained in this email or any attachments. If you have received this email in error, please contact us immediately and delete this email. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IPv6 possible?
On Tue, Apr 19, 2005 at 16:44:58 +0100, Chris Hills wrote: Ronald Wiplinger wrote: I have IPv6 (via tunnel) available. Is there a solution for IPv6 available? Hi Ronald This is something I would like as well. Unfortunately there is no support for IPv6 at present. Perhaps you could put in a bounty for it? This was announced on the -dev list recently. I haven't looked at it. - Date: Tue, 5 Apr 2005 01:01:06 +0200 From: Mikael Magnusson [EMAIL PROTECTED] To: asterisk-dev@lists.digium.com Subject: Re: [Asterisk-Dev] IPv6 On Mon, Apr 04, 2005 at 09:37:40PM +0100, David Woodhouse wrote: On Sun, 2005-04-03 at 20:30 +0200, Mikael Magnusson wrote: The patch will need a lot of testing, since it modifies (almost) all networking code in Asterisk. As far as I can tell that needs doing anyway, because at the moment we don't even handle falling back to the second and subsequent A record, let alone falling back from to A records. How close are you to having something to share? Are you doing IAX2 while you're at it? The experimental IPv6 patch can be downloaded from [1]. It adds IPv6 support to the manager interface and the SIP and IAX2 channels. It may break the IPv4 support, since It will never use IPv4 when communicating with servers that have records. diffstat: Makefile |3 acl.c | 245 +++ channels/Makefile | 16 channels/chan_iax2.c | 583 ++- channels/chan_sip.c| 953 ++-- channels/iax2-parser.c | 36 + channels/iax2-parser.h |8 include/asterisk/acl.h | 23 - include/asterisk/manager.h |3 include/asterisk/net.h | 90 include/asterisk/rtp.h |9 manager.c | 75 +-- net.c | 960 + pbx/Makefile |4 rtp.c | 376 + 15 files changed, 2391 insertions(+), 993 deletions(-) Use bindaddr=[::0] to enable both IPv4 and IPv6 support. And you may not disable rtp checksums since UDP checksums are required for IPv6. /Mikael [1] http://www.hem.za.org/asterisk-ipv6_20050404-2.patch.gz - rvdp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to post my impovements to ASTCC?
On 4/3/05, Ronald Wiplinger [EMAIL PROTECTED] wrote: You can't see the sweat, but ... I would like tp post my improvements to ASTCC somewhere, ... but where??? Post them as patches to bugs.digium.com and then they can be incorperated into the main code. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax and spandsp
Julian J. M. wrote: I want asterisk to receive incoming faxes (via rxfax application) and send them by mail. The problem is that, although the fax machine and the asterisk log report a succesful transfer, the tiff file is just I have not experienced this before, but I am using spandsp-0.0.2pre10, perhaps you could try this, to see if this matters? I will surely (in the near future) rebuild a box and try out the new spandsp (pre15) but maybe you can try downgrading as a test. Cheers Kristof ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OH323 incoming audio stutter
I'm using asterisk-oh323-0.6.5 with the Janus patch 4 versions of pwlib (v1.6.6.3) and openh323 (v1.13.5.3), and using it to connect to my provider's switch. The effect that I am seeing is that a call starts off fine, but suddenly after a few minutes the audio coming into Asterisk via OH323 gets very broken up to the point of being about 90% silence with occasional brief snippets of audio getting through. When this happens, the audio going out from Asterisk to the other end is still fine, with no disturbances. I have observed this both when using SIP for the local leg of the call and when using IAX. I'm not really sure where to look to diagnose this, not whether it is likely to be an Asterisk problem or something in the switch. Any advice would be appreciated! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting SIP username for CallerID
Hi :) When I send an incoming call to a queue, I'm doing this: exten = 6608140,1,SetCallerID(CCUK) exten = 6608140,2,SetCIDName(CCUK) exten = 6608140,3,Queue(ccuk,r) I want the phone to say 'CCUK' - the queue name is more important to know than the incoming Caller ID :) Unfortunately the SIP phone (a cheapy using the PA168S chip and 1.42 firmware) displays the caller ID of asterisk when I do this, and it's clear why: --- -- outgoing agentcall, to agent '1601', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1601 -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/1301|20|t) in new stack We're at 10.0.0.242 port 15334 12 headers, 12 lines Reliably Transmitting (NAT) to 10.0.0.82:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.242:5060;branch=z9hG4bK70ccd454;rport From: CCUK sip:[EMAIL PROTECTED];tag=as13d91518 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 20 Apr 2005 08:34:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 260 If I SetCallerID(12345678) then it is changed to sip:[EMAIL PROTECTED] as I'd expect, but if I use a string value, it stays at 'sip:[EMAIL PROTECTED]' So my question is, how can I change the sip username from sip:[EMAIL PROTECTED] to sip:[EMAIL PROTECTED] ? Am I doing something mind-bogglingly stupid? Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] friendly networks via **
I like the way FWD does to connect to other friendly networks via **, however, I am not sure how is the best way. Can I just use exten = **393.,1 exten = **394.,n ... ??? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax detected, but no fax extension
-- Starting simple switch on 'Zap/3-1'
-- Executing NoOp(Zap/3-1, 9229443944-) in new stack
-- Executing Answer(Zap/3-1, ) in new stack
-- Executing Zapateller(Zap/3-1, ) in new stack
-- Executing BackGround(Zap/3-1, custom/Welcome) in new stack
-- Playing 'custom/Welcome' (language 'default')
Apr 20 16:49:29 NOTICE[29109]: chan_zap.c:4300 zt_read: Fax detected,
but no fax extension
-- Executing BackGround(Zap/3-1, if-u-know-ext-dial) in new stack
-- Playing 'if-u-know-ext-dial' (language 'default')
-- Executing Dial(Zap/3-1,
SIP/601SIP/621ZAP/1r1SIP/610|30|tr) in new stack
-- Called 601
Apr 20 16:49:43 NOTICE[29109]: app_dial.c:973 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3)
-- Called 1r1
Apr 20 16:49:43 NOTICE[29109]: app_dial.c:973 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3)
-- Zap/1-1 is ringing
-- SIP/601-49e2 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- SIP/601-49e2 answered Zap/3-1
-- Hungup 'Zap/1-1'
I tried to use the exension 2201, but it did not work, so I have changed
it to s, but it does not work either.
[fax]
exten = s,1,Macro(faxreceive)
;exten = 2202,1,Macro(faxreceive)
;exten = 2203,1,Macro(faxreceive)
exten = h,1,system(/usr/local/sbin/mailfax ${FAXFILE} \
${EMAILADDR} ${CALLERIDNUM} ${CALLERIDNAME})
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten = s,3,rxfax(${FAXFILE})
exten = s,103,SetVar([EMAIL PROTECTED])
exten = s,104,Goto(3)
How can I solve it?
bye
Ronald
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Re: [Asterisk-Users] Setting SIP username for CallerID
Gavin Hamill wrote: Hi :) When I send an incoming call to a queue, I'm doing this: exten = 6608140,1,SetCallerID(CCUK) exten = 6608140,2,SetCIDName(CCUK) exten = 6608140,3,Queue(ccuk,r) I want the phone to say 'CCUK' - the queue name is more important to know than the incoming Caller ID :) Unfortunately the SIP phone (a cheapy using the PA168S chip and 1.42 firmware) displays the caller ID of asterisk when I do this, and it's clear why: --- -- outgoing agentcall, to agent '1601', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1601 -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/1301|20|t) in new stack We're at 10.0.0.242 port 15334 12 headers, 12 lines Reliably Transmitting (NAT) to 10.0.0.82:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.242:5060;branch=z9hG4bK70ccd454;rport From: CCUK sip:[EMAIL PROTECTED];tag=as13d91518 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 20 Apr 2005 08:34:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 260 If I SetCallerID(12345678) then it is changed to sip:[EMAIL PROTECTED] as I'd expect, but if I use a string value, it stays at 'sip:[EMAIL PROTECTED]' So my question is, how can I change the sip username from sip:[EMAIL PROTECTED] to sip:[EMAIL PROTECTED] ? Am I doing something mind-bogglingly stupid? Shouldn't be there a quote mark and two values, like: SetCallerID(Ronald 123456789) bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA Help
Hi, I have a Cisco ATA 186 that I bought on my recent overseas trip and its the I2 series which has higher impedance than the New Zealand standard 600ohm. Is there something I can do to make it listen to my DTMF tones? Regards, Sahil Gupta VoiceValley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting SIP username for CallerID
On Wednesday 20 April 2005 10:32, Ronald Wiplinger wrote: So my question is, how can I change the sip username from sip:[EMAIL PROTECTED] to sip:[EMAIL PROTECTED] ? Shouldn't be there a quote mark and two values, like: SetCallerID(Ronald 123456789) Just tried a few combinations of that, and using the precise command above, the phone shows only the number. If I put a string inside the , * will still generate sip:[EMAIL PROTECTED]'... if I just put SetCallerID(CCUK) alone, I still get sip:[EMAIL PROTECTED] I am using CVS HEAD as of yesterday :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RealTime ignoring switch = Realtime/context@realtime_ext
Me wrote: OK, been messing with RealTime like a week off and on, I can safely say it's killing me! I have dug and dug and dug to find what I am missing, no dice. I am running the latest version of * from CVS as of about a week ago. Call comes in from a PRI into the todd_test_1 extension, if I uncomment the lines for the _888 number directly in the extensions.conf file the call is answered without a problem. If I comment the lines and just leave the switch in place it's suppose to lookup the extensions from the mysql table from what I understand. All I get when calling in from the PRI is this: -- Extension '8885551212' in context 'todd_test_1' from '2145551212' does not exist. Rejecting call on channel 0/1, span 1 It appears that the switch command is totally being ignored. I also checked the MySQL logs to see if Asterisk/RealTime was even hitting it but I see nothing in the MySQL logs at all that would indicate Asterisk is talking to it. My phone numbers/passwords etc. have been changed but most everything else in my configs are as is. Any help would be appreciated, I am sure I am just missing something really simple and I am gonna smack myself in the head when it's brought to my attention. ### extensions.conf# [todd_test_1] switch = Realtime/[EMAIL PROTECTED] shouldn't it be Realtime/[EMAIL PROTECTED] or [todd_test_1] include = todd_test_2 [todd_test_2] switch = Realtime/[EMAIL PROTECTED] ??? BTW, the numbering of the priorities should increase: 1;todd_test_2;_888NXX;1;Wait;2 2;todd_test_2;_888NXX;2;SayNumber;102 3;todd_test_2;_888NXX;1;Playback;pbx-invalid bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax detected, but no fax extension
In your incoming context add
exten = fax,1,Goto(fax,2202,1)
On Wed, 2005-04-20 at 12:26, Ronald Wiplinger wrote:
-- Starting simple switch on 'Zap/3-1'
-- Executing NoOp(Zap/3-1, 9229443944-) in new stack
-- Executing Answer(Zap/3-1, ) in new stack
-- Executing Zapateller(Zap/3-1, ) in new stack
-- Executing BackGround(Zap/3-1, custom/Welcome) in new stack
-- Playing 'custom/Welcome' (language 'default')
Apr 20 16:49:29 NOTICE[29109]: chan_zap.c:4300 zt_read: Fax detected,
but no fax extension
-- Executing BackGround(Zap/3-1, if-u-know-ext-dial) in new stack
-- Playing 'if-u-know-ext-dial' (language 'default')
-- Executing Dial(Zap/3-1,
SIP/601SIP/621ZAP/1r1SIP/610|30|tr) in new stack
-- Called 601
Apr 20 16:49:43 NOTICE[29109]: app_dial.c:973 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3)
-- Called 1r1
Apr 20 16:49:43 NOTICE[29109]: app_dial.c:973 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3)
-- Zap/1-1 is ringing
-- SIP/601-49e2 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- SIP/601-49e2 answered Zap/3-1
-- Hungup 'Zap/1-1'
I tried to use the exension 2201, but it did not work, so I have changed
it to s, but it does not work either.
[fax]
exten = s,1,Macro(faxreceive)
;exten = 2202,1,Macro(faxreceive)
;exten = 2203,1,Macro(faxreceive)
exten = h,1,system(/usr/local/sbin/mailfax ${FAXFILE} \
${EMAILADDR} ${CALLERIDNUM} ${CALLERIDNAME})
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten = s,3,rxfax(${FAXFILE})
exten = s,103,SetVar([EMAIL PROTECTED])
exten = s,104,Goto(3)
How can I solve it?
bye
Ronald
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Re: [Asterisk-Users] help needed for sound device setup
U don't need to have sound device for * sound service running just make sure that you have in modules.conf noload = chan_alsa.so noload = chan_oss.so On Wed, 2005-04-20 at 08:44, [EMAIL PROTECTED] wrote: Hi, I installed asterisk-1-0-7 and running it succesfully. But iam unable to use the sound services. I have the following warning messages when i launch asterisk Apr 19 21:15:40 WARNING[10918]: chan_oss.c:992 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found Apr 19 21:15:40 WARNING[10918]: chan_oss.c:239 sound_thread: Read error on sound device: Resource temporarily unavailable what i need to do, to install the sound device properly. Does it require any hardware support. i have the following kernel modules for audio support [EMAIL PROTECTED] asterisk-1.0.7]# lsmod | grep audio i810_audio 27720 1 (autoclean) ac97_codec 13640 0 (autoclean) [i810_audio] soundcore 6404 2 (autoclean) [i810_audio] thanks, Somesh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using voicemail independently from Asterisk PBX
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 John Riek wrote: | I would like to use Asterisk as a standalone voicemail server and | integrate it with a Cisco Call Manager PBX. I need to know how to | run the voicemail system independently. Does anybody know how to | do this? | What Version of CCM do you have ? Is it CCM ou CCM Express ? If it is CCM 4.0 then i advise you to use a SIP trunk between CCM and *. Create your mailbox(s) in voicemail.conf, make your dialplan and route the calls to * and it's done ;) Rgds João Amaro | __ Do you Yahoo!? Plan great trips | with Yahoo! Travel: Now over 17,000 guides! | http://travel.yahoo.com/p-travelguide | ___ Asterisk-Users | mailing list Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users To | UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFCZipLJUm/Bor63CERAsHJAJ94yoa53M1FjpVLX556LOQ0te+RZACePKNk g7+vypFXwz6p2YJsdop/qCI= =IGCi -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NAT issues
Hi there I have got a really strange issue and my problem is not that it is not working, but why it is working. I have Asterisk set up on a public IP, but the clients are behind a Port Restricted NAT with no support for UPnP. My clients dial into a meetme conference. When I don't specify nat=yes in the sip.conf file, then it works?? But not sure why it works because I cannot find any reference to the IP of the NAT in the SIP messages. I have not put in any nat support in my custom built client either. The reason that this is a problem is I a not sure if it will work on other LANs, and also find it hard to debug if it is working when my research tells me that it should not be working. I tried putting in nat=yes in the sip.conf file, and asterisk then rewrites the sip message with the IP of the Nat and the external port. It still works, but only if there is a constant flow of rtp traffic. If there is a break in the traffic, then the connection is lost. However, this problem may be to do with the fact that pinging is disabled on our network, but not sure. I am really stuck here. I have read that dealing with NATs can be a big problem, but it seems to work better when I dont put in any NAT support. Am I missing something here? Does anyone have any ideas or advice? Many thanks Steven ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting SIP username for CallerID
Try using SetCIDNum(CCUK) -Arun On 4/20/05, Gavin Hamill [EMAIL PROTECTED] wrote: On Wednesday 20 April 2005 10:32, Ronald Wiplinger wrote: So my question is, how can I change the sip username from sip:[EMAIL PROTECTED] to sip:[EMAIL PROTECTED] ? Shouldn't be there a quote mark and two values, like: SetCallerID(Ronald 123456789) Just tried a few combinations of that, and using the precise command above, the phone shows only the number. If I put a string inside the , * will still generate sip:[EMAIL PROTECTED]'... if I just put SetCallerID(CCUK) alone, I still get sip:[EMAIL PROTECTED] I am using CVS HEAD as of yesterday :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax detected, but no fax extension
Vladyslav wrote:
In your incoming context add
exten = fax,1,Goto(fax,2202,1)
It did not work ;-(
[incoming_88097680]
exten = s,1,NoOp(${CALLERIDNUM})
exten = s,2,Wait(1)
exten = s,3,SetCallerId(9${CALLERIDNUM})
exten = s,4,GotoIfTime(08:00-21:20|sun-sat|*|*?house-day,s,1)
exten = s,5,Goto(house-night,s,1)
exten = fax,1,Goto(fax,2201,1)
[fax]
exten = 2201,1,Macro(faxreceive)
;exten = 2202,1,Macro(faxreceive)
;exten = 2203,1,Macro(faxreceive)
exten = h,1,system(/usr/local/sbin/mailfax ${FAXFILE} \
${EMAILADDR} ${CALLERIDNUM} ${CALLERIDNAME})
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten = s,3,rxfax(${FAXFILE})
exten = s,103,SetVar([EMAIL PROTECTED])
exten = s,104,Goto(3)
On Wed, 2005-04-20 at 12:26, Ronald Wiplinger wrote:
-- Starting simple switch on 'Zap/3-1'
-- Executing NoOp(Zap/3-1, 9229443944-) in new stack
-- Executing Answer(Zap/3-1, ) in new stack
-- Executing Zapateller(Zap/3-1, ) in new stack
-- Executing BackGround(Zap/3-1, custom/Welcome) in new stack
-- Playing 'custom/Welcome' (language 'default')
Apr 20 16:49:29 NOTICE[29109]: chan_zap.c:4300 zt_read: Fax detected,
but no fax extension
-- Executing BackGround(Zap/3-1, if-u-know-ext-dial) in new stack
-- Playing 'if-u-know-ext-dial' (language 'default')
-- Executing Dial(Zap/3-1,
SIP/601SIP/621ZAP/1r1SIP/610|30|tr) in new stack
-- Called 601
Apr 20 16:49:43 NOTICE[29109]: app_dial.c:973 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3)
-- Called 1r1
Apr 20 16:49:43 NOTICE[29109]: app_dial.c:973 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3)
-- Zap/1-1 is ringing
-- SIP/601-49e2 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- SIP/601-49e2 answered Zap/3-1
-- Hungup 'Zap/1-1'
I tried to use the exension 2201, but it did not work, so I have changed
it to s, but it does not work either.
[fax]
exten = s,1,Macro(faxreceive)
;exten = 2202,1,Macro(faxreceive)
;exten = 2203,1,Macro(faxreceive)
exten = h,1,system(/usr/local/sbin/mailfax ${FAXFILE} \
${EMAILADDR} ${CALLERIDNUM} ${CALLERIDNAME})
bye
Ronald
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[Asterisk-Users] TE410P PCI-slot
Hi, I was just wondering about a comment I found in the voip-info.org wiki: The DIGIUM TE410 PRI card, requires a motherboard with a 64bit 3.3v PCI slot. Given the bandwidth requirements, it would be better to have a 133Mhz slot if available. Since the card seems to always clock at 33MHz. I can't really see how a 133MHz slot would make any difference? -- Regards, Tais M. Hansen ComX Networks A/S Tel: +45-70257474 Fax: +45-70257374 pgpyKgaP0qbBx.pgp Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and VAD
Hi guys, Is it possible turn on/off VAD (silence suspression) w/ Asterisk? Thanks in advance :), Pavel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 360s and Asterisk
Hi all, has anyone else had any experience with these new Snom devices. I am having trouble with placing calls on hold. The hold works fine, the music on hold kicks in, but when you take it off of hold the voice goes really choppy. I can't tell if it is a server side issue or not but I am running Asterisk in verbose mode (like a 50 setting) and don't get any feedback. This is from a PSTN connected user inbound to internal Snom 360 extension. Any other relevant experiences with these phones would be appreciated. Thanks! CM ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT issues
Steven Langley wrote: I tried putting in nat=yes in the sip.conf file, and asterisk then rewrites the sip message with the IP of the Nat and the external port. It still works, but only if there is a constant flow of rtp traffic. If there is a break in the traffic, then the connection is lost. However, this problem may be to do with the fact that pinging is disabled on our network, but not sure. If you are the same person I spoke with on IRC then you forgot to mention that the SIP clients use VAD and that it cannot be disabled. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Phone Compatability
The Sipura SPA-841 has everything except memory buttons but has a directory and speeddials so I don't think that's so important. Cheap and well made, although if the speaker phone is very important, get Polycoms, it's the business they are best in. Chris Mason www.anguillaguide.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Wednesday, April 20, 2005 12:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP Phone Compatability Every once in a while I read messages about people having problems with certain models of SIP phones, some of them being well known models. I'm interested in purchasing new SIP phones for my office and wanted to know which brand/model is most stable with Asterisk, which allows most office features. These features should include: multiple-line appearances (at least 3), call conference, blind and non-blind transfer, memory buttons or speed dials, voice message light indicator, speaker phone, mute, redial, caller-id display. Anything on top of these features is a plus but not really a requirement. Any suggestions? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and VAD
Pavel Siderov wrote: Is it possible turn on/off VAD (silence suspression) w/ Asterisk? Asterisk does not support VAD so it doesn't make sense to be able to disable it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting SIP username for CallerID
On Wednesday 20 April 2005 11:15, Arunachala wrote: Try using SetCIDNum(CCUK) Nope, the most I can ever extract from any combination of the three 'CID' commands is this in the SIP messages :( From: CCUK sip:[EMAIL PROTECTED] Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPAM SPAM SPAM SAM SPAM SPAM SPAM
Yeah... unbelieveable but true: spam, defined often us undesired bulk mail comes in many forms, including messages from this list. I tried several times - obviously unsuccessfully, as you can see - to unsubscribe from this list, and sice that did not work, i set my mail server to BOUNCE list messages. But nope, nobody gets it - the mail server is deaf and dumb and keeps sending me dozens and dozens of messages. If there is anybody reading this who can reach the person(s) in charge of the mailing list server, please tell them that they are having (and causing) a problem. ;-) Al -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 2800 with Asterisk
Hi, Has anyone used Cisco 2800 Integrated services router to intiate SIP call to Asterisk. I would like to use it as gateway on to which T1 terminates and make Asterisk as my session target for few lines. Please let me know if there are any issues. Thanks, Sharath ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX realtime HELP
I have been looking and have tried many many things but not have been able to get it working I am running Connected to Asterisk CVS-HEAD-04/14/05-11:56:07 currently running on localhost (pid = 1927) Regards Paul Dracevich Wireless Technology Consultant Wayby Group Mobile +64 29 638 9675 Phone +64 9 623 2143 Fax +64 9 623 1380 email [EMAIL PROTECTED] website www.vnet.cc the freedom to communicate is the right of every individual in the 21st century Intellectual Property protection is the key to the Knowledge Economy This email was sent to you via YOUtopia ... it's all about YOU. The information contained in this email and any attachments is confidential and may be legally privileged. If you are not the intended recipient then you must not use, disseminate, distribute or copy any information contained in this email or any attachments. If you have received this email in error, please contact us immediately and delete this email. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT and only been able to have 1 SIP phone behind
Anton Krall wrote: Guys. Ive read on the wiki that a common problem with nat is that you can only have 1 sip phone behind, how do you get around this issue? Having a sip enabled router behind the nat like the GS 488 489 or 486? Or how have you done it without having any kind of linux box (SER or *) behind the nat. The idea is to have 2 or more sip clients behind some NAT and been able to connect to a remote asterisk box. I don't know why people think this. Any router with PAT (Port Address Translation) should work with multiple SIP clients behind NAT. Most routers support PAT (but may not call it that). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPAM SPAM SPAM SAM SPAM SPAM SPAM
Al wrote: Yeah... unbelieveable but true: spam, defined often us undesired bulk mail comes in many forms, including messages from this list. You receive messages from this list because YOU signed up for it. If you knew anything about email (or mailing lists), you could have looked in the email headers to find the unsubscribe address: http://lists.digium.com/mailman/listinfo/asterisk-users I tried several times - obviously unsuccessfully, as you can see - to unsubscribe from this list, and sice that did not work, i set my mail server to BOUNCE list messages. Oh great. Somebody with their own mailserver who doesn't know email, mail filters or mailing lists. But nope, nobody gets it - the mail server is deaf and dumb and keeps sending me dozens and dozens of messages. Which in the worst case scenario (unsubscribe don't work! - doubtful) you could filter out or simply drop at the mail server level. If there is anybody reading this who can reach the person(s) in charge of the mailing list server, please tell them that they are having (and causing) a problem. ;-) Somehow I think you're your own problem... Cheers, Jean-Michel. -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting SIP username for CallerID
On Wednesday 20 April 2005 11:55, Arunachala wrote: Hi Gavin, Just went through the code. There is a check in the code to check whether the CIDNum is a phone number (0-9,#,*) or no. If it is not a phone number, it is replaced with the default CIDNum asterisk. Hm, really smart :) If the SIP username can be alpha-numeric, I wonder what's prompted this check? If you really want to fix this, you can do the following in the code: Thanks for the tip - I'll be sure to give that a go :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPAM SPAM SPAM SAM SPAM SPAM SPAM
Al wrote: Yeah... unbelieveable but true: spam, defined often us undesired bulk mail comes in many forms, including messages from this list. I tried several times - obviously unsuccessfully, as you can see - to unsubscribe from this list, and sice that did not work, i set my mail server to BOUNCE list messages. But nope, nobody gets it - the mail server is deaf and dumb and keeps sending me dozens and dozens of messages. If there is anybody reading this who can reach the person(s) in charge of the mailing list server, please tell them that they are having (and causing) a problem. ;-) Al -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Have you tried to read and follow the last line of each message? -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing
To breifly recap Your main asterisk box runs linux, asterisk, ASTCC and MySQL Another box runs linux, mysql, apache The two sql servers are joined, updating each other? or have I missed something? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Fax detect/transfer problem?
I've been running into something similar. The fax detect works reliably to auto transfer the call. I see it Goto the new context, but instead of actually ringing the fax extension, it just fails over as though it's busy or something. I can always manually dial to the fax extension with success. Sometimes the auto transfer will ring successfully. But then for at least several minutes after a successful ring through, an auto transfer will continue to jump but fail to ring. I'll paste some log results in a follow up to illustrate. Detection seems to be working okay. If I call in with a voice call, my two voice SIP phones ring as normal, etc. If I call in with a fax call, it seems like Asterisk is detecting the fax correctly, but the fax never rings, and the call is just dropped. Fax rings fine and is answered if I call ext. 2003 though. Here's the event log: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Norton AntiSpam] [Asterisk-Users] IAX realtime HELP
Paul Dracevich wrote: I have been looking and have tried many many things but not have been able to get it working I am running Connected to Asterisk CVS-HEAD-04/14/05-11:56:07 currently running on localhost (pid = 1927) With the ingredients you provide you can earn at most an answer like It works for me! (Sorry, I could not resist to say that!) 18 lines of your signature deleted!!! bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Issues of reliability, hardware, platforms
I'm sure this has been debated before, I'd like to get peoples input. I see the hard drive as the single most likely point of failure on an * PBX. How reasonable would it be to run the OS and config files from a CF card, mount the /var/partition on a hard drive for the CDR recortds, logs and databases, and do some kind of test on bootup that would turn off those features if the hard drive was unavailable (failed)? I am messing around with a firewall that mounts from a cf card, it got me thinking about asterisk from a cf card, as there would be little chance of failure. Other than that, what have you done to ensure reliability. I am planning a RAID1 hard drive install for my serious pbx customers, in my experience that makes for a very reliable machine. Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (646)722-0001 Fax: (815)301-9759 Yahoo IM: [EMAIL PROTECTED] Skype ID: netconcepts ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA-841 Phone Review
It works for me with the default SIP settings. I am using the latest firmware. I have found that you have to restart * for it to pick up on the new PIN code however. On Wed, 2005-04-20 at 01:58, Master Abi wrote: if I have conf = 80,111 in meetme.conf, I dial 80# and connect to the conference, then I dial 111#, it indicates pin is incorrect. with other phones it works. Is there something special in the sipura config that will allow more digits after the # master Craig wrote: I found the speaker phone and the headset work ok on the original v.9.x software that came with the units, when I upgraded 2 of them to v 3.x the headset and speakerphone become unusable. I am looking to try and downgrade these units back to v 0.9 so I can use the headset on them. It would be nice to use share call appearances with * so I can turn them into a key telephone system like the system they replaced, but that is something I will have to work on. Apart from that they are brilliant for the price craig Date: Tue, 19 Apr 2005 12:36:09 -0400 (EDT) From: Paul Dugas [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sipura SPA-841 Phone Review To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain;charset=iso-8859-1 On Tue, April 19, 2005 10:05 am, Me said: If Sipura could make the headset jack solid, it would be a great, affordable phone in my opinion. Never had a problem with the headset jack. Now the speakerphone... They ought to be ashamed of themselves for advertising it as a feature of the unit. It absolutely stinks. Totaly useless. Also, very little in response to repeated request for attention on a fix other than try the latest firmware which does little other than making it even worse. Criminal! Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Testing my TDM01A
When at the CLI, show channels shows nothing. Look for ztcfg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P PCI-slot
Hi, The card need only 3.3V pci slot. 133MHz pci slots are 3.3V. On Wed, 2005-04-20 at 12:21 +0200, Tais M. Hansen wrote: Hi, I was just wondering about a comment I found in the voip-info.org wiki: The DIGIUM TE410 PRI card, requires a motherboard with a 64bit 3.3v PCI slot. Given the bandwidth requirements, it would be better to have a 133Mhz slot if available. Since the card seems to always clock at 33MHz. I can't really see how a 133MHz slot would make any difference? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk + Adit 600 questions
Is it possible to make Asterisk to execute a task when a called party answeres? Does the MGCP protocol include support for notificate when a call is answered? I have one Adit 600 w/ 40 FXS lines. When a call is initiated from such line to the PSTN through our E1 EuroISDN, I would like the Adit to, somehow, indicate on the FXS-line that the other user has answered (lifted his/her handset). By changing battery polarity or maybe an signal? Automatic equipment does use almost every FXS-line of ours, and they need to know when the call is answered. Any ideas please? -- BR Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference solution for 100+ users
Hi List, Hi! 1-Skype-like softphone for *. is there any? None that I know of. But IAX isn't bad in most of the firewalled environments, give it a try. It only has to get a udp socket open for an outbound connection (may well be NAT-ed) and to receive the answer packets back. 2-Just do audio streaming and have the customers use windows media player. (I dont know how to do this) This would mean exactly the same prerequisites as an iax-based solution as the media stream (usually udp) has to be received by the media players. One technique that circumvents this is using HTTP/1.1 streaming which may or may not work through an application level http-proxy. 3-Use some kind of Softphone with VPN... Again, if you are able to do an outside connect through the firewall (as with openvpn which uses udp or with ipsec which uses ip), you can also do some other things by this means (e.g. iax). 4- Do Softphone---Port 80--- SER---Asterisk w/meetme. Only reason for this might be an application level http-proxy that allows for outbound 'connect' calls, since I don't think you want to encapsule SIP in HTTP, do you?. And for the outbound 'connect' method, port 443 might be a better choice for your port number, but you have to choose a protocol that uses TCP and only one single socket for this to work. Maybe using iax over some sort of UDP-in-TCP tunnel could work (like zeebeedee). Whatever solution I come up with MUST allow anybody to listen in assuming nobody can change firewalls. Any one has already done this? Any feedback will be much appreciated. We're working on similar problems, so if you come up with a perfect solution, please let me know. Also, if you are interested in a commercial solution feel free to contact me off-list. Stefan Märkle -- Stefan Märkle Netpioneer GmbH Head Software Architect Beiertheimer Allee 18 [EMAIL PROTECTED] 76137 Karlsruhe, Germany ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Rhino Channel Bank
Hi I have just purchased a Rhino Channel Bank and am using it connected to asterisk via a digium TE410P. I am having problems with connecting phones to the channel bank. I have channel one connected to a patch panel, a line adaptor chonnecting the phone cord to the patch panel, and then the phone. When i pick up the channel bank does not detect this. The phone does not ring when called. Using a multimeter i checked voltage across the line and its 48V all the way up to the phone, so the wiring is fine... Any ideas as to what could be wrong? Regards Dan Goscomb ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax detected, but no fax extension
You need to add that to context where you have BackGround application
running.
house-day and house-night I believe.
On Wed, 2005-04-20 at 13:16, Ronald Wiplinger wrote:
Vladyslav wrote:
In your incoming context add
exten = fax,1,Goto(fax,2202,1)
It did not work ;-(
[incoming_88097680]
exten = s,1,NoOp(${CALLERIDNUM})
exten = s,2,Wait(1)
exten = s,3,SetCallerId(9${CALLERIDNUM})
exten = s,4,GotoIfTime(08:00-21:20|sun-sat|*|*?house-day,s,1)
exten = s,5,Goto(house-night,s,1)
exten = fax,1,Goto(fax,2201,1)
[fax]
exten = 2201,1,Macro(faxreceive)
;exten = 2202,1,Macro(faxreceive)
;exten = 2203,1,Macro(faxreceive)
exten = h,1,system(/usr/local/sbin/mailfax ${FAXFILE} \
${EMAILADDR} ${CALLERIDNUM} ${CALLERIDNAME})
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten = s,3,rxfax(${FAXFILE})
exten = s,103,SetVar([EMAIL PROTECTED])
exten = s,104,Goto(3)
On Wed, 2005-04-20 at 12:26, Ronald Wiplinger wrote:
-- Starting simple switch on 'Zap/3-1'
-- Executing NoOp(Zap/3-1, 9229443944-) in new stack
-- Executing Answer(Zap/3-1, ) in new stack
-- Executing Zapateller(Zap/3-1, ) in new stack
-- Executing BackGround(Zap/3-1, custom/Welcome) in new stack
-- Playing 'custom/Welcome' (language 'default')
Apr 20 16:49:29 NOTICE[29109]: chan_zap.c:4300 zt_read: Fax detected,
but no fax extension
-- Executing BackGround(Zap/3-1, if-u-know-ext-dial) in new stack
-- Playing 'if-u-know-ext-dial' (language 'default')
-- Executing Dial(Zap/3-1,
SIP/601SIP/621ZAP/1r1SIP/610|30|tr) in new stack
-- Called 601
Apr 20 16:49:43 NOTICE[29109]: app_dial.c:973 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3)
-- Called 1r1
Apr 20 16:49:43 NOTICE[29109]: app_dial.c:973 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3)
-- Zap/1-1 is ringing
-- SIP/601-49e2 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- SIP/601-49e2 answered Zap/3-1
-- Hungup 'Zap/1-1'
I tried to use the exension 2201, but it did not work, so I have changed
it to s, but it does not work either.
[fax]
exten = s,1,Macro(faxreceive)
;exten = 2202,1,Macro(faxreceive)
;exten = 2203,1,Macro(faxreceive)
exten = h,1,system(/usr/local/sbin/mailfax ${FAXFILE} \
${EMAILADDR} ${CALLERIDNUM} ${CALLERIDNAME})
bye
Ronald
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[Asterisk-Users] A question about queues
Hi there, quick question about queues (B (BWhen calling a queue (which contains eg 4 extensions) it tells me what (Bposition I am in the queue and then plays some music$B!D(Jthat is fine$B!D(J (Bhowever, If there is no-one in the queue , it tells me that im first in line (Band then plays hold music while the phones ring. This is annoying my callers (Bquite a bit . How do I get it so that if I ring the queue, it just puts me (Bstraight through to one of the available 4 phones, and only if all 4 phones (Bare busy (ie on calls) then announce a position in the queue and play music? (B (BFor example (B (BUser 1 dials 7272 $B"*(J goes through to agent 1 (BUser 2 dials 7272 $B"*(J goes through to agent 2 (BUser 3 dials 7272 $B"*(J goes through to agent 3 (BUser 4 dials 7272 $B"*(J goes through to agent 4 (BUser 5 dials 7272 $B"*(J announces message that you are first in line (BUser 6 dials 7272 $B"*(J announces message that you are second in line (B (B (BAny help on this would be greatly appreciated (B (B (B___ (BAsterisk-Users mailing list (BAsterisk-Users@lists.digium.com (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IPSwitchBoard connects to CDR
The latest Version 0.92 of IPSwitchBoard can connect to your MySQL database and show you call records with filtering on extension and from- and to date. IPS now also will check if theres a newer version available on start-up and offer to start the download. The Configuration page has changed layout (Hopefully for the better) Download for free here: http://ipswitchboard.thorben.dk Thorben PS: If you have the possibility of giving me access to a server configured with a CAPI card, I would be grateful. I would love to add support for CAPI. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] capi segfault when incoming call is answered
On 4/7/05, Thomas Andrews [EMAIL PROTECTED] wrote: On Thu, Apr 07, 2005 at 10:15:09AM +0200, Thomas Andrews wrote: I have a Fritz! card set up to use capi, however when incoming calls to the card are answered, asterisk segfaults. Have you tried a make clean then make install in the chan_capi source directory make sure the header files are built correctly. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rhino Channel Bank
I don't know about channel banks, but when you go T1 to T1 device with a cable, you need the RX/TX pairs cross connected. Do you have a T1 crossover cable in play or a straight through? On 4/20/05, Dan Goscomb [EMAIL PROTECTED] wrote: Hi I have just purchased a Rhino Channel Bank and am using it connected to asterisk via a digium TE410P. I am having problems with connecting phones to the channel bank. I have channel one connected to a patch panel, a line adaptor chonnecting the phone cord to the patch panel, and then the phone. When i pick up the channel bank does not detect this. The phone does not ring when called. Using a multimeter i checked voltage across the line and its 48V all the way up to the phone, so the wiring is fine... Any ideas as to what could be wrong? Regards Dan Goscomb ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rhino Channel Bank
certainly do... and asterisk and the rhino see each other... On Wed, 2005-04-20 at 08:38 -0400, BJ Weschke wrote: I don't know about channel banks, but when you go T1 to T1 device with a cable, you need the RX/TX pairs cross connected. Do you have a T1 crossover cable in play or a straight through? On 4/20/05, Dan Goscomb [EMAIL PROTECTED] wrote: Hi I have just purchased a Rhino Channel Bank and am using it connected to asterisk via a digium TE410P. I am having problems with connecting phones to the channel bank. I have channel one connected to a patch panel, a line adaptor chonnecting the phone cord to the patch panel, and then the phone. When i pick up the channel bank does not detect this. The phone does not ring when called. Using a multimeter i checked voltage across the line and its 48V all the way up to the phone, so the wiring is fine... Any ideas as to what could be wrong? Regards Dan Goscomb ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone problems to non-na numbers
Yes, same problem here. Sign-ed up with VoipJet and seems to work just fine (prices for most areas we call are cheaper too from what I saw). Only been using them for 24 hours so can't say much about long-term stability, but so far so good. Pedro On 4/19/05, Matthew Asham [EMAIL PROTECTED] wrote: Is anyone else having problems with Nufone dialing international (non NA) numbers? Pretty much every intl number dialed comes up with a voice intercept saying the call could not be completed as dialed. Tried it with two separate accounts, and the numbers themselves work from the local telco. The problem appears to have started within the last few days (and yes I have emailed [EMAIL PROTECTED], just wondering if we're the only ones having the problem). Matthew -- Matthew Asham - the B.C. Wireless Network Society www.bcwireless.net - +1 604 484 5289 x1006 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with [codec_g729.c:196 g729tolin_framein: Invalid data]
Hi All Can anyone help with this message? We are using a Swissvoice with G729 on the latest CVS of Asterisk Apr 20 15:11:34 WARNING[5123]: codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end) Apr 20 15:11:34 WARNING[5123]: codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end) Apr 20 15:11:34 WARNING[5123]: codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end) Apr 20 15:11:34 WARNING[5123]: codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end) Apr 20 15:11:34 WARNING[5123]: codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end) Apr 20 15:11:34 WARNING[5123]: codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end) Apr 20 15:11:34 WARNING[5123]: codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end) Apr 20 15:11:34 WARNING[5123]: codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end) Apr 20 15:11:34 WARNING[5123]: codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end) Apr 20 15:11:36 WARNING[5123]: codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end) Apr 20 15:11:36 WARNING[5123]: codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end) Thanks D ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RealTime ignoring switch= Realtime/context@realtime_ext
- Original Message - From: Ronald Wiplinger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 20, 2005 4:42 AM Subject: Re: [Asterisk-Users] RealTime ignoring switch= Realtime/[EMAIL PROTECTED] Me wrote: OK, been messing with RealTime like a week off and on, I can safely say it's killing me! I have dug and dug and dug to find what I am missing, no dice. I am running the latest version of * from CVS as of about a week ago. Call comes in from a PRI into the todd_test_1 extension, if I uncomment the lines for the _888 number directly in the extensions.conf file the call is answered without a problem. If I comment the lines and just leave the switch in place it's suppose to lookup the extensions from the mysql table from what I understand. All I get when calling in from the PRI is this: -- Extension '8885551212' in context 'todd_test_1' from '2145551212' does not exist. Rejecting call on channel 0/1, span 1 It appears that the switch command is totally being ignored. I also checked the MySQL logs to see if Asterisk/RealTime was even hitting it but I see nothing in the MySQL logs at all that would indicate Asterisk is talking to it. My phone numbers/passwords etc. have been changed but most everything else in my configs are as is. Any help would be appreciated, I am sure I am just missing something really simple and I am gonna smack myself in the head when it's brought to my attention. ### extensions.conf# [todd_test_1] switch = Realtime/[EMAIL PROTECTED] shouldn't it be Realtime/[EMAIL PROTECTED] or [todd_test_1] include = todd_test_2 [todd_test_2] switch = Realtime/[EMAIL PROTECTED] ??? BTW, the numbering of the priorities should increase: 1;todd_test_2;_888NXX;1;Wait;2 2;todd_test_2;_888NXX;2;SayNumber;102 3;todd_test_2;_888NXX;1;Playback;pbx-invalid bye Ronald Well, I am confused then about two things.. 1- In switch = Realtime/[EMAIL PROTECTED] I am referring to todd_test_2 which is my context inside of the DB for the records I am referencing, I was not aware that this context also needed to exist within the text file extensions.conf. 2- Can I not have one context within the extensions.conf that has the switch command in it and then as many other context as I like within the database? I thought this was the whole idea, controlling the extensions from the DB which in my opinion includes using different context. 3- Someone mentioned to me the other day that I shouldn't have the same context in the DB as I have in the text file. For example, I think they told me it was a bad idea to have a context within the extensions.conf called todd_test_1 which had a switch command in it, then also have todd_test_1 as the context in the DB. Maybe I totally misunderstood this person the other day regarding this. Basically this is why I now have two context todd_test_1 and todd_test_2. Regarding my priority numbering, I know it was off but I am pretty sure that based on the error I am getting in the CLI when calling in as well as the fact that * never hits MySQL at all according to the logs, I would say the process never makes it to the database at all to even get to this error about the priority. But, thanks for letting me know, sometimes it's little things like this that can bugger you up along the way. For your reference the error is below, this shows that it dies within todd_test_1: -- Extension '8885551212' in context 'todd_test_1' from '2145551212' does not exist. Rejecting call on channel 0/1, span 1 Again, if I just add some lines to handle the call right under the switch command, all works well which tells me the switch command is likely being ignored totally. FYI, I did install Asterisk-Addons, I am running the latest CVS as of a week or so ago, I do have the MySQL client and header libs installed. The MySQL server is on a box on the same LAN and is operational for other live services right now. I have double and triple checked my MySQL permissions, besides if it was rejected for permission reasons, I would show it in my MySQL logs. I hate to be a ding bat here but, can someone tell me how to turn on Debug mode and where the debug logs show up? I am sure there is a Wiki page on this so a URL would be great, I will go dig for it some more now. Thanks folks for all the help so far! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] capi segfault when incoming call is answered
On Thu, Apr 07, 2005 at 10:15:09AM +0200, Thomas Andrews wrote: I have a Fritz! card set up to use capi, however when incoming calls to the card are answered, asterisk segfaults. Have you tried a make clean then make install in the chan_capi source directory make sure the header files are built correctly. I'm not totaly sure, but I think I had the same problem when I upgraded from capi4k-utils-2004-10-06.tar.gz to a newer version. As soon as I downgraded, it started working normally again. Good luck, Ivan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] US$200 bounty for * paging feature
Henry Devito wrote:
I am already doing this with AGI, PERL, and PHP to set up the page
groups. I will release the code as open source if people are
interested. I'm not the best PERL scripter in the world but it works.
Attached is the agi I'm using. This is a modified script from a post on
voip-info. This works with our Cisco's that are setup like this: Line1
- XXX and Line2 -- XXX_i (for intercom).
The modifications from the stock script are paging to SIP/XXX_i (not
SIP/XXX), dynamic conferences based on original callerid, and playing
the beeps (Cisco just answers so this gives users a warning).
There is code to see if SIP/XXX is in use, and if so not to call
SIP/XXX_i, but the users wanted to see all pages so it is commented out.
Zones would be real easy with some arrays (as the conference is dynamic
based on the person calling) and the variables are there to check inuse,
etc.
extensions.conf:
[paging]
exten = *999,1,AGI(page.agi|${CALLERIDNUM})
exten = *999,2,Wait(1)
exten = *999,3,Playback(beep)
exten = *999,4,MeetMe(${CALLERIDNUM},dtqpA)
exten = *999,5,Hangup
[add-to-allcall]
exten = _X.,1,Playback(beep)
exten = _X.,2,MeetMe(${EXTEN},dmqpwx)
exten = _X.,3,Hangup
Really easy to modify. Have fun.
Again this could be cleaner, but I got that other script working and
haven't had the time or need to clean it up.
Jeb Campbell
[EMAIL PROTECTED]
#!/usr/bin/perl
#
#
# allcall.agi will add all your Polycom sip phones to a meet me
# conference for use in office wide paging
#
# It takes arguments in the form of SIP/ where is your
# sip extension. (can be any number of digits) The first argument
# is the originating caller and additional arguments are any other
# phone lines you wish to exclude
#
use strict;
use File::Copy;
# A Few Variables to Set and Initialize
#
#
my $outgoing = '/var/spool/asterisk/outgoing';
my $temp = '/var/tmp';
my $asterisk = '/usr/sbin/asterisk';
my $audio_out= 'console/dsp';
my @bypass = ();
my @meetme_calls = ();
my @rawsips = ();
my @sips = ();
my @intercoms= ();
my $callerid = Error;
# Parse out the Sip phones to exclude
#
# This truly shows my lack of understanding of perl
#
foreach (@ARGV) {
@bypass = split ( / /, $_ );
}
# This is our originating caller. I need his
# callerid so that others will know who the paging
# pest is:
#
$callerid = $bypass[0];
$callerid =~ s-SIP/--g;
# Setup an array with all the sip phones
#
# I think I could use the Asterisk::AGI here
# and also the incominglimit in sip.conf to accomplish
# this, but I'm not that good.
@rawsips = `$asterisk -rx sip show inuse`;
chomp(@rawsips);
shift (@rawsips);
shift (@rawsips);
@rawsips = sort (@rawsips);
#Jeb
# split to sips and intercoms
@sips = grep ( /^\d{3,4} / , @rawsips );
@intercoms = grep ( /^\d{3,4}_i / , @rawsips );
# Now check each sip phone to see if it's in use and also
# against our exclude list. If it passes both, it's
# added to our array of calls to make
foreach (@sips) {
my $sipphone = $1 if /(\d{3,4}) /;
my $sipinuse = substr( $_, 16, 1 );
unless ( grep ( /$sipphone/i, @bypass ) ) {
#if ( grep ( /${sipphone}_i/i , @intercoms ) and $sipinuse == 0 ) {
if ( grep ( /${sipphone}_i/i , @intercoms ) ) {
push ( @meetme_calls, make_call(SIP/${sipphone}_i) );
#push ( @meetme_calls, SIP/${sipphone}_i );
}
}
}
# The array is complete. The push line is uncommented
# if you want to add audio out to the intercom
#
#
# push ( @meetme_calls, make_call($audio_out) );
# Now move each call file to the outgoing directory
#
# Here's some more perl ugly
#
foreach my $call (@meetme_calls) {
move( $temp . '/' . $call, $outgoing . '/' . $call );
}
#print join(\n,@meetme_calls) . \n;
exit 0;
sub make_call {# makes the call file and returns the name
my $stripslash = $_[0];
$stripslash =~ s/\///g;
my $tempcall = $temp . '/' . $stripslash . $$;
my $callbase = $stripslash . $$;
open( call, $tempcall );
print call EOF;
Channel: $_[0]
MaxRetries: 1
Retry: 0
RetryTime: 60
Context: add-to-allcall
Extension: $callerid
Priority: 1
SetVar: ALERT_INFO=Ring Answer
CallerID: All-Call $callerid
EOF
close(call);
return $callbase;
}
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Re: [Asterisk-Users] US$200 bounty for * paging feature
On Tue, Apr 19, 2005 at 06:24:09PM -0700, trixter http://www.0xdecafbad.com said: as a whole. I enjoy cheap computers, if it were not for microsoft creating windows, making computers easier to use for everyone, the mass production and highly competitive hardware market would not exist. If that didnt happen the $300 computer of today would likely not exist, and if it did it would cost more like computers did 20 years ago, $2000+ for a bare system. rantmode Um, that's total bullshit. Low computer prices and ease of use would have existed if MS was never around. You completely dismiss billions of man hours of hard work by those outside MS making advances in hardware and software around the world. To make a statement like that, you show a total lack of knowledge of the industry. I have worked for over 10 years in the software development industry and Then you entered the industry far too late to know the real history of computing, have read too many MS revisionist history books, or were hiding under a rock. For example, The Amiga for example had a wonderful OS, great multi-tasking, awesome windowing interface etc. over 10 years before MS (some would argue longer.) Comodore didn't have a chance against the mighty combo of IBM, MS, Compaq. and other x86 hardware and software vendors in the business world (the Amiga was originally designed as a game machine and could never escape the stigma AND had the same bone-headed single hardware source issue that Apple has. Poor management / marketing also contributed to the companies death.) (Speaking of Apple, it boggles the mind that it took them over 15 years to add multi-tasking to their product line - and yes, I am dismissing their prior failed unix attempt.) MS has no effective competition due to their illegal business practices, killing off alternatives (BeOS is a recent example) by pressuring large ISV's to only write for the Windows OS, restrictive contracts with hardware vendors, and other sleezy tactics. They effectivly killed Java on the desktop. They continue with a powerful FUD campaign against Linux, Apple, Firefox, etc. I could go on, and on, and on. In my opinion, MS has held the world of computing back about 15 years (unless you think that having the worst security model / track record in computing history, and proprietary interfaces and file formats with no publicly available documentation is a good thing.) Unfortunately the reality of business means that we have to deal with this horrible corporation and their aweful software. MS and their single platform (for servers and desktop anyway) means that we are still saddled with the horrible x86 architecture, the interrupt structure, bus, bios, etc. (essentially most everything about a PC.) By the way, that architecture is why it's so hard to make reliable hardware, why we need an external card to get a reliable timer device, etc. Before you spout off about how great MS has been to the industry, maybe you should learn a little about that industry and it's history first, M-kay? /rantmode ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A question about queues
Can you post your config's? What version of * are you using? This doesn't (Bhappen on any of my queues. I have queues set up on several customers (Bsystems. If there are agents/members available the caller rings them (Bdirectly, no announcements played. (B- Original Message - (BFrom: "Brett, Gary" [EMAIL PROTECTED] (BTo: "Asterisk Users Mailing List - Non-Commercial Discussion" (Basterisk-users@lists.digium.com (BSent: Wednesday, April 20, 2005 7:21 AM (BSubject: [Asterisk-Users] A question about queues (B (B (B Hi there, quick question about queues (B (B When calling a queue (which contains eg 4 extensions) it tells me what (B position I am in the queue and then plays some music$B!D(Bthat is fine$B!D(B (B however, If there is no-one in the queue , it tells me that im first in (B line (B and then plays hold music while the phones ring. This is annoying my (B callers (B quite a bit . How do I get it so that if I ring the queue, it just puts me (B straight through to one of the available 4 phones, and only if all 4 (B phones (B are busy (ie on calls) then announce a position in the queue and play (B music? (B (B For example (B (B User 1 dials 7272 $B"*(B goes through to agent 1 (B User 2 dials 7272 $B"*(B goes through to agent 2 (B User 3 dials 7272 $B"*(B goes through to agent 3 (B User 4 dials 7272 $B"*(B goes through to agent 4 (B User 5 dials 7272 $B"*(B announces message that you are first in line (B User 6 dials 7272 $B"*(B announces message that you are second in line (B (B (B Any help on this would be greatly appreciated (B (B (B ___ (B Asterisk-Users mailing list (B Asterisk-Users@lists.digium.com (B http://lists.digium.com/mailman/listinfo/asterisk-users (B To UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users (B (B___ (BAsterisk-Users mailing list (BAsterisk-Users@lists.digium.com (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FXO lines on TDM04B not responding
-Original Message-From: Goutam Shaw [mailto:[EMAIL PROTECTED]Sent: Tuesday, April 19, 2005 11:22 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] FXO lines on TDM04B not responding I ran into the situation where 3 of the 4 lines on the FXO card stopped responding to the incoming call. I have 2 cards with a total of 8 FXO lines. A month ago we have replaced the old cards with the latest Digium X100M RevB. Before the card replacement the whole system used to get locked up but this time only 3 of the lines were not resonding on one card and the rest were fine. Digium guys dont say anything except reloading the driver and asterisk. However, in telephony as we all know this is not an acceptable solution. Is Digium HW is really bad.[David Brodbeck]Ihave mine automatically reloadearly on Sunday morning, when call volume is pretty much nonexistent, to get around this problem. I agree it sucks to have to do this. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Which free calling card app most suited forcommercial use?
My opinion is that both are Crap. Both of them have a flaw in their base design, which is difficult to explain in a post like this. Suffice to say that these two applications neither support nor designed for mutilpe routes ( multiple Area codes with Destination groups) nor multiple rate plans(Provider rates or buying rates and selling rates) nor multiple business models(retail, wholesale, corporate customers) Hence both of them cannot be the base for a commercial grade billing system for a Calling card Model. These apps canot be used for a realtime call control using CPD (Call Progress Detection) and Prepaid amounts for a post-paid Billing and call disconnect. Without this very essential feature for a commercial Calling card billing application, you would be better off calculating the calls from the Master.csv file for a post paid bill management. AreskiCC is a little more thought-driven and hence can be improved upon. If anyone is interested in developing a full fledged billing system, I have created a deisgn document ( a very elaborate rough draft infact) which I can share with you. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of snacktime Sent: Tuesday, April 19, 2005 5:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Which free calling card app most suited forcommercial use? I'm working on an * billing system, and instead of reinventing the wheel I would prefer to use an existing codebase for the calling card portion. The two that look most promising are astcc and the * prepaid billing application that uses postgresql. Any comments? Chris NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone problems to non-na numbers
On Wed, 20 Apr 2005, Pedro wrote: Yes, same problem here. Sign-ed up with VoipJet and seems to work just fine (prices for most areas we call are cheaper too from what I saw). Only been using them for 24 hours so can't say much about long-term stability, but so far so good. I had this problem and it turned out that I was sending my outgoing calls to switch-2. I switched my peer to send calls to switch-1 and things came right. Jeremy says that switch-2 is considered the backup and call routes on there are being redone (or words to that effect). Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] General voip mailing list
Does anyone here know of any general, good voip mailing list? I am having a hard time with broadvoice and the company is not answering its phone. TIA, GM ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A question about queues
You could work this out with setgroup checkgroup, and create 2 queues, (Bif checkgroup jumps to 101+ (its the fifth caller) it goes to the (Bsecond queue. Make sure that the only difference between the second (Band first queue is the announcement. (BUsing the above you will have to know in advance what the number of members are. (B (BOn 4/20/05, Brett, Gary [EMAIL PROTECTED] wrote: (B Hi there, quick question about queues (B (B When calling a queue (which contains eg 4 extensions) it tells me what (B position I am in the queue and then plays some music$B!D(Bthat is (B fine$B!D(B (B however, If there is no-one in the queue , it tells me that im first in line (B and then plays hold music while the phones ring. This is annoying my callers (B quite a bit . How do I get it so that if I ring the queue, it just puts me (B straight through to one of the available 4 phones, and only if all 4 phones (B are busy (ie on calls) then announce a position in the queue and play music? (B (B For example (B (B User 1 dials 7272 $B"*(B goes through to agent 1 (B User 2 dials 7272 $B"*(B goes through to agent 2 (B User 3 dials 7272 $B"*(B goes through to agent 3 (B User 4 dials 7272 $B"*(B goes through to agent 4 (B User 5 dials 7272 $B"*(B announces message that you are first in line (B User 6 dials 7272 $B"*(B announces message that you are second in line (B (B Any help on this would be greatly appreciated (B (B ___ (B Asterisk-Users mailing list (B Asterisk-Users@lists.digium.com (B http://lists.digium.com/mailman/listinfo/asterisk-users (B To UNSUBSCRIBE or update options visit: (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (B (B___ (BAsterisk-Users mailing list (BAsterisk-Users@lists.digium.com (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS-HEAD and CheckGroup/SetGroup
On 4/20/05, Sean A. Newton [EMAIL PROTECTED] wrote:
Do the SetGroup and CheckGroup functions behavior differently in CVS-HEAD
vs CVS v1-0?
When I upgrade to CVS-HEAD my call waiting disable doesn't seem to work,
using:
exten = s,1,SetGroup(SIP${ARG1})
exten = s,2,CheckGroup(1)
exten = s,3,Dial(Sip/${ARG1},15,t)
Do you not need a
exten = s,103,Congestion()
otherwise the checkgroup has nowhere to go ?
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Re: [Asterisk-Users] A question about queues
it sounds like the default behaivor of an [EMAIL PROTECTED] setup. (B (Bnot that I am knocking [EMAIL PROTECTED] in anyway - its a great way to test (Bnew features. (B (BOn 4/20/05, Henry Devito [EMAIL PROTECTED] wrote: (B Can you post your config's? What version of * are you using? This doesn't (B happen on any of my queues. I have queues set up on several customers (B systems. If there are agents/members available the caller rings them (B directly, no announcements played. (B - Original Message - (B From: "Brett, Gary" [EMAIL PROTECTED] (B To: "Asterisk Users Mailing List - Non-Commercial Discussion" (B asterisk-users@lists.digium.com (B Sent: Wednesday, April 20, 2005 7:21 AM (B Subject: [Asterisk-Users] A question about queues (B (B Hi there, quick question about queues (B (B When calling a queue (which contains eg 4 extensions) it tells me what (B position I am in the queue and then plays some music$B!D(Bthat is (B fine$B!D(B (B however, If there is no-one in the queue , it tells me that im first in (B line (B and then plays hold music while the phones ring. This is annoying my (B callers (B quite a bit . How do I get it so that if I ring the queue, it just puts me (B straight through to one of the available 4 phones, and only if all 4 (B phones (B are busy (ie on calls) then announce a position in the queue and play (B music? (B (B For example (B (B User 1 dials 7272 $B"*(B goes through to agent 1 (B User 2 dials 7272 $B"*(B goes through to agent 2 (B User 3 dials 7272 $B"*(B goes through to agent 3 (B User 4 dials 7272 $B"*(B goes through to agent 4 (B User 5 dials 7272 $B"*(B announces message that you are first in line (B User 6 dials 7272 $B"*(B announces message that you are second in line (B (B (B Any help on this would be greatly appreciated (B (B (B ___ (B Asterisk-Users mailing list (B Asterisk-Users@lists.digium.com (B http://lists.digium.com/mailman/listinfo/asterisk-users (B To UNSUBSCRIBE or update options visit: (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (B (B ___ (B Asterisk-Users mailing list (B Asterisk-Users@lists.digium.com (B http://lists.digium.com/mailman/listinfo/asterisk-users (B To UNSUBSCRIBE or update options visit: (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (B (B___ (BAsterisk-Users mailing list (BAsterisk-Users@lists.digium.com (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPAM SPAM SPAM SAM SPAM SPAM SPAM
On 4/20/05, Al [EMAIL PROTECTED] wrote: Yeah... unbelieveable but true: spam, defined often us undesired bulk mail comes in many forms, including messages from this list. I tried several times - obviously unsuccessfully, as you can see - to unsubscribe from this list, Obviously the problem is you, because it works for all of us. and sice that did not work, i set my mail server to BOUNCE list messages. If you set your server to bounce it, how do you still get it? do you check in the queue to make sure that it's archived as bad mail? But nope, nobody gets it - the mail server is deaf and dumb and keeps sending me dozens and dozens of messages. If there is anybody reading this who can reach the person(s) in charge of the mailing list server, please tell them that they are having (and causing) a problem. ;-) I think it's you that is having a problem, eat breadfast and then try again. Al -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G723.1 and G729 on Athlon 64
I would like to install G723.1 and G729 on an Athlon 64. I looked at http://readytechnology.co.uk but I could not get a clue how to compile / get all the things for an Athlon. It seems it is only for Intel architecture, ... Has anybody a clue how to get G723.1 and G729 on an Athlon 64 to work? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone have a GXP-2000 working with Asterisk yet?
http://www.grandstream.com/y-gxp2000.htm Looks like the phone is $139 from DigitNetworks.. Price looks good.. If anyone has one working with Asterisk, how does it sound/work? Also, does it have caller ID with name? The Budgettones only support plain old callerID number.. Very annoying!! Thanks, - Andre ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference solution for 100+ users
Thanks Vamsi for your feedback. I would love to do it with Asterisk since I can do a lot more eventually. I did try a couple of iax2 clients and I couldnt go past the FW in a particular customer. Thanks for your email. Regards, Sergio, Date: Wed, 20 Apr 2005 09:28:24 +0530 From: Vamsi Pottangi [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Conference solution for 100+ users To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Since all would be listening, it's good to have a web streaming. Users could just use the media players rather than going for new softphones. This mailing list is not the appropriate one to discuss the above. But if you want to consider the asterisk solution, we can very well have the audience to participate in conference say for QA session. You could could use IAX2 clients behind the firewalls. ~Vamsi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 incoming audio stutter
The effect that I am seeing is that a call starts off fine, but suddenly after a few minutes the audio coming into Asterisk via OH323 gets very broken up to the point of being about 90% silence with occasional brief snippets of audio getting through. hi, any errors or warnings in Asterisk console? more info please... __ Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. http://mobile.yahoo.com/maildemo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P PCI-slot
On Wednesday 20 April 2005 13:56, Domjan Attila wrote: I was just wondering about a comment I found in the voip-info.org wiki: The DIGIUM TE410 PRI card, requires a motherboard with a 64bit 3.3v PCI slot. Given the bandwidth requirements, it would be better to have a 133Mhz slot if available. Since the card seems to always clock at 33MHz. I can't really see how a 133MHz slot would make any difference? The card need only 3.3V pci slot. 133MHz pci slots are 3.3V. Yes, I'm very well aware of this. But telling people that the Digium card would perform better in a 133MHz slot seems quite odd to me. -- Regards, Tais M. Hansen ComX Networks A/S Tel: +45-70257474 Fax: +45-70257374 pgpnht2N1cyAt.pgp Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G723.1 and G729 on Athlon 64
Ron, Here is what I think. Ready technology code will compile only with Intel IPPs. But there are two options, either 1) you compile the codec using Intel IPPs to provide the C and other base library functions, in which case you have to have the Intel libraries and license available on each system, you are running the .so Codec file. In this case the .so file size will remain smaller. When I did this the codec_g729X.so file size is 300k on a Redhat9. G723 is about the same. 2) In the second option you can chose to compile the codec by opting to embed the libraries into the .so file itself. This way when the module is loaded, it will not look for intel libraries. But the file size will be larger. When I did this the G729 codec file size 418,653 bytes. Probably the second option can be used and the module file copied to Athlon based Asterisk box and that might work. I don't use AMD hence I did not test this. But if you test it, please let me know, one way or the other. Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Wednesday, April 20, 2005 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] G723.1 and G729 on Athlon 64 I would like to install G723.1 and G729 on an Athlon 64. I looked at http://readytechnology.co.uk but I could not get a clue how to compile / get all the things for an Athlon. It seems it is only for Intel architecture, ... Has anybody a clue how to get G723.1 and G729 on an Athlon 64 to work? bye Ronald NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference solution for 100+ users
Stefan, Thanks for your feedback. I am testing everything to find the right solution. It is an interesting project since the listeners will vary everytime. Most of them are corporate users and thus unable to touch the corporate FW. I found a large international corporation that allows me to run tests from within their networks but without touching the FW. So that is good. So far none of the iax2 clients worked. The only thing that works is Skype and the MSN only for internal voice. They cant use MSN to speak with other MSN users outside their network. So I assume they either opened the Skype ports or Skype just happened to work. I will start playing with streaming audio and see what happens. My only concern here is that streaming usually does a little buffering before playing the audio. This might be an issue since they already have a chat system for questions an answers. So if someone asks a question via chat the speaker might get it when he is on another topic. But I would love to make Asterisk work. I am not giving up on it and that's why Im on this list. Take care and I will let you know how it turns out and / or if I need help with a solution. Thanks againg Sergio Veltri www.pointhorizon.com Message: 25 Date: Wed, 20 Apr 2005 14:06:18 +0200 From: Stefan M?rkle [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Conference solution for 100+ users To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hi List, Hi! 1-Skype-like softphone for *. is there any? None that I know of. But IAX isn't bad in most of the firewalled environments, give it a try. It only has to get a udp socket open for an outbound connection (may well be NAT-ed) and to receive the answer packets back. 2-Just do audio streaming and have the customers use windows media player. (I dont know how to do this) This would mean exactly the same prerequisites as an iax-based solution as the media stream (usually udp) has to be received by the media players. One technique that circumvents this is using HTTP/1.1 streaming which may or may not work through an application level http-proxy. 3-Use some kind of Softphone with VPN... Again, if you are able to do an outside connect through the firewall (as with openvpn which uses udp or with ipsec which uses ip), you can also do some other things by this means (e.g. iax). 4- Do Softphone---Port 80--- SER---Asterisk w/meetme. Only reason for this might be an application level http-proxy that allows for outbound 'connect' calls, since I don't think you want to encapsule SIP in HTTP, do you?. And for the outbound 'connect' method, port 443 might be a better choice for your port number, but you have to choose a protocol that uses TCP and only one single socket for this to work. Maybe using iax over some sort of UDP-in-TCP tunnel could work (like zeebeedee). Whatever solution I come up with MUST allow anybody to listen in assuming nobody can change firewalls. Any one has already done this? Any feedback will be much appreciated. We're working on similar problems, so if you come up with a perfect solution, please let me know. Also, if you are interested in a commercial solution feel free to contact me off-list. Stefan Märkle -- Stefan Märkle Netpioneer GmbH Head Software Architect Beiertheimer Allee 18 [EMAIL PROTECTED] 76137 Karlsruhe, Germany ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can I do something with Caller-ID?
I have setup my system to give a company announcement if somebody calls, ... I would like to avoid these announcements, if the caller is known by the system. Each caller I would like to put into a database with name. Now we know them! If we know them, we do not announcement. Is there anything out there to accomplish this? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wait in Dial String
Dear All,
My boss has placed a requirement for me to forward all our IDD calls
through a partner's IDD service, which requires us to call a 8 digit
number, wait for 1 sec, before we send in the foreign number we're trying
to call.
As I can't find anything on getting the PBX to wait, i'm only removing the
1st 3 digits (900) and sending in an extra 1 to simulate the wait. It
works, but not all the time. Is there anyway that I can place a wait
command here? I'm tried placing w / p but both don't works. Would like to
seek your kind assistance!
exten = _9001.,1,Dial(Zap/g1/64919669,,D(${EXTEN:3}),)
exten = _9001.,n,Hangup()
Best Regards,
==
David Choo
Systems Engineer
Business Technology Division
Engineered for Changing Businesses
Espore Corp Pte Ltd
68 Kallang Pudding Rd
#04-03 SYH Logistics Bldg
Singapore 349327
Tel: 65-68487806
Fax : 65-6842 2724
E-mail :[EMAIL PROTECTED]
=
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Re: [Asterisk-Users] Sample AGI Scripts in C needed.
Here you have a sample that i used to test that agi was doing well.
#include stdio.h
main()
{
char line[80];
setlinebuf(stdout);
setlinebuf(stderr);
while (1)
{
fgets(line,80,stdin);
if ( strlen(line) = 1 )
{
break;
}
}
printf(SAY NUMBER 55 \\\n);
fgets(line,80,stdin);
printf(SAY NUMBER 66 \\\n);
fgets(line,80,stdin);
}
Good look.
On 4/20/05, Bharat M. Sarvan [EMAIL PROTECTED] wrote:
Hello Everybody,
Could anybody please send me sample AGI scripts in
C.? I was looking forward to code AGI scripts in C. It would be very kind
enough if you do the needful.
Regards,
Bharat M. Sarvan
EZZI BPO Pvt Ltd.,
C2-7, 2nd Floor, Bramha Estate,
NIBM Junction, Khondwa,
PUNE 410048.
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Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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Re: [Asterisk-Users] G723.1 and G729 on Athlon 64
Ronald Wiplinger napisa(a): I would like to install G723.1 and G729 on an Athlon 64. I looked at http://readytechnology.co.uk but I could not get a clue how to compile / get all the things for an Athlon. It seems it is only for Intel architecture, ... Has anybody a clue how to get G723.1 and G729 on an Athlon 64 to work? It isn't possible, because this patch depends on Intel's proprietary code - IPP. Another platforms performs cold restart after loading codec. On the other hand even on Intel platform inband DTMF doesn't work. -- Marcin Kwiatkowski Senior IT Specialist Telebonus Sp. z o.o. Legionow 30 43-300 Bielsko-Biala pho/fax: +48 (33) 828 25 21 mob: +48 605 923 944 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A question about queues
I'm getting the same behavior, and can't seem to figure out where to set it to act differently. 1.06 is the version I'm using. I'm using AgentCallBack so my agents don't have to keep the line open -- perhaps that has something to do with it? I can't post my configs now (not at the office), but wanted to drop an email so the original poster doesn't think he's alone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS Head and SetLanguage
I upgraded to CVS-HEAD-04/20/05-09:25:13 yesterday and I am now having problems because Asterisk is not setting the language properly. My server runs in Spanish so I use the SetLanguage option so my prompts are read from the es directory inside the sounds directory. But now for some reason * is not finding my spanish prompts: -- Starting simple switch on 'Zap/2-1' -- Executing Wait(Zap/2-1, 2) in new stack -- Executing Answer(Zap/2-1, ) in new stack -- Executing ResponseTimeout(Zap/2-1, 5) in new stack -- Set Response Timeout to 5 -- Executing SetLanguage(Zap/2-1, es) in new stack -- Executing GotoIfTime(Zap/2-1, 10:00-18:00|mon-fri|*|*?abierto|s|1) in new stack -- Executing Wait(Zap/2-1, 2) in new stack -- Executing BackGround(Zap/2-1, cerrado) in new stack Apr 20 09:29:27 WARNING[26799]: file.c:489 ast_openstream_full: File cerrado does not exist in any format Apr 20 09:29:27 WARNING[26799]: file.c:793 ast_streamfile: Unable to open cerrado (format unknown): No such file or directory Apr 20 09:29:27 WARNING[26799]: pbx.c:5600 pbx_builtin_background: ast_streamfile failed on Zap/2-1 for cerrado -- Executing BackGround(Zap/2-1, bienvenida) in new stack -- Playing 'bienvenida' (language 'default') As you can see I set the language but when it tries to play the prompt it is not using the proper one. This was working until the last CVS I was using which was from the first week of April. Anyone know what may be happening? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RxFax not hanging up...
I have a line dedicated to receive faxes. It basically answers, gives
you a prompt to dial 1 for fax, an extension or wait on the line for a fax tone.
After a few seconds it will timeout (using the t extension) and give the
user a fax tone. The problem is that if the user hangs up RxFax will continue
trying to receive a fax forever and will not hang up the line until I kill the
channel manually.
I am using spandsp.0.0.2pre15 with Asterisk CVS-HEAD 04/20/05-09:25:13.
Is there a way to make rxfax hangup after a max time in case the fax does not
go through? I use the following fax macro:
[ Context 'macro-faxin' created by 'pbx_config' ]
's' =1. SetVar(FAXFILE=/var/spool/fax/${UNIQUEID}.tif)
[pbx_config]
2. rxfax(${FAXFILE}) [pbx_config]
3. system(/usr/local/bin/mailfax) [pbx_config]
4. Hangup() [pbx_config]
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
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[Asterisk-Users] signate.com webcall
Signate offers an interesting product they call 'webcall', which basically contacts a client at a number they provide then connects that person to a sales staff. Some potential for abuse but a nice idea for support etc. I know that it is possible to do (obviously) and well documented but has anyone actually released an open product similar to signate's webcall or even a basic web initiated call interface (ie for calling cards). I wasn't able to track via google or the wiki any ongoing projects - is anyone interested in working on something like this? J ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Issues of reliability, hardware, platforms
Chris Mason (Lists) wrote: I'm sure this has been debated before, I'd like to get peoples input. I see the hard drive as the single most likely point of failure on an * PBX. How reasonable would it be to run the OS and config files from a CF card, mount the /var/partition on a hard drive for the CDR recortds, logs and databases, and do some kind of test on bootup that would turn off those features if the hard drive was unavailable (failed)? I am messing around with a firewall that mounts from a cf card, it got me thinking about asterisk from a cf card, as there would be little chance of failure. This all assumes that CF cards are more reliable than hard drives (and power supplies). Other than that, what have you done to ensure reliability. I am planning a RAID1 hard drive install for my serious pbx customers, in my experience that makes for a very reliable machine. The best solution is a cluster setup, with multiple machines, and no single point of failure. -SteveK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Phone Compatability
I currently use an SPA-841 on my desk and don't have any problems with it http://www.geekgazette.com/index.php?option=com_contenttask=viewid=24 I have been looking at these phones and they have more office features http://www.zultystechnologies.com/index.jsp?tab=product_listtype=phones -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Wednesday, April 20, 2005 3:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP Phone Compatability The Sipura SPA-841 has everything except memory buttons but has a directory and speeddials so I don't think that's so important. Cheap and well made, although if the speaker phone is very important, get Polycoms, it's the business they are best in. Chris Mason www.anguillaguide.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Wednesday, April 20, 2005 12:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP Phone Compatability Every once in a while I read messages about people having problems with certain models of SIP phones, some of them being well known models. I'm interested in purchasing new SIP phones for my office and wanted to know which brand/model is most stable with Asterisk, which allows most office features. These features should include: multiple-line appearances (at least 3), call conference, blind and non-blind transfer, memory buttons or speed dials, voice message light indicator, speaker phone, mute, redial, caller-id display. Anything on top of these features is a plus but not really a requirement. Any suggestions? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: CVS-HEAD and CheckGroup/SetGroup
Do the SetGroup and CheckGroup functions behavior differently in
CVS-HEAD
vs CVS v1-0?
When I upgrade to CVS-HEAD my call waiting disable doesn't seem to
work,
using:
exten = s,1,SetGroup(SIP${ARG1})
exten = s,2,CheckGroup(1)
exten = s,3,Dial(Sip/${ARG1},15,t)
Do you not need a
exten = s,103,Congestion()
otherwise the checkgroup has nowhere to go ?
Is this to disable the call waiting on Polycom phones? If so, I don't
think there wouldn't be a need to have it on the s extension. You'd
just need to have it on any extension that the Polycom phone would call
(other handsets, outside lines, voicemailmain, etc). CheckGroup does
increment n+101, but unless you want them to get a busy signal, I'd
probably move them on to the next line appearance on the phone, or to a
voicemail box.
- Noah
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Re: [Asterisk-Users] Wait in Dial String
On Wednesday 20 April 2005 10:29 am, David Choo wrote:
Dear All,
My boss has placed a requirement for me to forward all our IDD calls
through a partner's IDD service, which requires us to call a 8 digit
number, wait for 1 sec, before we send in the foreign number we're trying
to call.
As I can't find anything on getting the PBX to wait, i'm only removing the
1st 3 digits (900) and sending in an extra 1 to simulate the wait. It
works, but not all the time. Is there anyway that I can place a wait
command here? I'm tried placing w / p but both don't works. Would like to
seek your kind assistance!
exten = _9001.,1,Dial(Zap/g1/64919669,,D(${EXTEN:3}),)
exten = _9001.,n,Hangup()
Try 'w',
E.g. for my old bridge to BizFon, I had to dial 9, wait, then the number:
exten = _NX,1,Dial(Zap/g1/9w${EXTEN})
Just put the 'w' between the numbers that you want it to 'wait' at.
-josiah
--
Josiah Bryan
IT Coordinator
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224
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[Asterisk-Users] Cisco 7960 SIP registration???
So, here's my quandary: 1) Asterisk running CVS HEAD as of a couple days ago 2) Cisco 7960 SIP phones in a different subnet than the Asterisk server 3) NAT/Firewall device between 7960's and * I can initiate a call from the 7960's just fine. They can call out using our Broadvoice account and access any of the vmail stuff on *. When calling in from the outside world and dialing one of their extensions, however, I always get a this user is on the phone message. The console spits out this nugget: == CDR updated on SIP/4252780761-933d -- Executing Macro(SIP/4252780761-933d, stdsip|tycisco|101) in new stack -- Executing Dial(SIP/4252780761-933d, SIP/tycisco) in new stack Apr 20 08:14:59 NOTICE[32728]: app_dial.c:973 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time (1:0/1/0) A showing of the sip peers: sip show peers Name/username HostDyn Nat ACL Mask Port Status rickcisco/cisco2 (Unspecified)D N 255.255.255.255 0UNKNOWN tycisco/cisco1 (Unspecified)D N 255.255.255.255 0UNKNOWN sip.broadvoice.com/425278 147.135.4.128 255.255.255.255 5060 OK (127 ms) 3 sip peers [1 online , 2 offline] I'm sure the reason I can't call to an extension is that they are appearing offline. How can I remedy this, however? I'm an * newbie, so go easy on me. :^) Thanks, Ty Christensen MCP, MCSP, MCSB Master Mind Productions Inc. www.mastermindpro.com http://www.mastermindpro.com/ (425) 378-7724 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MF instead of DTMF
Yes, SIT messages and CLASS messages like... Your selective call rejection service is now off Your calls are being forwarded to XXX-XXX- etc. -- Mike - Original Message - From: jltaylor [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 19, 2005 3:49 PM Subject: RE: [Asterisk-Users] MF instead of DTMF Are you talking about SIT and all of the announcements that are for: Work stoppage no dial tone switch blockage emergency announcments misdialing vacant disconnects etc? James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael B. Murdock Sent: Tuesday, April 19, 2005 11:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MF instead of DTMF Thank for the reply Jim, I realize FGD uses MF and obviously there is a MF decoder in asterisk. What I am trying to determine is if asterisk can detect MF digits after the call has been presented using FGD call setup. What I am trying to determine is if Asterisk can be used as a replacement for the Class Announcement periphial for a Class-5 (DMS-10) switch. I am trying to get the specific Nortel or Telcordia spec on this feature but have been told by one switch tech that the specific announcement (or string of announcements) to play is indicated by a variable number of outpulsed MF digits after the trunk is seized. -- Mike - Original Message - From: jltaylor [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 19, 2005 9:56 AM Subject: RE: [Asterisk-Users] MF instead of DTMF MF works with FGD FGC signaling. Are you taking FGD with a tandem connection? James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael B. Murdock Sent: Tuesday, April 19, 2005 8:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] MF instead of DTMF I am looking into using Asterisk for an application where the upsteam switch will provide MF digits instead of DTMF after establishing a call. This is not during the call set up but after the call is established additional MF digits will be passed to indicated features to provide to the caller. Trunking will be EM T1. Does asterisk support MF detection in addition to DTMF? Has anyone done anything like this before? -- Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer of incoming call from external to internal number
When I place a call on my softphone to a external number the call is placed, when I click transfer, dial internal extrention (e.g. 202) then hit transfer again, the call is transfered to the 202 extention fine. However, when the other way Internal call comes in, extension 201 answers, and transfer to extension 202 the call is ended at 201 but 202 does not ring, the caller on the external line just hears nothing and the call is lost internally. FreeBSD, Asterisk X100P FXO PCI Card 4 Asterisk Linux IP PBX Xten SoftPhone zapata.conf looks like this: ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=no hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=yes faxdetect=incoming channel=1 Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Which free calling card app most suitedforcommercial use?
I think the word crap is a pretty strong word and is not fare to the authors. Everyone have their own requirements of how billing should or should not work. Everyone is exposed to a different way a pre-paid calling card platform should behave. I have been in pre-paid environment for almost 15 years and seen/implemented some interesting business models. All of them depend on the provider and how the product is brought to market. Point is, there will never be a unified billing system that will satisfy every requirement of every pre-paid carrier in the world. I think these guys did a good job showing the community how pre-paid billing should be implemented and interfaced with asterisk and therefore deserve a credit for that. If you don't like the way they implemented things, then contribute extensions or patches. If you don't like the architecture and don't think a particular approach can be extended, then contribute your own work and show everyone that it is better. Until you do, avoid the words like crap when referring to other people efforts. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Wednesday, April 20, 2005 9:46 AM To: snacktime; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Which free calling card app most suitedforcommercial use? My opinion is that both are Crap. Both of them have a flaw in their base design, which is difficult to explain in a post like this. Suffice to say that these two applications neither support nor designed for mutilpe routes ( multiple Area codes with Destination groups) nor multiple rate plans(Provider rates or buying rates and selling rates) nor multiple business models(retail, wholesale, corporate customers) Hence both of them cannot be the base for a commercial grade billing system for a Calling card Model. These apps canot be used for a realtime call control using CPD (Call Progress Detection) and Prepaid amounts for a post-paid Billing and call disconnect. Without this very essential feature for a commercial Calling card billing application, you would be better off calculating the calls from the Master.csv file for a post paid bill management. AreskiCC is a little more thought-driven and hence can be improved upon. If anyone is interested in developing a full fledged billing system, I have created a deisgn document ( a very elaborate rough draft infact) which I can share with you. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of snacktime Sent: Tuesday, April 19, 2005 5:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Which free calling card app most suited forcommercial use? I'm working on an * billing system, and instead of reinventing the wheel I would prefer to use an existing codebase for the calling card portion. The two that look most promising are astcc and the * prepaid billing application that uses postgresql. Any comments? Chris NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Want to use Asterisk instead of existingMeridianNorstar system ... need some help
On Apr 19, 2005, at 9:12 AM, Mike Robinson wrote: Yes, you CAN use your existing Meridian phones. There is a product called a Handset Gateway that converts traditional digital PBX telephones (Norstar, Meridian, Definity, NEC, etc) into SIP signaling so the existing phones and wiring can work with *. It's a heck of a lot cheaper than buying new IP phones (plus the LAN upgrade to power them), and you still get all the features of the digital phones (speakerphone, message waiting indicator, feature buttons, etc). Also, it makes the conversion a lot simpler because users don't have to learn how o use a new phone and you don't have to go around to all the desks and swap the phones. Handset Gateways are available from Mitel, 3Com, Lucent (Lucent, not Avaya), and Citel Technologies. I think only the Lucent and Citel ones work with * though. Check it out and save yourself a bunch-o-headache. I'm looking for a Gateway that works with ESI 16-key digital handsets... anyone out there heard of one? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Re: a simple question
Thanks a lot , Make update is ok, But where i can check the version of my Asterisk ? Obviously it is another simple one . :( Date: Fri, 15 Apr 2005 22:01:28 -0400 From: Steve Totaro [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] a simple question . To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=gb2312 a simple question .save this http://www.szmidt.org/asterisk/asterisk-update.sh to /usr/src and run it. you can also go into /usr/src/asterisk and type make update - Original Message - From: Weiming Jiang To: asterisk-users@lists.digium.com Sent: Friday, April 15, 2005 9:04 PM Subject: [Asterisk-Users] a simple question . Hi, I am a new asterisker ,: ( need your help . a simple question : How should i do to upgrade my asterisk to new version ? or who can share some documents or info about the upgrade ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UIP200
I have a Uniden UIP200 behind a NAT and an * server behind another NAT. I am able to register with * and place calls. However, once the call is established, I cannot hear anything from either end (UIP200 as well as the called destination). Then, I did the exact same thing with X-Lite and everything worked perfectly. Does anyone have any idea what settings I need to use on the UIP200 or why would the media be blocked? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitor via Manager question
Hello. I checked in the wiki and read a bunch of old threads from this mailing list but haven't found what I'm looking for. I'm using a simple PHP script, and here is the relevant portion: fputs($socket, Action: Monitor\r\n); fputs($socket, Channel: Zap/1-1\r\n\r\n); That works fine. As does this: fputs($socket, Action: Monitor\r\n); fputs($socket, Channel: SIP/8000-h4d8\r\n\r\n); But what I need to be able to do is this: fputs($socket, Action: Monitor\r\n); fputs($socket, Channel: SIP/8000\r\n\r\n); And have it record either the first call that is up on SIP/8000, or the last, or whatever (doesn't matter, only one call at a time will be up on this line). However, if I try this, it always comes back to me with: Response: Error Message: No such channel Because I am not specifying the actual call that is up. Is there any way to do this? Or can I somehow easily look up what Zap channel is used by SIP/8000 and pass that? The other twist is that SIP/8000 will be specified by a variable passed through a form. Basically, I want a web form with two buttons and a text box: Start Rec., Stop Rec., and User Ext.. I didn't start out that complex though, just right now it's a simple PHP script, and it was taken from the Wiki. I need to get the core functionality working properly before I add the buttons and whatever. Thanks in advance for any advice. -- Dana ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT and only been able to have 1 SIP phone behind
Thats exactly what I thought but there are many parts on the wiki where they mention the more than 1 SIP client behind NAT mmyth. Oh well, maybe an urban legend? :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Miércoles, 20 de Abril de 2005 06:04 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] NAT and only been able to have 1 SIP phone behind Anton Krall wrote: Guys. Ive read on the wiki that a common problem with nat is that you can only have 1 sip phone behind, how do you get around this issue? Having a sip enabled router behind the nat like the GS 488 489 or 486? Or how have you done it without having any kind of linux box (SER or *) behind the nat. The idea is to have 2 or more sip clients behind some NAT and been able to connect to a remote asterisk box. I don't know why people think this. Any router with PAT (Port Address Translation) should work with multiple SIP clients behind NAT. Most routers support PAT (but may not call it that). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wait in Dial String
Try one w:
exten = _9001.,1,Dial(Zap/g1/64919669,,D(w${EXTEN:3}),) ***
exten = _9001.,n,Hangup()
Robert Andrew Keller
Ferndale School District #502
[EMAIL PROTECTED]
360-383-9228 PH.
360-383-9218 FAX
Paving the way for tomorrows genius.
From: David Choo [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Wed, 20 Apr 2005 22:29:27 +0800
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: James Lim [EMAIL PROTECTED]
Subject: [Asterisk-Users] Wait in Dial String
Dear All,
My boss has placed a requirement for me to forward all our IDD calls
through a partner's IDD service, which requires us to call a 8 digit
number, wait for 1 sec, before we send in the foreign number we're trying
to call.
As I can't find anything on getting the PBX to wait, i'm only removing the
1st 3 digits (900) and sending in an extra 1 to simulate the wait. It
works, but not all the time. Is there anyway that I can place a wait
command here? I'm tried placing w / p but both don't works. Would like to
seek your kind assistance!
exten = _9001.,1,Dial(Zap/g1/64919669,,D(${EXTEN:3}),)
exten = _9001.,n,Hangup()
Best Regards,
==
David Choo
Systems Engineer
Business Technology Division
Engineered for Changing Businesses
Espore Corp Pte Ltd
68 Kallang Pudding Rd
#04-03 SYH Logistics Bldg
Singapore 349327
Tel: 65-68487806
Fax : 65-6842 2724
E-mail :[EMAIL PROTECTED]
=
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Re: [Asterisk-Users] Asterisk and T.38.
Hi Jairo, Try with other values for the jitter in your Gateway (H323). One customer have a scenario like this: Phone/Fax Gateways H323 -with 16/8/2/1 Port FXS- --- GNUGK --- Asterisk --- Zap (E1) .. and only we need modified the jitter settings in two Gateways. Rafael Gonzalez Lomeña El mar, 19-04-2005 a las 12:09 +0200, Jairo Buendia escribió: Hi! I want to send fax to PSTN with Asterisk, but by now I can't. I am using the following boxs: Internet Zap E1 Phone/Fax Gateway(H323)--- Asterisk--- PSTN The gateway H323 has T38 and T30. Before I began with Asterisk, I used Cisco to connect with PSTN, and the Fax worked very well in T38 (T30 didn't work, I think the reason is the jitter). I have read that Asterisk doesn't support T.38, is this correct?, is there any comercial implementation in Asterisk?. If Asterisk doesn't support T.38, how can I use it to send fax to PSTN? Thanks in advance. __ Renovamos el Correo Yahoo!: ¡250 MB GRATIS! Nuevos servicios, más seguridad http://correo.yahoo.es ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rafael Angel Gonzalez Lomeña Ingeniero I+D / RD Ingenieer [EMAIL PROTECTED] Avanzada7, S.L. Parque Tecnologico de Andalucía Edf. Centro de Empresas Avd. Juan Lopez Peñálver,17 3ª 29590 Campanillas Málaga (España/Spain) Telf. 911830149 Ext. 703 951014943 Ext. 703 951014947 Fax.951010922 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Which free calling card app most suitedforcommercial use?
I have actually setup AstCC and got it working. I have found a couple problems with it and I dont think the problems have anything to do with my setup. The problems that I am seeing are: 1) Out of the box, the CDRs dont work. I have a quick document that explains why and how to fix it. If you would like this, let me know and I can send it to you. 2) At the one minute mark (one minute left on the card) AstCC will play a sound telling you this. After this, it seems like the RTP stream breaks as neither side can hear the other. I have not done any packet sniffing to confirm this but it seems like that is what the problem is. If it is not the RTP stream it is something that would act like it. The CDR problem is a minor thing in perspective to the RTP problem. As far as call cost and routing, you are able to set up multiple routes and different call costs based on a REGEX. Example: ^314.* Would set any call starting with 314 to which ever cost. You also have the ability to select which trunk the call will go out on. Just some things that I have found. If you want the CDR fix, let me know. Dave Kettmann NetLogic 314-266-4000 -Original Message- From: Kanuri, Seshu (Company IT) [mailto:[EMAIL PROTECTED] Sent: Wednesday, April 20, 2005 8:46 AM To: snacktime; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Which free calling card app most suitedforcommercial use? My opinion is that both are Crap. Both of them have a flaw in their base design, which is difficult to explain in a post like this. Suffice to say that these two applications neither support nor designed for mutilpe routes ( multiple Area codes with Destination groups) nor multiple rate plans(Provider rates or buying rates and selling rates) nor multiple business models(retail, wholesale, corporate customers) Hence both of them cannot be the base for a commercial grade billing system for a Calling card Model. These apps canot be used for a realtime call control using CPD (Call Progress Detection) and Prepaid amounts for a post-paid Billing and call disconnect. Without this very essential feature for a commercial Calling card billing application, you would be better off calculating the calls from the Master.csv file for a post paid bill management. AreskiCC is a little more thought-driven and hence can be improved upon. If anyone is interested in developing a full fledged billing system, I have created a deisgn document ( a very elaborate rough draft infact) which I can share with you. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of snacktime Sent: Tuesday, April 19, 2005 5:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Which free calling card app most suited forcommercial use? I'm working on an * billing system, and instead of reinventing the wheel I would prefer to use an existing codebase for the calling card portion. The two that look most promising are astcc and the * prepaid billing application that uses postgresql. Any comments? Chris NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wait in Dial String
On Wed, 20 Apr 2005 10:24:37 -0500
Josiah Bryan [EMAIL PROTECTED] wrote:
On Wednesday 20 April 2005 10:29 am, David Choo wrote:
Dear All,
My boss has placed a requirement for me to forward all
our IDD calls
through a partner's IDD service, which requires us to
call a 8 digit
number, wait for 1 sec, before we send in the foreign
number we're trying
to call.
As I can't find anything on getting the PBX to wait, i'm
only removing the
1st 3 digits (900) and sending in an extra 1 to simulate
the wait. It
works, but not all the time. Is there anyway that I can
place a wait
command here? I'm tried placing w / p but both don't
works. Would like to
seek your kind assistance!
exten = _9001.,1,Dial(Zap/g1/64919669,,D(${EXTEN:3}),)
exten = _9001.,n,Hangup()
Try 'w',
E.g. for my old bridge to BizFon, I had to dial 9, wait,
then the number:
exten = _NX,1,Dial(Zap/g1/9w${EXTEN})
Just put the 'w' between the numbers that you want it to
'wait' at.
-josiah
And as an added tidbit... If I remeber correctly, each w
is about a 1/2 second. So to get a second pause you would
need ww in the string.
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