Re: [asterisk-users] Audiocodes MP-114 noise
Jessee, Thank you for your help. I downloaded firmware and sample configuration files. But the firmware was old version for MP118 and MP124. Where can i download recent one? Can i upload only ini file to change countrycoefficient ? Regards, Jason. --- Jessee J Holmes [EMAIL PROTECTED] wrote: Jason, First, before you start reading, get to the latest firmware from Audiocodes (MP118_SIP_F4.80A.034.004.cmp), there have been significant echo improvements in this version. After many days of working with Audiocodes on this problem and much time spent here by multiple technicians trying to reproduce and resolve this issue; this morning, Atacomm received an email from Audiocodes with a full explanation to this now confirmed issue with all MP-11x units. Atacomm will immediately begin work on a KB article within our website that confirms this issue and outlines the manufacturer recommended steps to resolve this problem. Apparently, there have been some changes with the MP-11x's that can negatively affect line noise and echo. Below are some steps which can help to correct these problems: 1. The new design did away with the Coefficent file. Audiocodes, now instead, introduced a configurable parameter called countrycoefficient. This parameter can be adjusted to a specific country based on known configurations. For the most part this should work. 70(USA) is the default value. More can be found in the Users manual. 2. In just about every case, an FXO is added to a Pre-existing PBX or CO line, you can expect echo. This comes from the fact that delay (IP Network) is being introduced, and what used to be Side tone is now delayed so much it is echo. Just about every difference on the line that can be heard between the pre fxo and post fxo installation can be traced to echo, or line quality issues. 3. Going forward, Audiocodes would like to suggest that when installing the product do the following: A) Make sure the Line coming from the PBX or CO is a Loop Start line. Ground start is not supported on the MP-11x series of gateways. (The M1K FXO will in 5.0) B) Check that the Line can deliver for a 600 Ohm Impedance line -52 to -24 V of Off Hook Voltage -15 to -6 V of On Hook Voltage 20 to 35 ma of loop current. If you know the line is not 600 Ohm, please gather metrics on the line, and the make and model of the PBX or switch it is attached too, plus country of origin. If it is not from the USA, please look up the country of origin and then find the CountryCoefficient to match this. Load the .ini file to the board with this setting and reset. Make sure the Gateway has a firmware version of 4.60.035 or higher or 4.80.030 or higher. C) Put the device on the network with Voice Volume set to 0 and input gain set to 0. Make calls, if there is no issue, you can stop here. However, Echo is still expected most of the time. D) The echo should be heard by the IP side participant as their voice is reflected back. If this is the case, then what needs to be done is to lower the voicevolume (IPTEL). This way the speakers reflected voice will comeback low enough for the ECAN to cancel it out (-6 is usually recommended as the value to plug in here). A little experimentation is needed as the loss for all lines will vary based on length from the CO. Echo is usually taken care of in this manner. E) The incoming speaker from the PSTNs voice seems low, set InputGainLocation =1, and then slowly increment the Input Gain Parameter(Tel?P) to adjust for this. In past releases (see the note about loads above), the input gain was always applied prior to the ECAN which had the effect of amplifying the returned echo and noise on the line causing crosstalk and clipping issues. This is no longer the case. If the above does not resolve the issues, then you need to go ahead and collect DSP, Ethereal and Syslog traces along with the board.ini, these are to be sent to your support agent, who will then send these to Audiocodes for their engineers to evaluate. This should not happen often. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http:// voipstore.atacomm.com/ On Nov 3, 2006, at 12:14 AM, Jason Kim wrote: Jessee, I tried many combinations of Voice Volume, Input Gain and packetization time , but it's noisy steel. I'm using G.711A-law and packetization time is 20ms. It can be impedance mismatch problem but i cannot adjust impedance of FXO port of MP-114. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Operating queues with clients on a legacy PABX
Hi guys! I'm having one or two issues with queues hosted by an Asterisk machine where the clients are on a legacy PABX - at least for the interim. I fully expect most of these issues to be non-resolvable, but thought I'd at least ask to find out if there is some way of working around the issues. The legacy PABX is an NEC 7400 ICS connected to Asterisk via an E1 ISDN link. Calls are passed to the NEC without a problem. The biggest issue is when an agent transfers a call to another person on the NEC. Obviously using the transfer button on the phones gives Asterisk no clue that the call has been transferred, meaning that the agent then does not receive another call until the transferred call has been completed. Can anyone think of a workaround for this? signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] flash transfer problem in asterisk integration with old PBX
I've tried to transfer a call using the Flash command, but with my configuration it doesn't work. I have a traditional PBX connected with a zap channel to Asterisk that acts like an IVR: TELCO line -- traditional PBX (FXS) -- (FXO) Asterisk From the TELCO line I can make a call to the traditional PBX and reach Asterisk, the IVR system on Asterisk answers the call and I can dial an extension (for example 42 that is on the traditional PBX). In the asterisk dialplan I've set to transfer the call using Flash() like in this example: exten = 42,1,Flash() exten = 42,2,Background(silence/1) wait 1 second for the traditional PBX exten = 42,3,SendDTMF(42,250) exten = 42,4,Background(silence/1) wait 1 second for the traditional PBX exten = 42,5,Hangup() When I dial the extension 42, the phone 42 on the traditional PBX rings but when I answer there isn't communication with the call from the TELCO line and after a few seconds the line hangup. Here you can see what happen in asterisk CLI console: Executing Answer(Zap/4-1, ) in new stack -- Executing BackGround(Zap/4-1, a_suoni_plink/menu_esterno2) in new stack -- Playing 'a_suoni_plink/menu_esterno2' (language 'it') == CDR updated on Zap/4-1 -- Executing Flash(Zap/4-1, ) in new stack -- Flashed channel Zap/4-1 -- Executing BackGround(Zap/4-1, silence/1) in new stack -- Playing 'silence/1' (language 'it') -- Executing SendDTMF(Zap/4-1, 42) in new stack -- Executing BackGround(Zap/4-1, silence/1) in new stack -- Playing 'silence/1' (language 'it') -- Executing Hangup(Zap/4-1, ) in new stack == Spawn extension (incoming, 42, 5) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' I've tried the following changes to the dialplan in my example but transfer still doesn't work: - I've tried to use wait(1) instead of Background(silence/1) - I've tried without Background(silence/1) or wait(1): exten = 42,1,Flash() exten = 42,2,SendDTMF(42,250) exten = 42,3,Hangup() - I've tried without the Hangup() instructions at the end Has anyone the same problem like me and any suggestions? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue forks asterisk and then leaves the extra processes lying around
Hi,
I have a problem with Queue where by a call comes in to the queue and
if all the phones are busy and the queue reaches the timeout, it will
fork a process and leave it sitting there before going off to the next
step in the dial plan and continuing normally. This doesn't cause any
problems except for I assume that it will eventually use up all the
memory on the machine and it messes with my process monitoring.
It doesn't seem to matter what I have as the next step after the Queue
command and it happens only sometimes. It seems like it might even be
a timing issue given that it's less likely to happen if any one of the
phones ring.
The new asterisk processes that get started up look like they think
they're new asterisk instances or though they don't actually do
anything or interfere with the first asterisk instance.
Has anyone had any problems like this? Am I doing something wrong?
The appropriate part of my dial plan looks like this:
exten = 101,1,Answer
exten = 101,n,GotoIf($[${CONTEXT}=from-internal]?USERCID:SETCID)
exten = 101,n(USERCID),Macro(user-callerid,)
exten = 101,n(SETCID),Set(CALLERID(name)=${CALLERIDNAME})
exten =
101,n,Set(MONITOR_FILENAME=/var/spool/asterisk/monitor/q${EXTEN}-${TIMESTAMP}-${UNIQUEID})
exten = 101,n,Queue(101|tr|||30)
exten = 101,n,Goto(ext-local,83,1)
exten = 101*,1,Macro(agent-add,101,)
exten = 101**,1,Macro(agent-del,101,101)
and from queues.conf
[101]
wrapuptime=0
timeout=15
strategy=ringall
retry=5
queue-youarenext=
queue-thereare=
queue-thankyou=queue-thankyou
queue-callswaiting=
music=default
monitor-join=yes
monitor-format=
member=Local/[EMAIL PROTECTED],0
member=Local/[EMAIL PROTECTED],0
maxlen=2
leavewhenempty=no
joinempty=Yes
context=
announce-holdtime=no
announce-frequency=0
and some logs to show what I mean by the new asterisk process thinking
that it is actually a new asterisk.
-- snip --
Nov 8 21:44:38 DEBUG[25896] channel.c: Hanging up channel 'Local/[EMAIL
PROTECTED],2'
Nov 8 21:44:38 DEBUG[25627] devicestate.c: Changing state for Local/[EMAIL
PROTECTED] - state 0 (Unknown)
Nov 8 21:44:38 DEBUG[25904] app_queue.c: Device 'Local/[EMAIL PROTECTED]'
changed to state '0' (Unknown)
Nov 8 21:44:38 DEBUG[25897] cdr_addon_mysql.c: cdr_mysql: inserting a CDR
record.
Nov 8 21:44:38 DEBUG[25897] cdr_addon_mysql.c: cdr_mysql: SQL command as
follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2006-11-08 21:44:38','49761450','49761450','83','from-internal',
'Local/[EMAIL
PROTECTED],2','','AGI','recordingcheck|20061108-214438|1162975478.49',0,0,'NO
ANSWER',3,'','1162975478.49')
Nov 8 21:44:38 DEBUG[25897] channel.c: Hanging up channel 'Local/[EMAIL
PROTECTED],2'
Nov 8 21:44:38 DEBUG[25627] devicestate.c: Changing state for Local/[EMAIL
PROTECTED] - state 0 (Unknown)
Nov 8 21:44:38 DEBUG[25905] app_queue.c: Device 'Local/[EMAIL PROTECTED]'
changed to state '0' (Unknown)
Nov 8 21:44:38 VERBOSE[25902] logger.c: == Parsing
'/etc/asterisk/extconfig.conf': Nov 8 21:44:38 DEBUG[25902] config.c:Parsing
/etc/asterisk/extconfig.conf
Nov 8 21:44:38 VERBOSE[25902] logger.c: == Parsing
'/etc/asterisk/extconfig.conf': Found
Nov 8 21:44:38 VERBOSE[25902] logger.c: == Parsing
'/etc/asterisk/manager.conf': Nov 8 21:44:38 DEBUG[25902] config.c: Parsing
/etc/asterisk/manager.conf
... lots of asterisk start up logs ...
-- snip --
Regards,
Nigel
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RE: [asterisk-users] channel.c: Unable to request channel ZAP
Zap show channels shows only 2 or 3 channels to be in use ... others are not. Resetinterval is set to 1200 (20 minutes). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Forrest Beck Sent: Thursday, November 02, 2006 4:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] channel.c: Unable to request channel ZAP What does zap show channels show? Are all the channels shown as in use? What is set in zapata.conf for resetinterval= ? If anything. I believe that resetinterval is used to reset unused channels for any channels that are left open. On 10/31/06, Asterisk [EMAIL PROTECTED] wrote: Hi All, I have one rather annoying problem...my PBX can work great for weeks, when suddenly I start receiving these messages when I try to make a zaptel call: Oct 31 13:52:47 NOTICE[15636] app_dial.c: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) Oct 31 13:52:49 NOTICE[15648] channel.c: Unable to request channel ZAP/g1/247 I'm using Sangoma A104 card (with four E1 spans), and these problems are only occurring on the first two spans (which are connected to a legacy PBX) - the second two spans, which are connected to the Telco, work perfectly. Even more: when these messages start to occur, I can hardly initiate any call via problematic two spans (1st and 2nd), where I can with no problem initiate a new call thru the unproblematic two spans (3rd and 4th). Restart of the Asterisk is the only cure so far... Does anyone know what could possibly be the cause, or how could I troubleshot this problem? Regards. Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Queue forks asterisk and then leaves the extraprocesses lying around
Are you using freePBX by any chance?
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nigel
Roberts
Sent: 08 November 2006 08:55
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Queue forks asterisk and then leaves the
extraprocesses lying around
Hi,
I have a problem with Queue where by a call comes in to the queue and if
all the phones are busy and the queue reaches the timeout, it will fork
a process and leave it sitting there before going off to the next step
in the dial plan and continuing normally. This doesn't cause any
problems except for I assume that it will eventually use up all the
memory on the machine and it messes with my process monitoring.
It doesn't seem to matter what I have as the next step after the Queue
command and it happens only sometimes. It seems like it might even be a
timing issue given that it's less likely to happen if any one of the
phones ring.
The new asterisk processes that get started up look like they think
they're new asterisk instances or though they don't actually do anything
or interfere with the first asterisk instance.
Has anyone had any problems like this? Am I doing something wrong?
The appropriate part of my dial plan looks like this:
exten = 101,1,Answer
exten = 101,n,GotoIf($[${CONTEXT}=from-internal]?USERCID:SETCID)
exten = 101,n(USERCID),Macro(user-callerid,)
exten = 101,n(SETCID),Set(CALLERID(name)=${CALLERIDNAME})
exten =
101,n,Set(MONITOR_FILENAME=/var/spool/asterisk/monitor/q${EXTEN}-${TIMES
TAMP}-${UNIQUEID})
exten = 101,n,Queue(101|tr|||30)
exten = 101,n,Goto(ext-local,83,1)
exten = 101*,1,Macro(agent-add,101,)
exten = 101**,1,Macro(agent-del,101,101)
and from queues.conf
[101]
wrapuptime=0
timeout=15
strategy=ringall
retry=5
queue-youarenext=
queue-thereare=
queue-thankyou=queue-thankyou
queue-callswaiting=
music=default
monitor-join=yes
monitor-format=
member=Local/[EMAIL PROTECTED],0
member=Local/[EMAIL PROTECTED],0
maxlen=2
leavewhenempty=no
joinempty=Yes
context=
announce-holdtime=no
announce-frequency=0
and some logs to show what I mean by the new asterisk process thinking
that it is actually a new asterisk.
-- snip --
Nov 8 21:44:38 DEBUG[25896] channel.c: Hanging up channel
'Local/[EMAIL PROTECTED],2'
Nov 8 21:44:38 DEBUG[25627] devicestate.c: Changing state for
Local/[EMAIL PROTECTED] - state 0 (Unknown) Nov 8 21:44:38 DEBUG[25904]
app_queue.c: Device 'Local/[EMAIL PROTECTED]' changed to state '0'
(Unknown) Nov 8 21:44:38 DEBUG[25897] cdr_addon_mysql.c: cdr_mysql:
inserting a CDR record.
Nov 8 21:44:38 DEBUG[25897] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,dura
tion,billsec,disposition,amaflags,accountcode,uniqueid) VALUES
('2006-11-08 21:44:38','49761450','49761450','83','from-internal',
'Local/[EMAIL PROTECTED],2','','AGI','recordingcheck|20061108-214438
|1162975478.49',0,0,'NO ANSWER',3,'','1162975478.49') Nov 8 21:44:38
DEBUG[25897] channel.c: Hanging up channel
'Local/[EMAIL PROTECTED],2'
Nov 8 21:44:38 DEBUG[25627] devicestate.c: Changing state for
Local/[EMAIL PROTECTED] - state 0 (Unknown) Nov 8 21:44:38 DEBUG[25905]
app_queue.c: Device 'Local/[EMAIL PROTECTED]' changed to state '0'
(Unknown)
Nov 8 21:44:38 VERBOSE[25902] logger.c: == Parsing
'/etc/asterisk/extconfig.conf': Nov 8 21:44:38 DEBUG[25902]
config.c:Parsing /etc/asterisk/extconfig.conf
Nov 8 21:44:38 VERBOSE[25902] logger.c: == Parsing
'/etc/asterisk/extconfig.conf': Found
Nov 8 21:44:38 VERBOSE[25902] logger.c: == Parsing
'/etc/asterisk/manager.conf': Nov 8 21:44:38 DEBUG[25902] config.c:
Parsing /etc/asterisk/manager.conf
... lots of asterisk start up logs ...
-- snip --
Regards,
Nigel
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Re: [asterisk-users] Queue forks asterisk and then leaves the extraprocesses lying around
Hi Lee, On Wed, 08 Nov 2006 at 09:00:27 -, Lee Archer wrote: Are you using freePBX by any chance? Yes, version 2.1.1. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Queue forks asterisk and then leaves theextraprocesses lying around
Hi, have a look at http://www.freepbx.org/trac/ticket/1174 it's currently in the bug list. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nigel Roberts Sent: 08 November 2006 09:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue forks asterisk and then leaves theextraprocesses lying around Hi Lee, On Wed, 08 Nov 2006 at 09:00:27 -, Lee Archer wrote: Are you using freePBX by any chance? Yes, version 2.1.1. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Agents that handle calls from multiple queues
Title: Message Hi everybody, I've got an Asterisk configuration where an agent handles calls from multiple queues. At the moment I'm using the default Queue application and I encountered the following problem: When there are calls waiting in multiple queues the selection of which call is handled by the Agent is more or less random. It would be nice if the call that was waiting the longest was handled first. I've been looking at ICD as an alternative to the Queue application but as far as I could see this project hasn't been updated for quite some time now. Does anybody know of an alternative or a way to get the desired behaviour? Thanks, Ardjan Zwartjes. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Glitches in sound every time that Asteriskreceives reINVITEs
My Asterisk server is working fine, although every time that in the middle of any call there is a reinvite, the user hears a glitch. Why is this happening? How can I solve this problem? That's because a REINVITE is generally used to change from one codec to another. For some reason this involves stopping the existing audio, waiting a little while and then starting a new audio stream. So far this one of the reasons why I don't like reinvite... -- Andreas SikkemaBBeyond Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie questions about Voice mail
Brian, I should concur with all that Dean raised. Given the experience level you describe and the clear business case for what you want to do, had you considered a commerical solution ? It would give you the peace of mind that all will work. It will also allow you to do many of the smaller features such as Outlook Integration in a click and drop manner as well as the group issues, setting up of voicemail delivery to email etc. See some other comments below. Steve (of course would be more than happy to promote our own but there are others you could do well to look at) - Original Message - From: [EMAIL PROTECTED] To: Dean Collins [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, November 05, 2006 9:01 PM Subject: RE: [asterisk-users] Newbie questions about Voice mail Dean Thanks for responding. I have added more info in your reply. Right now we do not operate our own PBX or voice mail system. All of the service is provided by the telco. As a start I was wondering if I could simply put in asterisk to do just voicemail. I am assuming the telco can configure all the phone to automatically call forward to asterisk on no answer. If asterisk can handle this I am assuming that a user would just call some number to retiev voice mail. They would lose the call waiting light on their phone so the email notification of a voice mail would be necessary. ..Brian On Sun, 5 Nov 2006, Dean Collins wrote: Date: Sun, 5 Nov 2006 00:04:36 -0500 From: Dean Collins [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Newbie questions about Voice mail Hi Brian, I'm sure some other people will give you better answers but quick answers are; 1/ Depends on volume of message leaving/collection, is it in a single location? Multiple locations with multiple time zones? Two locations, one time zone. Could be two different systems since they are in two different cites connected by a 1G connection. Estimate the number of voicemails left per hour and reply with this. There are about 3000 phones. Some are busier than others os lets say 2 messages per phone per day. An they are mostly in the peak work day so lets say 500 per hour and the average length is 30 seconds. This is less than 5 concurrent messages :) I think you will need to have at least a T1 system because you are going to face some fairly extreme variations in usage. 2/ retrieve either via deliver to email or dial in to a number to collect voicemail via phone (or collect and play via a website) What does the conversion and how does one handle bulk updates? to users? How much control does the user have? How is a user informed that voicemail are waiting for them ? What is you existing PBX, how would the Asterisk based system interface with it ? does it use SIP ? or T1 interface ? How are the retrieving their voicemail now? Do you want to replicate this for ease of replacement as near as possible? Right now we are using the voice mail service provided by the teclo and are spending $0.06 per minute. The user connects to the voice mail by dialing *99 and entering a password on their office set or remoetely by dialing 123-MAIL on any phone (123 is the three digit prefix of their phone number) and then entering their password. They do not have any voice to email service today. If possible I would like to ease the transition if it can be done. Lots of stepswill follow discovery if it can be done. 3/ Not sure what you mean by tie in? How do you match a voice mail box to an email address? Can there be multiple email addresses for one voice mail box? You can program the Asterisk but with a good interface, click and drop. 4/ Sure, how do you have this configured at the moment? Why not replicate voicemail group delivery in the same format? Talkmail is a service provided by the telco where you group a bunch of numbers together so you can send the same message to all of them at the same time. Again it can be programmed but click and drop may be easier. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, 4 November 2006 11:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie questions about Voice mail I am totally ignorant about actually using asterisk for any purpose. I have read some of the docs but not all. I am currently doing a telephone audit for my company and one of the issues is voice mail. We are spending quit a bit of money with our telco for voice mail services and I was wondering about using asterisk as just a voice mail system. We are not quite ready to move to a full VOIP system yet but if I can get this system in place the VOIP will follow. Could I get
[asterisk-users] Asterisk CTI - SAP R/3 Intergration Certification
Hi everybody Does anyone know if exists some CTI software than is certificated by SAP? Thanks Sílvia Gallego Gonzalez [EMAIL PROTECTED] Optisistem: Optimización de Sistemas Empresariales S.L. Telf. (+34) 902 500 388 Fax. (+34) 93 217 67 77 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Glitches in sound every time that Asteriskreceives reINVITEs
In fact as far as I know, Asterisk stands in the middle of calls, breaking one transaction and initiating another to the other side, doing the bridge between them... Although good in some cases like permitting to start a new transaction to the next hop changing codecs, in my case I don't need that feature because I'm using reINVITEs to implement session-timer support in the user agent to solve problems of whong accounting if power failure or link happens... Is there any way to disable those breaks in audio stream? Regards, Ricardo. Andreas Sikkema wrote: My Asterisk server is working fine, although every time that in the middle of any call there is a reinvite, the user hears a glitch. Why is this happening? How can I solve this problem? That's because a REINVITE is generally used to change from one codec to another. For some reason this involves stopping the existing audio, waiting a little while and then starting a new audio stream. So far this one of the reasons why I don't like reinvite... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Tones occuring randomly
Hi Eric, i have replied but nobody seems to got a deeper knowledge of the problem. I have searched for talkoff, i found a lot of stuff, like check IRQs (checked, and good) and/or set relaxdtmf=no (it is set) or check the dtmf modes to be the same or or. But nothing of the things i found match to my problem except one thing i cant understand - there is an thread at digium with the advice to use the variable dtmfthreshold to set the level of dtmf detection, i cant find any variable like this. Do you know something where i can search ? I got this problem since 6 or 7 months and tried MANY solutions to get to my stable Asterisk, but i got no luck. What do you think about switching from rfc2833 to inband to solve this problem ? Thanks, Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no sound when bridging 2 asterisk SIP connections
Hello, here is our layout: asterisk-A --- WAN --- asterisk-HQ --- WAN --- asterisk-B calls are routed with SIP between asterisk's (found IAX to unreliable). When asterisk-HQ attempts to native-bridge OR simply forward calls between A and B no sound is sent. If either leg (A - HQ or B - HQ) is converted to IAX, then sound flows normally. We are using 1.2.13. What could be the problem? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk servers being greedy and not letting goof the media path. (using IAX2 channels)
For the benefit of the archives, my problem was a simple one. I hadn't forwarded the IAX port on the router of the remote * server connection, so when voip provider was trying to connect directly to the remote * server, it couldn't. Hurray for wasting an entire day over a simple silly little thing ;-) Mat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mat Stace Sent: 06 November 2006 17:42 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk servers being greedy and not letting goof the media path. (using IAX2 channels) Evening everyone (obviously depends on when you're readin this, but hey). I'm trying to set up a multi * server situation, and am falling over at the second server, and after a day of google etc, have come up against somewhat of a brick wall. I can make calls each way between the two servers no problem, and can include the required extension at the remote * server as part of my main incoming dialplan. My problem comes with * attempting to pass the media path to the other server. What is happening is: Incoming call from iax2 provider to main * server -- dial sip extension on main * server -- setup IAX2 channel to remote * server (which then rings extension) Pickup call on extension on remote * server -- main server sip extension stops ringing -- ast console on main server I get : - -- Attempting native bridge of IAX2/voipprovider/6 and IAX2/remote*server/7 -- Channel 'IAX2/voipprovider/6' unable to transfer -- Channel 'IAX2/remote*server/7' unable to transfer - In the user/friend declarations (user for incoming voip provider, friend for remote * server) in the two iax.conf files I have notransfer=no, and also up in the [general] section of the iax.conf. The problem is that when remote * user answers the phone, and then transfers the call to an extension on the main * server, there is massive (ie 2 seconds) delay, and using IAX2 show channels at the two consoles, the call is doing the following: PSTN - VOIP PROVIDER - main * server - remote * server - main * server - SIP extension on main * server. Anyone have any ideas on how to make the * servers give up the media path? Cheers Mat ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.13.28/518 - Release Date: 04/11/2006 17:30 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] faxing times!
Hello ppl, Reading all over the net. Learnt quite a lot, but that has left me confused-a-lot as well. Need answers to a few questions. Before that, I have an ISP(fax gateway) which will help me send/recv faxes using the T.38 protocol. I am using Asterisk 1.2.12.1. Now to the few questions I had: 1) Do I need any additional hardware on the Asterisk box?? I did download the spandsp and rxfax and txfax, n email2fax packages. But it seems, all those work on the Zap channels. 2) So far, I've worked ONLY with SIP and IAX. So, is it possible to do fax-ing over these? How? Help!! - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and peep tone (network tone)
Hi,I need to play a peep tone(to warn that he is going to another network) before ringing tone, when user is calling to mobile network. But peep tone must work under certain conditions, when destination is available( if unavailable - hangup). Is it possible to do with asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] flash transfer problem in asterisk integration with old PBX
HiCan you verify whether your PBX expects a hook flash for transfer or if it uses the Recall (or Flash) button on a telephone? Not an expert but I'm told by the real experts that they're different and my investigations http://lists.digium.com/pipermail/asterisk-users/2006-November/171749.htmlshow that the generated signals are different. Am going through the same problem trying to figure out how to generate the same signal that Recall does (for basically the same reason). Haven't had a response to my post, will let you know if I come up with anything.CheersOn 11/8/06, Andrea Giuliani [EMAIL PROTECTED] wrote:I've tried to transfer a call using the Flash command, but with my configuration it doesn't work.I have a traditional PBX connected with a zap channel to Asterisk that actslike an IVR:TELCO line -- traditional PBX (FXS) -- (FXO) AsteriskFrom the TELCO line I can make a call to the traditional PBX and reach Asterisk, the IVR system on Asterisk answers the call and I can dial anextension (for example 42 that is on the traditional PBX). In the asteriskdialplan I've set to transfer the call using Flash() like in this example: exten = 42,1,Flash()exten = 42,2,Background(silence/1) wait 1 second for the traditionalPBXexten = 42,3,SendDTMF(42,250)exten = 42,4,Background(silence/1) wait 1 second for the traditional PBXexten = 42,5,Hangup()When I dial the extension 42, the phone 42 on the traditional PBX rings butwhen I answer there isn't communication with the call from the TELCO lineand after a few seconds the line hangup. Here you can see what happen in asterisk CLI console: Executing Answer(Zap/4-1, ) in new stack-- Executing BackGround(Zap/4-1, a_suoni_plink/menu_esterno2) in new stack-- Playing 'a_suoni_plink/menu_esterno2' (language 'it')== CDR updated on Zap/4-1-- Executing Flash(Zap/4-1, ) in new stack-- Flashed channel Zap/4-1-- Executing BackGround(Zap/4-1, silence/1) in new stack -- Playing 'silence/1' (language 'it')-- Executing SendDTMF(Zap/4-1, 42) in new stack-- Executing BackGround(Zap/4-1, silence/1) in new stack-- Playing 'silence/1' (language 'it') -- Executing Hangup(Zap/4-1, ) in new stack== Spawn extension (incoming, 42, 5) exited non-zero on 'Zap/4-1'-- Hungup 'Zap/4-1'I've tried the following changes to the dialplan in my example but transfer still doesn't work:- I've tried to use wait(1) instead of Background(silence/1)- I've tried without Background(silence/1) orwait(1):exten = 42,1,Flash()exten = 42,2,SendDTMF(42,250) exten = 42,3,Hangup()- I've tried without the Hangup() instructions at the endHas anyone the same problem like me and any suggestions?___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why dont my messages get through
There is an option on the list server membership configuration screen that will disable receiving your own posts to the list. Maybe the OP accidentally disabled this feature. Bob... On Wed, 2006-11-08 at 06:09 +0200, Dovid B wrote: I have seen this mainly with gmail. the logic is why do you need your own postings. Fish around to see if there is a setting in Gmail where it will keep the email. I know for myself I want the email's that I sent. It lets me know that they went out as well as it helps for sorting the emails. - Original Message - From: Alex Robar To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 08, 2006 5:08 AM Subject: Re: [asterisk-users] Why dont my messages get through They do get through. Messages you send to the list won't get sent back to you, because you sent them. On 11/7/06, Christian [EMAIL PROTECTED] wrote: Hi, My messages to the list don't get through. This must be the tenth message i am trying to send! Please ignore this test message. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to reboot a Polycom phone remotely
Hi Rick, Well, if I told you I'd have to kill you :-) Seriously, taken from a very hidden Wiki page: http://www.voip-info.org/wiki/view/Polycom+reboot+hardphone+script From the CLI: sip notify polycom-check-cfg xxx being the registered name in SIP.conf. It is meant for the phones to check for new configuration and download it. Your phone of course has to be registered with your Asterisk and you need to have a provisoning server also,because Ibelieve the phone won't reboot if there isnt a new phone.cfg file in your provisioning server to download. I am not sure if it works when you only change the sip application (sip.ld) to a new version. I haven't really tried to find out the full and precise functionality, but it works on my phones when change anything in phone1.cfg on my provisioning server. It's even intelligent enough to wait until the end of a call of the phone is being used. Mike, happy to contribute answers instead of questions for once. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rick SmithSent: November 7, 2006 8:44 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] "Sticky" Polycom 501 keys and handset hmm, Id like to know that. How do you reboot remotely ? J From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MikeSent: Tuesday, November 07, 2006 2:13 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] "Sticky" Polycom 501 keys and handset Disregard my previous message, I succeeded in downgrading my phones. And it worked, thanks Rick for the info. Is there any Polycom-specific mailing list I should be on to be aware of stuff like that? Also, would you know how to check the version of sip.ld remotely? I know how to reboot remotely, and I did for a few phones, but my paranoid self would like to double check and see if the sip.ld 1.6.7 re-installed ok by checking the current version. Is that even possible? Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rick SmithSent: November 7, 2006 11:28 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] "Sticky" Polycom 501 keys and handset I had this EXACT same problem, and 2.0.x is the problem according to Polycom Tech Support. I had such a hard time explaining the problem, too Downgraded to 1.6.7 and all worked well again. Polycom says if youre using Asterisk, dont go past 1.6.7 until they say to. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MikeSent: Tuesday, November 07, 2006 11:02 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [asterisk-users] "Sticky" Polycom 501 keys and handset Hi, I've recently bought new Polycom 501 phones, upgraded to bootrom 3.2.2 and SIP 2.0.1. I just noticed something, which I first blamed on Asterisk and NATs (a 2 second silence at the beginning of a call). Something I'venoticed also on my old phone (which is having the same problem now, but its also been upgraded). My keys are sticky. Simple as that. Sometimes I press a number and the key comes up (the hardware seems fine) but the phone produces this lng tone as if I had pressed the key for 3 seconds. Even the receiver is sticky, giving my dialtone when I lift it only1-2 seconds after I lift the handset. It simply looks like the phone can't keep up, like a sluggishcomputer. Anybody has ever seem this? I'd like to downgrde to SIP 1.6.7 to see if the new sip app was the problem. How can I do that? I've placed the old sip.ld file where I had to, but the phone wont pick it up. Short of that, can somebody point me to the newest firmware (2.0.2) to see if thatwould help? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Odd results from fxotune?
I recently ran fxotune against our incoming PSTN lines to try and help with some echo problems. It produced the following fxotune.conf file: 2=8,253,2,244,255,10,244,3,253 3=4,0,0,0,0,0,0,0,0 4=4,0,0,0,0,0,0,0,0 I'm a bit surprised by all of the '0's for channels 34, esp. given that it's populated values for channel 2. Is this considered 'normal' behaviour, or is something amiss? Thanks, j ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 and Queues RealTime
Hi all, I would like to use the Agent Login feature with real-time queues it is not possible with asterisk 1.2, as described here : http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue The mantis bug describing the implementation of realtime queue is bug 4037. This bug includes some discussion on how to extend dynamic queues to also work with the member login feature. So, did you know if this is possible natively in asterisk 1.4 (or will be) ? Thanks Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Queues and multiple lines
Michael Sampson wrote .. Say I have agents using a softphone like eyebeam that has 6 lines. They log in to the queue. Say there are 3 agents in my queue. 3 calls come in and all three agents are on a call. Now a fourth call comes in. Is it possible to have it setup so that the 4 call rings on line 2 of one of my agents, if they don't get it within the time limit it rings on line 2 of another agent and so on. An agent can then put their current call on hold and go to the new call, say something like thanks for calling please hold, then go back to their first call, finish it up and then go back to the second call. Michael, I don't think you want to do this in a Contact Centre environment. Remember that once the agent has answered the call you have now locked the caller to that agent. If another agent becomes available first, they will no longer get the call. The free agent will sit idle (or get the next call in queue which is NOT the caller who was answered). The caller who was answered on line x by the other agent must wait in perpetuity for the agent to become available, yet their TALK TIME clock is running as the call WAS ANSWERED and ASSIGNED to the agent. You are better to play announcements during the queue wait time to say whatever you want communicated to the people in queue. This way they maintain their position in queue, the availability to be assigned to any available agent that becomes available and their call stats work out. The call stats are really important as this is how you are going to measure you agents. Even if you can separate the hold/talk times, your stats for the agents will become meaningless and hurt your Work Force Management (WFM) programs and seriously impair you ability to manage/measure your people. dbc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Tones occuring randomly
Stefan Agethen wrote: Hi Eric, i have replied but nobody seems to got a deeper knowledge of the problem. I have searched for talkoff, i found a lot of stuff, like check IRQs (checked, and good) and/or set relaxdtmf=no (it is set) or check the dtmf modes to be the same or or. But nothing of the things i found match to my problem except one thing i cant understand - there is an thread at digium with the advice to use the variable dtmfthreshold to set the level of dtmf detection, i cant find any variable like this. Do you know something where i can search ? I got this problem since 6 or 7 months and tried MANY solutions to get to my stable Asterisk, but i got no luck. What do you think about switching from rfc2833 to inband to solve this problem ? What codec are you currently using for voice? I have found that when nothing else works, playing with the gains on the Zap channel helped. Usually lowering them. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Queues and multiple lines
This is not necessary - unless you are setting call-limit in sip.conf, and don't have any patches on 1.2 to prevent app_queue from sending multiple calls to the same member, they will automatically receive the call on the second line appearance. (And third, and forth, and so on.) Wes Baehr -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of brandon kruz Sent: Tuesday, November 07, 2006 8:05 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Queues and multiple lines Using SIP: Just create another user account say the softphones user's name is bob: create [bob] (bob's main line on his softphone) create [bob1] (same configuration options, then you can do all your other configurations for this user ) hope this helps anyone is open to correcting me :] my 2 cents `KruZ~ From: Michael Sampson [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [asterisk-users] Queues and multiple lines Date: Tue, 07 Nov 2006 12:39:25 -0600 MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc10-f2.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Tue, 7 Nov 2006 11:19:53 -0800 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 4DB2F7FC803;Tue, 7 Nov 2006 11:40:05 -0700 (MST) Received: from psmtp.com (exprod8mx47.postini.com [64.18.3.147])by lists.digium.com (Postfix) with SMTP id 5FE2E7FC6FAfor asterisk-users@lists.digium.com;Tue, 7 Nov 2006 11:39:21 -0700 (MST) Received: from source ([207.195.195.18]) (using TLSv1) byexprod8mx47.postini.com ([64.18.7.10]) with SMTP; Tue, 07 Nov 2006 10:39:30 PST Received: from [192.168.1.35] ([71.39.108.129])by unix18.sihope.com (8.12.10/8.12.10) with ESMTP id kA7IdQSc050775for asterisk-users@lists.digium.com;Tue, 7 Nov 2006 12:39:26 -0600 (CST)(envelope-from [EMAIL PROTECTED]) X-Message-Info: txF49lGdW41fv5JCf0u+LC0BEkvsM92gePhvdubElwo= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com User-Agent: Thunderbird 1.5.0.7 (Windows/20060909) X-pstn-levels: (S:49.30289/99.9 FC:95.5390 LC:95.5390 R:95.9108 P:95.9108M:97.0282 C:98.6951 ) X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c X-pstn-addresses: from [EMAIL PROTECTED] [db-null] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk- users,mailto:asterisk-users- [EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk- users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 07 Nov 2006 19:20:03.0766 (UTC) FILETIME=[BA0B9560:01C702A1] Say I have agents using a softphone like eyebeam that has 6 lines. They log in to the queue. Say there are 3 agents in my queue. 3 calls come in and all three agents are on a call. Now a fourth call comes in. Is it possible to have it setup so that the 4 call rings on line 2 of one of my agents, if they don't get it within the time limit it rings on line 2 of another agent and so on. An agent can then put their current call on hold and go to the new call, say something like thanks for calling please hold, then go back to their first call, finish it up and then go back to the second call. I hope that made sense. I'm sure there is a way to get it done, but how flexible is the current queue system in Asterisk with stuff like this? -- Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Get FREE company branded e-mail accounts and business Web site from Microsoft Office Live http://clk.atdmt.com/MRT/go/mcrssaub0050001411mrt/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Question on Aastra phones and Astrisk
It only happens when you go from IAX/SIP -- asterisk box -- aastra phone. Doesn't happen PSTN -- asterisk box -- aastra phone. The aastra people have said they believe it is a codec negotiation issue... but the newest firmware didn't fix it send them packet dumps. On 11/7/06, shadowym [EMAIL PROTECTED] wrote: Running several Aastra 9133i and 480CT phones with v1.4 firmware CentOS 4.4, Asterisk 1.2.13, Zaptel 1.2.10, Freepbx2.1.3. Using all default settings I have not seen that problem. I am not exactly sure we are creating those exact same conditions but it sounds like standard extension use to multiple incoming calls correct? That is all we are doing plus some more complicated outgoing stuff. -Original Message- From: Matt [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 07, 2006 5:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question on Aastra phones and Astrisk *bump* Anyone? On 11/6/06, Curt Shaffer [EMAIL PROTECTED] wrote: I wanted to add what we have both seen on traffic captures. You see Caller 1's RTP stream. Call 2 comes in and you see the creation of its RTP stream. After Call 2 is put on hold the RTP stream from Caller 1 disappears without a trace never to return and this is when the one way audio is happening. And I also wanted to add that I am running 1.4.0 firmware for this phone. Thanks again! -Original Message- From: Curt Shaffer [mailto:[EMAIL PROTECTED] Sent: Monday, November 06, 2006 6:58 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Question on Aastra phones and Astrisk I'm the friend mentioned here. I am using the Aastra 480i CT. It is SIP to my PBX and IAX termination from the PBX to my provider. My issue has a slight twist to it but the same result. For instance his is always where as mine is frequent but not always. After I got to finally see it first hand today, I had to start over from Caller 1 5 times to get it to happen again. Caller 1 calls in and Person A answers. Caller 2 calls in and Person B answers. Person B puts caller 2 on hold and audio drops on Caller 1. So Person A can hear caller 1 but caller 1 cannot hear Person A. This happens more often when Call 1 is on the handset and Call 2 is on the portable or vis a vi, but this is not always the case. It does happen to 1 set only but just less frequent. I have tried carrierinvite=yes and no but this does not change the issue. The phones are behind a router but the external IP of the router is on the same network as the * box. Thanks! Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Monday, November 06, 2006 6:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Question on Aastra phones and Astrisk Hi, Some odd behaviour here. A friend and I were talking tonight, and it seems we have both seen the same problem. We are both using aastra phones (I am using 9113is).We have a connection to and from providers via SIP and IAX.When I place a call on the local hold of the phone, and then pick them back up I can hear them, but they can not hear me.However, if I park the call, and then pick it up again, the audio is fine. Tonight I tried placing a call on hold using a Sipura/Linksys ATA (that is just hitting 'flash', which basically puts the call on local hold and starts music).The problem did not manifest itself. Has anyone else had this issue? Do you have a fix for it? It is an astrisk issue or an aastra issue? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pressing * makes Asterisk destroy my call
You'll need to use another key, instead of *. The * key is hard coded for that hangup feature in queues. On 11/7/06, Stefan Agethen [EMAIL PROTECTED] wrote: I got an up2date Asterisk with SNOM360 as SIP and mISDN with 2 ISDN Cards, if i press in a call the * Asterisk, Asterisk destroys the call not, Asterisk lets him hang and do nothing, if i hangup, Asterisk tell me in the warnings-log that the bridging was not successfull ?! If have disabled the function to hangup in the features.conf, but the key is still available, can someone explain me whats going on there ? Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I need (some) help in configuring PAP2.
Hello, I need (some) help in configuring PAP2. Best regards, Twanny Azzopardi. Mob: ( 356 ) 79713618 Email: [EMAIL PROTECTED] Web: http://line.sytes.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [asterisk-users] Why dont my messages get through
Hello Alex, I also apologize. I have now changed email on this list since I thought there were problems with the previous one i was using. I actually got a copy of one of my messages to my previous email and I thought why not get the others to. I also checked the list archives but they weren't there. Once again, apologize for my test messages! Many thanks, Christian On 2006-11-07 at 22:55 Alex Robar wrote: I _was_ sure until mention it just now... I certainly don't get a copy of any messages I sent to the list, whether I send from my personal or office accounts. Maybe the way my mail clients are handling it? If so, my apologies to Christian. Alex On 11/7/06, Nick Hoffman [EMAIL PROTECTED] wrote: On 11/7/06, Christian [EMAIL PROTECTED] wrote: Hi, My messages to the list don't get through. This must be the tenth message i am trying to send! Please ignore this test message. On Wed November 8 2006 13:08, Alex Robar [EMAIL PROTECTED] wrote: They do get through. Messages you send to the list won't get sent back to you, because you sent them. Hi Alex. Are you sure about that? I receive a copy of every email I send to the list. -- Nick E: [EMAIL PROTECTED] P: +61 7 5591 3588 F: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ringing phones
Hi, I have a system that connects to the PSTN.What do I need to do so that when a call comes in, the system will start ringing the hunt group I have setup but not actually answer the call? The problem is the system is answering the call, and then passing 'ringing tones' back to the caller, so this makes the phone companies call-forward-no-answer not work since the telco thinks they have answered! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to indicate an non-existent number?
On Mon, Nov 06, 2006 at 06:47:01PM -0600, Eric ManxPower Wieling wrote: Louis-David Mitterrand wrote: Hello, Using a PRI (E1) with the euroisdn protocol, I don't seem to get any specific message from the telco when attempting to dial a non-existent number. Asterisk returns a busy/congested code, but nothing indicating the number's real status. How do you guys manage that issue? Do you record a message (sorry, the number dialed can't be completed) and play it when the PRI or BRI returns a specific code? And what code is that? We check the value of HANGUPCAUSE. DIALSTATUS is a VERY generic indication of the disposition of the call. It seems PRI and BRI here always return 3 as HANGUPCAUSE From the wiki: #define AST_CAUSE_NO_ROUTE_DESTINATION 3 This is less than explicit regarding an unallocated number (basically I testes by dialling impossibles numbers). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.x and video
Hi, I would like to know which is the lasted Asterisk 1.2.x version (branch or trunk) for video support with h264 codec, and where I can downloaded it. Thank You Jorge Mendoza ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] operator console
Talk to the folks at Asteria. The have a product called Reign. It looks just like your old interface, runs off .NET as a client on the machine. http://www.asteriasgi.com/pbx/reign On 11/7/06, Stephen Wingfield [EMAIL PROTECTED] wrote: Andres, The Bicom Systems Operator Panel is probably what you are looking for. OPCOM http://www.bicomsystems.com/docs/opcom/1.0/html/ This is included with every copy of PBXware and is fully supported. If you care to register you may order a trial of PBXware with our SOHO. Regards Steve steve 'at' bicomsystems 'dot' com - Original Message - From: Andres Paglayan To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 30, 2006 5:27 PM Subject: [asterisk-users] operator console Hi, My users are currently using an operator console interface like this: see it at: http://www.whssf.org/interface.jpg which came with a Praxon PDX we got about 5 years ago, which is now unsupported, it works very good and converts any analog phone plugged into the system into a powerful console, (provided you have a computer next to it) you just provide the box ip, user login, user pass, and extension, and voila. I'll be switching the company's phone system to Asterisk. I know * is way much more flexible and rich featured than the box we currently have, ...but I'll need to give the users a good mean to see what's going on, who is busy, easy transfer with click here and there, easy conference, easy queue handler, easy way to see/use many lines at the same time is there any best console they can use? I don't mind using a commercial product, if the only part we have to pay for is the gui, besides, we will buying the enterprise * version Thanks a bunch, Andres ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HANGUPCAUSE for unalocated number?
Hello, On your BRI or PRI's what do you guys get as HANGUPCAUSE when dialing an unalocated number? I always get 3 (no route) which is less than helpful. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Performance issues in Realtime
Hello everybody, I'd like to hear some success stories about the use of Asterisk Realtime in medium-large contexts, like 50 extensions. Don't you think that in those contexts the system could be overloaded from the excessive number of queries to the DB? So.. is anybody using ARA in those kind of deployments? Thanks in advance, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing phones
You did not mention what your FXO (connection to PSTN) hardware is??? Depending on what it is there may be configuration options for things like 'ring thru' and wether the fxo answers or passes the call to * Doug On Wed, 8 Nov 2006, Matt wrote: Hi, I have a system that connects to the PSTN.What do I need to do so that when a call comes in, the system will start ringing the hunt group I have setup but not actually answer the call? The problem is the system is answering the call, and then passing 'ringing tones' back to the caller, so this makes the phone companies call-forward-no-answer not work since the telco thinks they have answered! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium Asterisk-GUI problem
I just installed the Digium asterisk-gui from svn on to an asterisk 1.4 beta3 configuration. I can get to the main page, cfgbasic.html, and then log in OK, however after I log in and then each time I click on a new menu item I receive Stack overflow at line: 0. None of the data Fields on the screens populate from the config files. I am running IE7 on Win XP SP2. Any assistance is appreciated. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VLANs and Quality
Hi all How much does configuring a network with VLANs improve or effect quality ? Is there much reason to justify the configuration of VLANs ( I know networking, but not VLANs at all) Would it not be better to find high traffic users and determine why? Your Thoughts Thanks Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing phones
Apologies.. we are using a sangom 4 port FXO card. It used to work (or so the company claims that has the PBX), but they are saying it stopped.. yet nothing has changed on the PBX system. I have verified it IS picking up and then passing the call onto the ringgroup (hence taking it out of the phone companies domain). On 11/8/06, Doug Crompton [EMAIL PROTECTED] wrote: You did not mention what your FXO (connection to PSTN) hardware is??? Depending on what it is there may be configuration options for things like 'ring thru' and wether the fxo answers or passes the call to * Doug On Wed, 8 Nov 2006, Matt wrote: Hi, I have a system that connects to the PSTN.What do I need to do so that when a call comes in, the system will start ringing the hunt group I have setup but not actually answer the call? The problem is the system is answering the call, and then passing 'ringing tones' back to the caller, so this makes the phone companies call-forward-no-answer not work since the telco thinks they have answered! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] jpeglib
Hello, When i try to install the sfftobmp3.1, the tribbox box give me the following error: ... checking for TIFFOpen in -ltiff... yes checking jpeglib.h usability... no checking jpeglib.h presence... no checking for jpeglib.h... no configure: error: jpeglib.h not found I try to find packages with jpeglib but i cannot find that... :( Someone can tell me where i can find that package? Thanks in advance! PS. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] flash transfer problem in asterisk integration with old PBX
What sort of interface are you using? Is there any way you could get diagnostics from the pbx? On 11/8/06, Andrea Giuliani [EMAIL PROTECTED] wrote:I've tried to transfer a call using the Flash command, but with my configuration it doesn't work.I have a traditional PBX connected with a zap channel to Asterisk that actslike an IVR:TELCO line -- traditional PBX (FXS) -- (FXO) AsteriskFrom the TELCO line I can make a call to the traditional PBX and reach Asterisk, the IVR system on Asterisk answers the call and I can dial anextension (for example 42 that is on the traditional PBX). In the asteriskdialplan I've set to transfer the call using Flash() like in this example: exten = 42,1,Flash()exten = 42,2,Background(silence/1) wait 1 second for the traditionalPBXexten = 42,3,SendDTMF(42,250)exten = 42,4,Background(silence/1) wait 1 second for the traditional PBXexten = 42,5,Hangup()When I dial the extension 42, the phone 42 on the traditional PBX rings butwhen I answer there isn't communication with the call from the TELCO lineand after a few seconds the line hangup. Here you can see what happen in asterisk CLI console: Executing Answer(Zap/4-1, ) in new stack-- Executing BackGround(Zap/4-1, a_suoni_plink/menu_esterno2) in new stack-- Playing 'a_suoni_plink/menu_esterno2' (language 'it')== CDR updated on Zap/4-1-- Executing Flash(Zap/4-1, ) in new stack-- Flashed channel Zap/4-1-- Executing BackGround(Zap/4-1, silence/1) in new stack -- Playing 'silence/1' (language 'it')-- Executing SendDTMF(Zap/4-1, 42) in new stack-- Executing BackGround(Zap/4-1, silence/1) in new stack-- Playing 'silence/1' (language 'it') -- Executing Hangup(Zap/4-1, ) in new stack== Spawn extension (incoming, 42, 5) exited non-zero on 'Zap/4-1'-- Hungup 'Zap/4-1'I've tried the following changes to the dialplan in my example but transfer still doesn't work:- I've tried to use wait(1) instead of Background(silence/1)- I've tried without Background(silence/1) orwait(1):exten = 42,1,Flash()exten = 42,2,SendDTMF(42,250) exten = 42,3,Hangup()- I've tried without the Hangup() instructions at the endHas anyone the same problem like me and any suggestions?___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing phones
Why don't you post your configuration?On 11/8/06, Matt [EMAIL PROTECTED] wrote: Apologies.. we are using a sangom 4 port FXO card. It used to work(or so the company claims that has the PBX), but they are saying itstopped.. yet nothing has changed on the PBX system.I have verifiedit IS picking up and then passing the call onto the ringgroup (hence taking it out of the phone companies domain).On 11/8/06, Doug Crompton [EMAIL PROTECTED] wrote: You did not mention what your FXO (connection to PSTN) hardware is??? Depending on what it is there may be configuration options for things like 'ring thru' and wether the fxo answers or passes the call to * Doug On Wed, 8 Nov 2006, Matt wrote: Hi, I have a system that connects to the PSTN.What do I need to do so that when a call comes in, the system will start ringing the hunt group I have setup but not actually answer the call?The problem is the system is answering the call, and then passing 'ringing tones' back to the caller, so this makes the phone companies call-forward-no-answer not work since the telco thinks they have answered! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safetydeserve neither liberty nor safety.-- Ben Franklin (1759) *Doug Crompton * *Richboro, PA 18954* *215-431-6307* ** * [EMAIL PROTECTED]* * http://www.crompton.com* ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I make this stop? (Bridging of IAX channels?)
So if I have notransfer=yes, why is it 'returning from native bridge'? Nov 8 10:07:51 VERBOSE[21620] logger.c: -- Attempting native bridge of IAX2/peer1-iax-10 and IAX2/peer2-test-14 Nov 8 10:13:06 DEBUG[21620] channel.c: Returning from native bridge, channels: IAX2/peer1-iax-10, IAX2/peer2-test-14 On 11/7/06, Joshua Colp [EMAIL PROTECTED] wrote: Matt wrote: -- Attempting native bridge of IAX2/peer1-iax-7 and IAX2/peer2-21 I want everything to stay in the VoIP server rather then briding. I have notransfer=yes on, but it still seems to bridge the call natively.. can I keep the RTP stream on the asterisk server some how? Asterisk is still going to try to native bridge the two channels. Once this occurs chan_iax2 is going to notice that you don't want a native transfer to happen and not do it. -- Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I make this stop? (Bridging of IAX channels?)
Yet.. I am getting CDR records.. or am I misunderstanding what a native bridge is? On 11/8/06, Matt [EMAIL PROTECTED] wrote: So if I have notransfer=yes, why is it 'returning from native bridge'? Nov 8 10:07:51 VERBOSE[21620] logger.c: -- Attempting native bridge of IAX2/peer1-iax-10 and IAX2/peer2-test-14 Nov 8 10:13:06 DEBUG[21620] channel.c: Returning from native bridge, channels: IAX2/peer1-iax-10, IAX2/peer2-test-14 On 11/7/06, Joshua Colp [EMAIL PROTECTED] wrote: Matt wrote: -- Attempting native bridge of IAX2/peer1-iax-7 and IAX2/peer2-21 I want everything to stay in the VoIP server rather then briding. I have notransfer=yes on, but it still seems to bridge the call natively.. can I keep the RTP stream on the asterisk server some how? Asterisk is still going to try to native bridge the two channels. Once this occurs chan_iax2 is going to notice that you don't want a native transfer to happen and not do it. -- Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Microsoft will enter VoIP market in earnestnextyear, says Ballmer
I took a look at this slide show and saw a lot that I like ... this level of integration between voice and email has been along time coming and I think it will eventually sell very well ... my initial reaction is, I WANT IT NOW ... we use an Exchange 2003 server in our own shop ... I have for some time considered moving to a Linux based platform for our Email as I believe it to be less costly/complex to manage and maintain ... And I really do not like the Active Directory model that Exchange relies on ... however, when this level of functionality becomes available in Exchange, I will have something new to consider in the equation ... I am not a big fan of Exchange ... I have installed and supported this system for many years and have developed something of a love/hate relationship with it ... however, if it becomes clear that I could only get these features with Exchange, I would give serious consideration to buying my Exchange server a box of candy and some flowers ... the questions I would like to see answers to are these ... where is the controlling interface for this integration ??? is it at the mail server ??? is it at the voice server ??? is it at the web client ??? or is it a combination of all of the above ??? if you have limited resources, where do you allocate them to most efficiently achieve the goal ??? is there a design concept that would permit this level of integration to be implemented using my choice of email server and client in combination with Asterisk ??? or am I doomed to maintaining a relationship with an 800 gorilla from Redmond ??? I would be very cautious about investing effort in an implementation that was Microsoft specific as the Microsoft API's have a history of being a moving target that only the chosen few can hit ... I think that all the individual pieces of what I saw in this slide show are available NOW as open source components ... it looks to me like the big challenge here is integrating all the pieces so they all work together to provide the desired functionality ... am I wrong about this analysis ??? G.Hendershot From: Curt Shaffer [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 07, 2006 11:08 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Microsoft will enter VoIP market in earnestnextyear, says Ballmer Take a look at OVA.. mms://wm.microsoft.com/ms/exchange/2007/Phone_Based_User_Experience_With_Out look_Voice_Access_300k.wmv From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Robar Sent: Tuesday, November 07, 2006 9:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Microsoft will enter VoIP market in earnest nextyear, says Ballmer Unified messaging would be nice. Not just having my VM's e-mailed to me, but to be able to manage them from with Outlook (or any other mail client for that matter) would be nice. I picture it sort of like an IMAP mailbox, and the mail client just has some kind of functionality to recognize that the message is a VM and not a mail message (so it could display length, date/time received, CID, and provide a play button). Just my two cents. Alex On 11/7/06, Dean Collins [EMAIL PROTECTED] wrote: http://www.siliconvalley.com/mld/siliconvalley/business/international/asia/1 5944981.htm There's not much in the article so only click through if super interested but I'm curious and looking for people's opinions. What application integration would you like to see between MS (either Office or other aspects of the vista/xp OS) and Asterisk. Apart from dial from outlook and number pop I'm kind of curious what other functionality there is to be developed (I'd also like to see drop and drag from outlook into conference calls. What would you like to see in asterisk, if we get some solid responses we'll see about organizing some bounties to get it developed. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VLANs and Quality
On Wed, 2006-11-08 at 10:15 -0500, Barry Fawthrop wrote: Hi all How much does configuring a network with VLANs improve or effect quality ? Is there much reason to justify the configuration of VLANs ( I know networking, but not VLANs at all) Would it not be better to find high traffic users and determine why? That goes for any network ;-). What you _really_ don't want is some guy uploading the lastest holiday movie to your fileserver and bringing down your entire companies' telephone system. However, you might care less about your fileserver running slow during that time. Or: someone plugs in a Apple Laptop with DHCP server enabled and your phones suddenly all get new IP-Addresses. So, you essentially build 2 seperate networks with (almost) seperate levels of bandwidth available. I prefer to use a seperate switch(es) for phones than for data but settle for vlans. I might even use multiple vlans for phones and multiple vlans for data, depending on topology and usage. So I say: yes - there is reason for configuring vlans. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FIC-GTA001
Anyone on the asterisk list have any thoughts about the new Open Moko linux mobile? http://www.theinquirer.net/default.aspx?article=35590 http://www.linuxdevices.com/news/NS2986976174.html http://linux.slashdot.org/linux/06/11/08/004230.shtml Is there any integration into Asterisk that we can look at? Anyone want to through some application ideas theyd like to see developed and we can throw some bounty money at it? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes MP-114 noise
Jason,I think it's something only supported in the newer firmware. Get in contact with the place you bought the unit from, they should be able to get the latest firmware for you.MP118_SIP_F4.80A.034.004.cmp should work for MP114, this is what we used and all that we could get from Audiocodes, I think they should have named it MP11x though. :) Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Nov 8, 2006, at 2:25 AM, Jason Kim wrote:Jessee,Thank you for your help.I downloaded firmware and sample configuration files.But the firmware was old version for MP118 and MP124.Where can i download recent one?Can i upload only ini file to changecountrycoefficient ?Regards,Jason.--- Jessee J Holmes [EMAIL PROTECTED] wrote: Jason,First, before you start reading, get to the latestfirmware from Audiocodes (MP118_SIP_F4.80A.034.004.cmp), therehave been significant echo improvements in this version.After many days of working with Audiocodes on thisproblem and much time spent here by multiple technicians trying toreproduce and resolve this issue; this morning, Atacomm receivedan email from Audiocodes with a full explanation to this nowconfirmed issue with all MP-11x units. Atacomm will immediately beginwork on a KB article within our website that confirms this issue andoutlines the manufacturer recommended steps to resolve thisproblem.Apparently, there have been some changes with theMP-11x's that can negatively affect line noise and echo. Below aresome steps which can help to correct these problems:1. The new design did away with the Coefficent file. Audiocodes, now instead, introduced a configurable parameter called countrycoefficient. This parameter can be adjustedto a specific country based on known configurations. For the mostpart this should work. 70(USA) is the default value. More can befound in the User’s manual.2. In just about every case, an FXO is added to aPre-existing PBX or CO line, you can expect echo. This comes from thefact that delay (IP Network) is being introduced, and what used tobe Side tone is now delayed so much it is echo. Just about everydifference on the line that can be heard between the pre fxo and postfxo installation can be traced to echo, or line quality issues.3. Going forward, Audiocodes would like to suggestthat when installing the product do the following:A) Make sure the Line coming from the PBX or CO is aLoop Start line. Ground start is not supported on the MP-11x seriesof gateways. (The M1K FXO will in 5.0)B) Check that the Line can deliver for a 600 OhmImpedance line-52 to -24 V of Off Hook Voltage-15 to -6 V of On Hook Voltage20 to 35 ma of loop current.If you know the line is not 600 Ohm, please gathermetrics on the line, and the make and model of the PBX or switch itis attached too, plus country of origin. If it is not from the USA,please look up the country of origin and then find theCountryCoefficient to match this. Load the .ini file to the board with this settingand reset. Make sure the Gateway has a firmware version of 4.60.035or higher or 4.80.030 or higher.C) Put the device on the network with Voice Volumeset to 0 and input gain set to 0. Make calls, if there is no issue, youcan stop here. However, Echo is still expected most of the time.D) The echo should be heard by the IP sideparticipant as their voice is reflected back. If this is the case, then whatneeds to be done is to lower the voicevolume (IP—TEL). This way thespeaker’s reflected voice will comeback low enough for theECAN to cancel it out (-6 is usually recommended as the value to plugin here). A little experimentation is needed as the loss for alllines will vary based on length from the CO. Echo is usually takencare of in this manner.E) The incoming speaker from the PSTN’s voice seemslow, set InputGainLocation =1, and then slowly increment theInput Gain Parameter(Tel?P) to adjust for this. In pastreleases (see the note about loads above), the input gain was alwaysapplied prior to the ECAN which had the effect of amplifying the returnedecho and noise on the line causing crosstalk and clipping issues.This is no longer the case.If the above does not resolve the issues, then youneed to go ahead and collect DSP, Ethereal and Syslog traces alongwith the board.ini, these are to be sent to your support agent, who willthen send these to Audiocodes for their engineers to evaluate. Thisshould not happen often.Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIPstore at http:// voipstore.atacomm.com/On Nov 3, 2006, at 12:14 AM, Jason Kim wrote:Jessee,I tried many combinations of "Voice Volume", "InputGain" and packetization time , but it's noisy steel.I'm using G.711A-law and packetization time is 20ms.It can be impedance mismatch problem but i
[asterisk-users] SIP CANCEL NOT WORKING
Hi All. I'm using asterisk 1.2.10, 1.2.13 and 1.4.0-beta3 wth softphone eyebeam 15 I do the following: eyebeam call to PSTN phone 911234567 and asterisk can't create a zap channel sends CANCEL to eyebeam. The log of eyebeam shows this: [06-11-08]16:52:07.415 | Info (debug) RESIP:TRANSACTION | Matching rule for CANCEL :[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.2.54:5060;branch=z9hG4bK4d29449f;rport=5060 Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] To: 621sip:[EMAIL PROTECTED];tag=6626f537 From: 916331591sip:[EMAIL PROTECTED] Call-ID: 0719856da42f542bZGE1NzllOTI3ZGU4NjIwNDhiOTVjOGJkZmFmOTgxNDk. CSeq: 101 CANCEL User-Agent: Asterisk PBX Content-Length: 0 The first line is incorrect, must be CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Where is sip between CANCEL and ':'? Thanks! --- Mario Fdez. Alonso ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Tones occuring randomly
What codec are you currently using for voice? I have found that when nothing else works, playing with the gains on the Zap channel helped. Usually lowering them. I use rfc2833 for dtmf, alaw as codec. Yes, a lowering could be a idea, but the problem is logged on any kind of channels in my system, like zap, misdn, sip and iax. That is my problem :( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing phones
Apologies.. we are using a sangom 4 port FXO card. It used to work (or so the company claims that has the PBX), but they are saying it stopped.. yet nothing has changed on the PBX system. I have verified it IS picking up and then passing the call onto the ringgroup (hence taking it out of the phone companies domain). Matt, check in your incoming context that you don't have an Answer before you dial the ringgroup. If you don't answer and just dial the ringgroup, Asterisk won't pickup the incoming call until a phone in the ringgroup answers it. hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: FIC-GTA001
I just found a link to this presentation that has some more information http://www.linuxdevices.com/files/article072/sld002.html Cheers, Dean From: Dean Collins Sent: Wednesday, 8 November 2006 11:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: FIC-GTA001 Anyone on the asterisk list have any thoughts about the new Open Moko linux mobile? http://www.theinquirer.net/default.aspx?article=35590 http://www.linuxdevices.com/news/NS2986976174.html http://linux.slashdot.org/linux/06/11/08/004230.shtml Is there any integration into Asterisk that we can look at? Anyone want to through some application ideas theyd like to see developed and we can throw some bounty money at it? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing phones
The config is pretty simple.. when a call comes in it does an Answer(), which obviously is going to stop the phone companies no-answer-call-forward from working. My question, better perhaps, is.. is there a way to cause asterisk to push the ringing through to my ring group, without actually answering the line? On 11/8/06, Andrew Joakimsen [EMAIL PROTECTED] wrote: Why don't you post your configuration? On 11/8/06, Matt [EMAIL PROTECTED] wrote: Apologies.. we are using a sangom 4 port FXO card. It used to work (or so the company claims that has the PBX), but they are saying it stopped.. yet nothing has changed on the PBX system. I have verified it IS picking up and then passing the call onto the ringgroup (hence taking it out of the phone companies domain). On 11/8/06, Doug Crompton [EMAIL PROTECTED] wrote: You did not mention what your FXO (connection to PSTN) hardware is??? Depending on what it is there may be configuration options for things like 'ring thru' and wether the fxo answers or passes the call to * Doug On Wed, 8 Nov 2006, Matt wrote: Hi, I have a system that connects to the PSTN.What do I need to do so that when a call comes in, the system will start ringing the hunt group I have setup but not actually answer the call? The problem is the system is answering the call, and then passing 'ringing tones' back to the caller, so this makes the phone companies call-forward-no-answer not work since the telco thinks they have answered! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Delay between DTMF Down Detected Digit
Good Morning,
I've recently gotten Asterisk installed and configured our IVR using
FreePBX. Things seem to be going well except a few of our inbound
callers are ending up in the wrong place when trying to connect to a
specific extension. The example I had this morning was someone trying to
call extension 212 and getting connected to the Sales queue which is
option 2 on the IVR. I looked in our log and it seems like there was a
seventeen second delay between the caller pressing the last 2 and when
Asterisk acknowledged it. By that time, Asterisk had decided that 21
wasn't a valid extension and the subsequent 2 dropped the caller into
the Sales queue.
I did my best to search for this issue in the archives and I found one
reference to relaxdtmf, but I wasn't sure if that would address the
issue and I wouldn't want it to cause talkoff.
For reference, we're using a Wildcard TE410P for these incoming calls.
I've included the configuration for the ivr and a scrubbed segment from
the log. If any additional information is needed, please let me know.
Any help is appreciated in advance!
Jon
[ivr-3]
include = ivr-3-custom
include = ext-findmefollow
include = ext-local
include = app-directory
exten = h,1,Hangup
exten = s,1,Set(LOOPCOUNT=0)
exten = s,n,Set(__DIR-CONTEXT=default)
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n(begin),Set(TIMEOUT(digit)=3)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(custom/RM_Daytime)
exten = hang,1,Playback(vm-goodbye)
exten = hang,n,Hangup
exten = 0,1,Goto(ext-queues,300,1)
exten = 1,1,Goto(ext-queues,300,1)
exten = 2,1,Goto(ext-queues,400,1)
exten = 7,1,Goto(ext-queues,700,1)
exten = t,1,Goto(ext-queues,300,1)
exten = i,1,Playback(invalid)
exten = i,n,Goto(loop,1)
exten = loop,1,Set(LOOPCOUNT=$[${LOOPCOUNT} + 1])
exten = loop,n,GotoIf($[${LOOPCOUNT} 2]?hang,1)
exten = loop,n,Goto(ivr-3,s,begin)
exten = fax,1,Goto(ext-fax,in_fax,1)
; end of [ivr-3]
Nov 8 11:13:53 VERBOSE[24018] logger.c: -- Accepting call from
'XX' to 's' on channel 0/7, span 1
Nov 8 11:13:53 DEBUG[24018] chan_zap.c: Enabled echo cancellation on
channel 7
...
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Exception on 23, channel 7
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Got event Event 131122(131122)
on channel 7 (index 0)
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: DTMF Down '2'
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Exception on 23, channel 7
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Got event Event 262194(262194)
on channel 7 (index 0)
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Detected digit '2'
Nov 8 11:13:58 DEBUG[3561] pbx.c: Oooh, got something to jump out with
('2')!
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Exception on 23, channel 7
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Got event Event 131121(131121)
on channel 7 (index 0)
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: DTMF Down '1'
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Exception on 23, channel 7
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Got event Event 262193(262193)
on channel 7 (index 0)
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Detected digit '1'
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Exception on 23, channel 7
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Got event Event 131122(131122)
on channel 7 (index 0)
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: DTMF Down '2'
Nov 8 11:14:01 VERBOSE[3561] logger.c: -- Invalid extension '21' in
context 'ivr-3' on Zap/7-1
Nov 8 11:14:01 VERBOSE[3561] logger.c: == CDR updated on Zap/7-1
Nov 8 11:14:01 VERBOSE[3561] logger.c: -- Executing
Playback(Zap/7-1, invalid) in new stack
Nov 8 11:14:01 DEBUG[3561] channel.c: Scheduling timer at 160 sample
intervals
Nov 8 11:14:01 DEBUG[24018] chan_zap.c: Echo cancellation already on
...
Nov 8 11:14:15 DEBUG[3561] chan_zap.c: Exception on 23, channel 7
Nov 8 11:14:15 DEBUG[3561] chan_zap.c: Got event Event 262194(262194)
on channel 7 (index 0)
Nov 8 11:14:15 DEBUG[3561] chan_zap.c: Detected digit '2'
___
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[asterisk-users] Re: Performance issues in Realtime
Hello everybody, I'd like to hear some success stories about the use of Asterisk Realtime in medium-large contexts, like 50 extensions. Don't you think that in those contexts the system could be overloaded from the excessive number of queries to the DB? So.. is anybody using ARA in those kind of deployments? Thanks in advance, Master database across network segment has performance limitations as Cluster scales Replicate the Master database to each registration server Slave database MySQL replication is simpler and cost less to implement than MySQL Cluster Increase performance, small log file sent from Master to Slave database Setup res_mysql to read from the local Slave database and write to the Master database (Digium Bug Tracker, Mantis Issue 5881) JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Agents that handle calls from multiple queues
Title: Message What about creating _two_ appearances on the phone, one for each queue? -Original Message-From: Ardjan Zwartjes [mailto:[EMAIL PROTECTED]Sent: Wednesday, November 08, 2006 2:20 AMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Agents that handle calls from multiple queues Hi everybody, I've got an Asterisk configuration where an agent handles calls from multiple queues. At the moment I'm using the default Queue application and I encountered the following problem: When there are calls waiting in multiple queues the selection of which call is handled by the Agent is more or less random. It would be nice if the call that was waiting the longest was handled first. I've been looking at ICD as an alternative to the Queue application but as far as I could see this project hasn't been updated for quite some time now. Does anybody know of an alternative or a way to get the desired behaviour? Thanks, Ardjan Zwartjes. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk and Max TNT PRI to SIP Authentication Issue
what is the sip.conf for 1239 which I'm going to assume is a extension on the TNT Barry JR Richardson wrote: Hi All, I have a lab setup with two asterisk servers and a MAX TNT in the middle like this: asterisk sip sip TNT pri pri asterisk exten 1239 is the CID Number from the originating caller on the PRI side, has no relation to the local user on the sip side of the call. -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk and Max TNT PRI to SIP Authentication Issue
When all else fails I resort to adding this in the sip.conf peer config: Insecure=invite,port It took me a while to figure out they can be used together. Regards, Scott Thanks Scott, i have it set to that, but that has no effect. The incoming call still requires proxy authentication. I've also tried in both general and max context insecure=invite,port autocreatepeer=yes allowguest=yes allowexternalinvites=yes trustrpid = yes There is something different in asterisk 1.0.10 and 1.2. I've tried variations of all sip.conf switches to accept unauthenticated calls and nothing seems to work. I'm wondering is there is a patch that will allow unauthenticated calls in sip? Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Echo Issues
On Nov 6, 2006, at 8:06 AM, Steven wrote: Matt, How does one check for this?? You would probably know from the dmesg output card, just make sure it's using the Octasic based echo canceler. I think it says something about a VPM450M in the dmesg logs if it's the version I'm thinking of. If it's not, talk to RMA and see if you can get it updated. Matthew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Tones occuring randomly
I get the same thing using inband -- funny thing I am the only one who hears the random tones -- other party does not hear them and they are not recorded with the monitor app. on Wednesday 11/08/2006 Eric \ManxPower\ Wieling([EMAIL PROTECTED]) wrote Stefan Agethen wrote: Hi Eric, i have replied but nobody seems to got a deeper knowledge of the problem. I have searched for talkoff, i found a lot of stuff, like check IRQs (checked, and good) and/or set relaxdtmf=no (it is set) or check the dtmf modes to be the same or or. But nothing of the things i found match to my problem except one thing i cant understand - there is an thread at digium with the advice to use the variable dtmfthreshold to set the level of dtmf detection, i cant find any variable like this. Do you know something where i can search ? I got this problem since 6 or 7 months and tried MANY solutions to get to my stable Asterisk, but i got no luck. What do you think about switching from rfc2833 to inband to solve this problem ? What codec are you currently using for voice? I have found that when nothing else works, playing with the gains on the Zap channel helped. Usually lowering them. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: HANGUPCAUSE for unalocated number?
You should get 3 if the number is not valid for any routing database. You should get 1 if there is an athorative switch for that number, but it is not assigned. With DIDs. you get 3 if the number has not been assigned to any telco. you get 1 if it is assigned to a telco, but not an und user. Ma Bell numbers are the same. If it will route to a CO switch, but it has been disconnected, you should get a 1. If you ALWAYS get a 3, your telco is doing it wrong. -- -- Steven http://www.glimasoutheast.org Louis-David Mitterrand [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hello, On your BRI or PRI's what do you guys get as HANGUPCAUSE when dialing an unalocated number? I always get 3 (no route) which is less than helpful. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing phones
Hi Matt - The config is pretty simple.. when a call comes in it does an Answer(), which obviously is going to stop the phone companies no-answer-call-forward from working. My question, better perhaps, is.. is there a way to cause asterisk to push the ringing through to my ring group, without actually answering the line? Yes, as suggested earlier, just don't use the Answer() statement. Just skip it and go directly to the Dial() command for your ring group. The only real reason to do an Answer() before a Dial() is if you're getting audio-skippage (a very technical term) at the beginning of a call. This can happen on some FXO cards and phone lines, but it should generally work without the Answer(). - Noah On 11/8/06, Andrew Joakimsen [EMAIL PROTECTED] wrote: Why don't you post your configuration? On 11/8/06, Matt [EMAIL PROTECTED] wrote: Apologies.. we are using a sangom 4 port FXO card. It used to work (or so the company claims that has the PBX), but they are saying it stopped.. yet nothing has changed on the PBX system. I have verified it IS picking up and then passing the call onto the ringgroup (hence taking it out of the phone companies domain). On 11/8/06, Doug Crompton [EMAIL PROTECTED] wrote: You did not mention what your FXO (connection to PSTN) hardware is??? Depending on what it is there may be configuration options for things like 'ring thru' and wether the fxo answers or passes the call to * Doug On Wed, 8 Nov 2006, Matt wrote: Hi, I have a system that connects to the PSTN.What do I need to do so that when a call comes in, the system will start ringing the hunt group I have setup but not actually answer the call? The problem is the system is answering the call, and then passing 'ringing tones' back to the caller, so this makes the phone companies call-forward-no-answer not work since the telco thinks they have answered! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing phones
Matt wrote: The config is pretty simple.. when a call comes in it does an Answer(), which obviously is going to stop the phone companies no-answer-call-forward from working. My question, better perhaps, is.. is there a way to cause asterisk to push the ringing through to my ring group, without actually answering the line? Yes, don't execute Answer() before the Dial. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing phones
Ahh ok.. thanks. On 11/8/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Matt wrote: The config is pretty simple.. when a call comes in it does an Answer(), which obviously is going to stop the phone companies no-answer-call-forward from working. My question, better perhaps, is.. is there a way to cause asterisk to push the ringing through to my ring group, without actually answering the line? Yes, don't execute Answer() before the Dial. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One-Way-Audio After placing call on hold
I (and some others) are having an issue with placing calls on hold. Our setup is as follows: IAX2 or SIP terminator/originator --- asterisk box --- SIP Phones I have tried asterisk verison 1.0.9, 1.2.6 and 1.2.12, and all have the same issue. When I place a call on hold that has come in a PSTN channel (through a PRI and a Digium card) everything is fine. When I place a call on hold that has come in or gone out an IAX2 or SIP terminator or originator, when I pick the call back up often there is one-way-audio (I can hear the caller, but the caller can not hear me). I have attached a packet capture of the situation, and as you can see at about packet #3803 the audio goes one way. Can anyone enlighten me as to why this is happening, and why asterisk is no longer sending audio back to the terminator? I would like to get this fixed, obviously. (This file was a tcpdump, and can be opened in ethereal/wireshark) EDIT: Packet capture at following address: http://hecate.chilitech.net/~matth/zootdump.cap ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk iax2 monitoring
On 11/8/06, Thomas Blanchin [EMAIL PROTECTED] wrote: Hi David. I read your post on : http://lists.digium.com/pipermail/asterisk-users/2006-September/167456.html I am in the same situation as you are. I'm looking for a way to monitor iax2 connexions on asterisk. I'm using sipsak for sip connexions. I'm looking for a very simple tool, like sipsak, because I'm using BigBrother for global monitoring, so I just need an app that returns something or an exit code, and then BB do the rest. If you founded something, please let me know, I'm interested. Cheers. Thomas Hi Thomas, I have not found a good solution yet. I did find a tiny app called iaxping. http://rpm.pbone.net/index.php3/stat/4/idpl/3029621/com/iaxping-0-1mdv2007.0.i586.rpm.html This worked well to test the connection, but I could not get it to exit cleanly. I was thinking I might try to fix the code, but I'm not much of a C programmer. The source code (iaxping.c) is available online. I did not actually try to recompile the code so the problem might just be with the rpm. After you look at it, let me know your thoughts. If nothing else, maybe we get together and hire a decent programmer to fix the app to exit properly with a return code. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: One-Way-Audio After placing call on hold
My iax.conf is: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) delayreject=yes disallow=all allow=ulaw allow=gsm jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=100 jittershrinkrate=1 notransfer=yes trunk=no [zoot] type=user secret= auth=plaintext host=zoot.xx.net notransfer=yes context=from-trunk On 11/8/06, Matt [EMAIL PROTECTED] wrote: I (and some others) are having an issue with placing calls on hold. Our setup is as follows: IAX2 or SIP terminator/originator --- asterisk box --- SIP Phones I have tried asterisk verison 1.0.9, 1.2.6 and 1.2.12, and all have the same issue. When I place a call on hold that has come in a PSTN channel (through a PRI and a Digium card) everything is fine. When I place a call on hold that has come in or gone out an IAX2 or SIP terminator or originator, when I pick the call back up often there is one-way-audio (I can hear the caller, but the caller can not hear me). I have attached a packet capture of the situation, and as you can see at about packet #3803 the audio goes one way. Can anyone enlighten me as to why this is happening, and why asterisk is no longer sending audio back to the terminator? I would like to get this fixed, obviously. (This file was a tcpdump, and can be opened in ethereal/wireshark) EDIT: Packet capture at following address: http://hecate.chilitech.net/~matth/zootdump.cap ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Re: Echo Issues
Mine is a VPM400 on a TE410P (2nd Gen) Purchased as a TE411P -- -- Steven http://www.glimasoutheast.org Matthew Fredrickson [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] On Nov 6, 2006, at 8:06 AM, Steven wrote: Matt, How does one check for this?? You would probably know from the dmesg output card, just make sure it's using the Octasic based echo canceler. I think it says something about a VPM450M in the dmesg logs if it's the version I'm thinking of. If it's not, talk to RMA and see if you can get it updated. Matthew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] talking caller ID
Hi all, Lets say I have my incoming calls transfered to my mobile phone. When a call comes in, Asterisk will answer the call and ask the caller to hold the line while the call is being transfered. I know how to do this, but i dont want the caller to hear me answer the mobile phone. They can hear some music on hold. When I answer Asterisk will read the callerID to me and I can then decide if this call is important or not. If I press one on the mobile phone it will be connected, other wise it will be transfered to my voicemail. I think this can be done through some macro, but not sure how to do this. All the best and thanks, Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: One-Way-Audio After placing call on hold
Seems like it is the IAX jitterbuffer. Can anyone offer any insight as to why? If I turn jitterbuffer=no or disabled (comment it) then my one way audio after hold issue goes away. On 11/8/06, Matt [EMAIL PROTECTED] wrote: My iax.conf is: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) delayreject=yes disallow=all allow=ulaw allow=gsm jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=100 jittershrinkrate=1 notransfer=yes trunk=no [zoot] type=user secret= auth=plaintext host=zoot.xx.net notransfer=yes context=from-trunk On 11/8/06, Matt [EMAIL PROTECTED] wrote: I (and some others) are having an issue with placing calls on hold. Our setup is as follows: IAX2 or SIP terminator/originator --- asterisk box --- SIP Phones I have tried asterisk verison 1.0.9, 1.2.6 and 1.2.12, and all have the same issue. When I place a call on hold that has come in a PSTN channel (through a PRI and a Digium card) everything is fine. When I place a call on hold that has come in or gone out an IAX2 or SIP terminator or originator, when I pick the call back up often there is one-way-audio (I can hear the caller, but the caller can not hear me). I have attached a packet capture of the situation, and as you can see at about packet #3803 the audio goes one way. Can anyone enlighten me as to why this is happening, and why asterisk is no longer sending audio back to the terminator? I would like to get this fixed, obviously. (This file was a tcpdump, and can be opened in ethereal/wireshark) EDIT: Packet capture at following address: http://hecate.chilitech.net/~matth/zootdump.cap ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Solaris
Has anybody tried running Asterisk on Solaris on a SUN SparcStation ? Or maybe the alternative of running Asterisk on a Linux Distro on a SUN SparcStation? I am asked to do this but I think it's almost impossible work to make it happen. Regards, Jorge A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Microsoft will enter VoIP market in earnestnextyear, says Ballmer
On 11:06, Wed 08 Nov 06, Gary G. Hendershot wrote: I think that all the individual pieces of what I saw in this slide show are available NOW as open source components ... it looks to me like the big challenge here is integrating all the pieces so they all work together to provide the desired functionality ... am I wrong about this analysis ??? Have a look at Covide. http://www.covide.net It comes close and is an opensource all-in-one deal. The functionality to tell wether a mail is email or voicemail is a nice candidate for a FR for covide. Greetz -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Corruption Problem
Asterisk People, I'm currently using Asterisk and with a SIP voip provider and I'm having problems where DTMF input in my IVR app is getting corrupted intermittently. For example, if someone enters 1025, it may come though correctly as 1025, or it may come trough as 10025, or 100255. DTMF digits will just double up. This doesn't happen all the time. Asterisk will just pick times to not be very friendly with DTMF, and other times it will just work flawlessly. I'm using RFC2833 on: Linux hostname 2.6.9-42.0.2.ELsmp #1 SMP Wed Aug 23 00:17:26 CDT 2006 i686 i686 i386 GNU/Linux with Asterisk 1.2.13. Also, I am not using a zaptel timer. Could this possibly be causing problems with DTMF?? Thanks! -- Justin Tunney ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk and Max TNT PRI to SIP Authentication Issue, a little closer
After mocking up an unauthenticated call from a different device, a spa942 phone, I found something strange in the SIP debug between the phone and the TNT. Asterisk is accepting unauthenticated calls as long as there is not a user in the SIP header from the calling device. Invite from the MAX: does not get passed to the dial plan -- SIP read from 10.10.14.131:5060: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 To: sip:[EMAIL PROTECTED]:5060;user=phone From: NO CID NAME sip:[EMAIL PROTECTED]:5060;user=phone;tag=1e82fc7f-1fb33c15-830e0a0a Remote-Party-Id: NO CID NAME sip:[EMAIL PROTECTED]:5060;user=phone;screen=no;id-type=subscriber;party=calling;privacy=off Call-ID: [EMAIL PROTECTED] CSeq: 803597 INVITE Via: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK007aa1ced4ace55a Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5060;user=phone Supported: replaces Content-Type: application/sdp Accept: application/sdp Accept-Encoding: Accept-Language: en User-Agent: Lucent-Universal-Gateway Content-Length: 232 Invite from the phone: gets passed to the dial plan in the [general] context= -- SIP read from 10.10.11.51:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.10.11.51:5060;branch=z9hG4bK-9fd7c0a9 From: 2001 sip:[EMAIL PROTECTED];tag=59f6242028d88691o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: 2001 sip:[EMAIL PROTECTED]:5060 Expires: 240 User-Agent: Linksys/SPA942-4.1.12(a) Content-Length: 391 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp The invite string from the TNT: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 The invite string from the phone: INVITE sip:[EMAIL PROTECTED] SIP/2.0 It appears that if a user= field is in the invite message, Asterisk looks for a user context and requires authentication. So the insecure=port,invite option should also include an insecure=user option to disregard any user info in the invite. Is there is another mechanism in Asterisk to disregard any user info from an invite? Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Microsoft will enter VoIP market in earnest
Thanks Curt, thats too cool for school, any idea on when this is coming to the MS SBS platform?I use SBS for myself at home and would love that level of functionality included.Does Asterisk therefore handoff voicemail storage etc to Exchange for this level of integration?Cheers,DeanFrom: Curt Shaffer [mailto:cshaffer at gmail.com] Sent: Tuesday, November 07, 2006 11:08 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] Microsoft will enter VoIP market inearnestnextyear, says BallmerTake a look at OVA.. http://wm.microsoft.com/ms/exchange/2007/Phone_Based_User_Experience_With_Outlook_Voice_Access_300k.wmv ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Corruption Problem
Justin Tunney wrote: Asterisk People, I'm currently using Asterisk and with a SIP voip provider and I'm having problems where DTMF input in my IVR app is getting corrupted intermittently. For example, if someone enters 1025, it may come though correctly as 1025, or it may come trough as 10025, or 100255. DTMF digits will just double up. This doesn't happen all the time. Asterisk will just pick times to not be very friendly with DTMF, and other times it will just work flawlessly. I'm using RFC2833 on: Linux hostname 2.6.9-42.0.2.ELsmp #1 SMP Wed Aug 23 00:17:26 CDT 2006 i686 i686 i386 GNU/Linux with Asterisk 1.2.13. Also, I am not using a zaptel timer. Could this possibly be causing problems with DTMF?? Thanks! -- Justin Tunney Justin, Have you tried 1.4 with vldtmf? -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Warning: Channel does not have a CDR when doing ForkCDR
Gang,
I'm having this error pop up when I do a ForkCDR, and I'm not sure how
to get around it. Here are a few log lines:
Nov 8 10:37:08 VERBOSE[28079] logger.c: -- Executing
ForkCDR(Zap/49-1, ) in new stack
Nov 8 10:37:08 WARNING[28079] app_forkcdr.c: Channel does not have a
CDR
The scenario occurs like this:
I use a .call file to generate a call on Zap/1-1
The called party is then bridged to Zap/49-1
After the bridge occurs, I would like a separate CDR to reflect a
successful bridge (or transfer or whatever we call it when two Zap
channels are connected)
From the error message it sounds like Zap/49-1 doesn't have a CDR to
begin with - is it possible to force the second leg, i.e. the call on
Zap/49-1, to generate a CDR?
Any help would be appreciated. Dial plan info is at the end of this
transmission.
Thanks,
MC
Dial plan info:
This is what I affectionately call blasterisk - I'm using Asterisk to
make lots of automated calls. I'm not really blasting away, but it
sounded cool. Anyway, here's how it works-
I start with a .call file that generates a phone call in
blasterisk_dialout,s,1
If the called party presses the correct digits, in this case 1, then the
call goes to blasterisk_english_right_party,s,1.
If the called party then dials 9, he is transferred to an agent, which
is where the macro Connect_to_agent comes in to play.
The agent is called on a separate zap channel and then presses 1 to
accept the call. Upon a successful connection to an agent I'd like to
generate a new CDR entry, which is why I'm doing the ForkCDR...
[blasterisk_dialout]
exten = s,1,DigitTimeout,1
exten = s,n,ResponseTimeout,10
exten = s,n,Answer
exten = s,n,AMD
exten = s,n,Noop(AMDSTATUS is '${AMDSTATUS')
exten = s,n,GotoIf($[${AMDSTATUS} = AMD_MACHINE]?lmtc,s,1:human)
exten = s,n(human),Set(NUMTRIES=1)
exten = s,n,SetCDRUserField(${dnum})
exten = s,n,AppendCDRUserField(:${cdn})
exten = s,n,AppendCDRUserField(:${dialednum})
exten = s,n(repeat),Background(Initial-greeting)
exten = s,n,Wait(.1)
exten = s,n,Flite(${fname})
exten = s,n,Flite(${lname})
exten = s,n,Background(If-this-person-press-1-else-press-2)
exten = s,n,Set(NUMTRIES=$[${NUMTRIES}+1])
exten = s,n,GotoIf($[${NUMTRIES} 2]?repeat)
exten = s,n,Goto(t,1)
exten = 1,1,Goto(blasterisk_english_right_party,s,1)
exten = 2,1,Goto(blasterisk_english_message,s,1)
exten = 5,1,Goto(blasterisk_spanish_main_greeting,s,1)
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup
[blasterisk_english_right_party]
exten = s,1,Answer
exten = s,n,Background(this-is-a-call-from-fcn-regarding-ref-num)
exten = s,n,WaitExten(.1)
exten = s,n,SayDigits(${dnum})
exten = s,n,WaitExten(.1)
exten = s,n,Background(to-speak-to-csr-press-9)
exten = s,n,Background(silence/1)
exten = s,n,Background(this-is-a-call-from-fcn-regarding-ref-num)
exten = s,n,WaitExten(.1)
exten = s,n,SayDigits(${dnum})
exten = s,n,WaitExten(.1)
exten = s,n,Background(to-speak-to-csr-press-9)
exten = s,n,Background(silence/5)
exten = s,n,Background(Not-right-party-live-Eng)
exten = s,n,SayDigits(${dnum})
exten = s,n,WaitExten(5)
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup
exten = t,1,ForkCDR
exten = t,n,Playback(vm-goodbye)
exten = t,n,Hangup
exten = 9,1,Playback(pls-hold-while-try)
exten = 9,n,Noop(Attempting to bridge to ${agentext})
exten = 9,n,Dial(Zap/g9/${agentext}|60|M(Connect_to_agent^${dnum}))
exten = 9,n,Noop(Done w/ x-fer to agent!)
exten = 9,n,Hangup
;exten = 9,n,Noop(Done with dialing, now hoping for the best)
exten = 9,h,ForkCDR
exten = 9,h,Hangup
exten = 9,103,Playback(im-sorry-unable-to-connect-to-eng)
exten = 9,104,Playback(vm-goodbye)
exten = 9,105,Hangup
[macro-Connect_to_agent]
exten = s,1,DigitTimeout(180)
exten = s,n,Noop(Inside macro, ARG1 is '${ARG1}') ;ARG1 = dnum
exten = s,n,Set(AGENT_TRIES=1)
exten = s,n,Noop(Agent tries = ${AGENT_TRIES})
exten = s,n(repeat),Wait(.1)
exten = s,n,Playback(your-account)
exten = s,n,Playback(number)
exten = s,n,Wait(.4)
exten = s,n,SayDigits(${ARG1})
exten = s,n,Read(ACCEPT|silence/5|1|noanswer|1|5)
exten = s,n,Noop(ACCEPT is ${ACCEPT})
exten = s,n,Set(AGENT_TRIES=$[${AGENT_TRIES} + 1])
exten = s,n,GotoIf($[${ACCEPT} 8]?agent:check)
exten = s,n(check),GotoIf($[${AGENT_TRIES} 40]?10:repeat)
exten = s,n,Noop(How did I get here?!)
exten = s,n(agent),Noop(Agent pressed ${ACCEPT} - call being
transferred)
exten = s,n,Set(ACCTDATA=${CDR(userfield)})
exten = s,n,ForkCDR() ; start new CDR if call actually got xfer'd
exten = s,n,SetCDRUserField(${ACCTDATA})
exten = s,n,AppendCDRUserField(:XFER_TO_AGENT)
exten = s,n,Noop(All done here!)
exten = t,1,Noop(Agent timeout, dropping call to queue)
exten = h,1,Hangup
exten = i,1,Noop(Invalid entry, dropping call to queue)
exten = 10,1,Playback(vm-goodbye)
exten = 10,n,Noop(Agent did not pickup call...)
exten = 10,n,Hangup
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RE: [asterisk-users] Microsoft will enter VoIP market in earnest
I do not know when they plan on SBS deployment of this. I wouldnt imagine it would not be soon because they just released 2003 R2. The biggest hurdle to this working with Asterisk from what I understand is that it requires SIP over TCP. I havent read the docs fully for 1.4 version of Asterisk is going to support that or not. I am not sure on the storage of the VM either. I would imagine if its not held by Exchange that Exchange will need some kind of rights to the VM server to add/remove/modify/forward VM messages. I have a beta version of it but I just do not have time to install it at the moment. I will be happy to post my results once I do get the time though J Curt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Wednesday, November 08, 2006 2:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Microsoft will enter VoIP market in earnest Thanks Curt, thats too cool for school, any idea on when this is coming to the MS SBS platform?I use SBS for myself at home and would love that level of functionality included.Does Asterisk therefore handoff voicemail storage etc to Exchange for this level of integration?Cheers,DeanFrom: Curt Shaffer [mailto:cshaffer at gmail.com] Sent: Tuesday, November 07, 2006 11:08 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] Microsoft will enter VoIP market inearnestnextyear, says BallmerTake a look at OVA.. http://wm.microsoft.com/ms/exchange/2007/Phone_Based_User_Experience_With_Outlook_Voice_Access_300k.wmv ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Corruption Problem
Migrating to 1.4 is not an option. I don't know what that is, but I doubt my voip provider supports it. On 11/8/06, Kristian Kielhofner [EMAIL PROTECTED] wrote: Have you tried 1.4 with vldtmf? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Off-Site Extensions That Would Show As In-Use?
Hello, list! I'd like to create an extension that points to an offsite location (a number on the PSTN), the purpose of which would be to see if that offsite location is still on a call forwarded to it by Asterisk. This way a receptionist could choose to transfer calls to a mobile phone only if it's finished with the last call the receptionist forwarded to it. If I configure a custom extension with the destination SIP/TrunkName/NXXNXX, the calls transfer fine but don't show as busy using the Flash Operator Panel (as an example). Any thoughts? Thanks in advance, Alex -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Solaris
Have a look at http://www.solarisvoip.comOn 11/8/06, Jorge Alayon [EMAIL PROTECTED] wrote:Has anybody tried running Asterisk on Solaris on a SUN SparcStation ? Or maybe the alternative of running Asterisk on a Linux Distro on a SUNSparcStation?I am asked to do this but I think it's almost impossible work to make ithappen.Regards,Jorge A.___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Operating queues with clients on a legacy PABX
What is your queues.conf? Can you dial the user outside of a queue after they transfer the call?On 11/8/06, Rob Hillis [EMAIL PROTECTED] wrote:Hi guys!I'm having one or two issues with queues hosted by an Asterisk machine where the clients are on a legacy PABX - at least for the interim.Ifully expect most of these issues to be non-resolvable, but thought I'dat least ask to find out if there is some way of working around the issues.The legacy PABX is an NEC 7400 ICS connected to Asterisk via anE1 ISDN link.Calls are passed to the NEC without a problem.The biggest issue is when an agent transfers a call to another person on the NEC.Obviously using the transfer button on the phones givesAsterisk no clue that the call has been transferred, meaning that theagent then does not receive another call until the transferred call hasbeen completed.Can anyone think of a workaround for this? ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Re: Echo Issues
I am now scheduled to replace this with a TE412P with the VPM450M EC module. Thanks for the heads up. -- -- Steven http://www.glimasoutheast.org Steven [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Mine is a VPM400 on a TE410P (2nd Gen) Purchased as a TE411P -- -- Steven http://www.glimasoutheast.org Matthew Fredrickson [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] On Nov 6, 2006, at 8:06 AM, Steven wrote: Matt, How does one check for this?? You would probably know from the dmesg output card, just make sure it's using the Octasic based echo canceler. I think it says something about a VPM450M in the dmesg logs if it's the version I'm thinking of. If it's not, talk to RMA and see if you can get it updated. Matthew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sms script on receive
the documentation for the sms app mentions a script that sends on the sms by e-mail once it's arrived
exten = _XX/_8005875290,n,System(/usr/lib/asterisk/smsin ${EXTEN:3})
Does anyone have any samples of such a script (it might be a good addition to the documentation) or does anyone have another way to send the sms on by e-mail.
Steve
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[asterisk-users] [FC5] How to update kernel/kernel-develop for Athlon?
Hello I'm following instructions on how to install Asterisk on Fedora 5, but I'm having a problem: - the host is an older i686 athlon i386 GNU/Linux - /etc/rpm/platform says athlon-redhat-linux - running yum update kernel downloaded kernel i686 2.6.18-1.2200.fc5 - running yum update kernel-devel wants to download kernel-devel i586 2.6.18-1.2200.fc5 I know that I shouldn't mix versions (i686 and i586), but I don't know how else to update the system to make it ready for Asterisk. = Should I use a specific repository for Yum to use, or should I download a couple of RPMs to update those two items before proceeding with Asterisk? Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] International dialing with GPX-2000 and early dial
Early dial is a feature on the phone that makes use of the 484 (Address
Incomplete) response.
This is desired for in-office, local (PSTN), and long distance dialing.
I'm really hoping to find a best-of-both-worlds solution to this.
Andrew Joakimsen wrote:
Does the GXP-2000 not have its own dialplan? Use that and disable
early dial
On 11/3/06, *Anthony Kepler* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I am trying to allow users to place outgoing international calls
from a
GPX-2000 with early dial enabled, connected to Asterisk 1.2.12.1
http://1.2.12.1
I have the following extension line:
exten = _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
When I attempt to place a call to a number in, for instance, Kenya, I
dial 011254...etc.
and I get this on the asterisk console:
Executing Dial(SIP/1001-081fb718, Zap/g1/0112) in new stack
-- Called g1/0112
It is attempting to dial out as soon as it receives a single digit to
represent the .
What I need is for it to wait a reasonable amount of time for
additional
digits.
I have tried using set(TIMEOUT(digit)=5), and I see the following
in the
asterisk console:
-- Executing Set(SIP/1001-081fb718, TIMEOUT(digit)=5) in
new stack
-- Digit timeout set to 5
However, this is printed far less than 5 seconds before the dial out
attempt.
I assume there must be something relatively obvious I'm missing
here...
if anyone can shed some light on this, it would be greatly
appreciated.
Thank you,
- Anthony Kepler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] |
SIP/Email
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[asterisk-users] I LOVE IT
After about one weeks time I've gone from no VoIP to a completely configured system for two of our offices to be able to page/communicate interoffice as well as handle existing PSTN communications (okay, waiting onf hardware for the PSTN side and I've likely jinxed myself now). I was sweating getting the two boxes talking to each other and I knocked that out in no time without even needing to look up online, FreePBX makes it to easy. Once my hardphones TDM400's get here hopefully by the end of this week I'll be in for full blown testing and rapid deployment there after. Props to all developers involved in Asterisk/FreePBX and everything in between. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Still problems with Asterisk on latest Debian
Hi all, I have now reinstalled my whole system because I had to change a few things wiht my drives. Here is what happens. I have done apt-get build-dep asterisk apt-get install linux-headers-2.6.17-2-686 which works just fine now. Downloaded the latest files from digiums ftp. First I unpacked zaptel. I am doing everything as root. Then I just type make. Here is what happens: checking for gcc... gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether gcc accepts -g... yes checking for gcc option to accept ISO C89... none needed checking how to run the C preprocessor... gcc -E checking for a BSD-compatible install... /usr/bin/install -c checking whether ln -s works... yes checking for GNU make... make checking for grep... /bin/grep checking for sh... /bin/sh checking for ln... /bin/ln checking for grep that handles long lines and -e... (cached) /bin/grep checking for egrep... /bin/grep -E checking for ANSI C header files... yes checking for sys/types.h... yes checking for sys/stat.h... yes checking for stdlib.h... yes checking for string.h... yes checking for memory.h... yes checking for strings.h... yes checking for inttypes.h... yes checking for stdint.h... yes checking for unistd.h... yes checking for initscr in -lcurses... yes checking curses.h usability... yes checking curses.h presence... yes checking for curses.h... yes checking for initscr in -lncurses... yes checking for curses.h... (cached) yes checking for newtBell in -lnewt... yes checking newt.h usability... yes checking newt.h presence... yes checking for newt.h... yes configure: creating ./config.status config.status: creating build_tools/menuselect-deps config.status: creating makeopts configure: *** Zaptel build successfully configured *** The configure script was just executed, so 'make' needs to be restarted. make: *** [config.status] error 1 Then I type make again and it seem to work fine. I have that output as well, but i can send that if someone is interested. Then I type make install and the following happens: make[1]: Entering directory `/root/zaptel-1.4.0-beta2' make -C /lib/modules/2.6.17-2-686/build SUBDIRS=/root/zaptel-1.4.0-beta2 modules make[2]: Entering directory `/usr/src/linux-headers-2.6.17-2-686' Building modules, stage 2. MODPOST WARNING: /root/zaptel-1.4.0-beta2/xpp/xpd_fxs: 'slic_cmd_direct_write' exported twice. Previous export was in /root/zaptel-1.4.0-beta2/xpp/xpd_fxo.ko WARNING: /root/zaptel-1.4.0-beta2/xpp/xpd_fxs: 'slic_cmd_direct_read' exported twice. Previous export was in /root/zaptel-1.4.0-beta2/xpp/xpd_fxo.ko WARNING: /root/zaptel-1.4.0-beta2/xpp/xpd_fxs: 'slic_cmd_indirect_write' exported twice. Previous export was in /root/zaptel-1.4.0-beta2/xpp/xpd_fxo.ko WARNING: /root/zaptel-1.4.0-beta2/xpp/xpd_fxs: 'slic_cmd_indirect_read' exported twice. Previous export was in /root/zaptel-1.4.0-beta2/xpp/xpd_fxo.ko WARNING: /root/zaptel-1.4.0-beta2/xpp/xpd_fxs: 'dump_slic_cmd' exported twice. Previous export was in /root/zaptel-1.4.0-beta2/xpp/xpd_fxo.ko make[2]: Leaving directory `/usr/src/linux-headers-2.6.17-2-686' make[1]: Leaving directory `/root/zaptel-1.4.0-beta2' build_tools/genudevrules /etc/udev/rules.d/zaptel.rules if [ -d /usr/lib/hotplug/firmware ]; then \ /usr/bin/install -c -m 644 wct4xxp/*.ima /usr/lib/hotplug/firmware; \ fi if [ -d /lib/firmware ]; then \ /usr/bin/install -c -m 644 wct4xxp/*.ima /lib/firmware; \ fi Installed firmware /usr/bin/install -c -D -m 755 libtonezone.a /usr/lib/libtonezone.a /usr/bin/install -c -D -m 755 libtonezone.so /usr/lib/libtonezone.so.1.0 if [ -z -a `id -u` = 0 ]; then \ /sbin/ldconfig || : ;\ fi rm -f /usr/liblibtonezone.so /bin/ln -sf libtonezone.so.1.0 \ /usr/lib/libtonezone.so.1 /bin/ln -sf libtonezone.so.1.0 \ /usr/lib/libtonezone.so if [ -z -x /usr/sbin/sestatus ] (/usr/sbin/sestatus | grep SELinux status: | grep -q enabled) ; then restorecon -v /usr/lib/libtonezone.so; fi /bin/sh: line 0: [: saknar ] /usr/bin/install -c -D -m 644 zaptel.h /usr/include/zaptel/zaptel.h /usr/bin/install -c -D -m 644 tonezone.h /usr/include/zaptel/tonezone.h /usr/bin/install: Cannnot create normal file /usr/include/zaptel/tonezone.h: File or directory does not exist. make: *** [install-include] Error 1 I am using the swedish language so had to translate a little of the above message. Any help would be apreciated! many thanks, Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reg errors? Other anomalies? Check those capacitors!
Three months ago, I was experiencing all sorts of issues with my Asterisk box maintaining a connection to multiple trunks, etc. I also experienced various timing issues as well. In addition, Asterisk would sometimes take almost a minute to fully load and register its SIP and IAX trunks. Puzzled, I recompiled several times. No result. I checked my hardware. Didn't find anything. However, I did overlook one thing:* The motherboard's capacitor!Yep, you guessed it! It was bad. Now, I do not have any problems (I didn't bother replacing the motherboard, ended up using a spare PC). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reg errors? Other anomalies? Check those capacitors!
The motherboard's capacitor? What is that? Since there are probably a hundred or more caps on the MB, how did you determine that? Was it burned? Other than that, without making either capacitance or noise tests I can't imagine how you would make that assumption. Doug On Wed, 8 Nov 2006, Ronald Lewis wrote: Three months ago, I was experiencing all sorts of issues with my Asterisk box maintaining a connection to multiple trunks, etc. I also experienced various timing issues as well. In addition, Asterisk would sometimes take almost a minute to fully load and register its SIP and IAX trunks. Puzzled, I recompiled several times. No result. I checked my hardware. Didn't find anything. However, I did overlook one thing: * The motherboard's capacitor! Yep, you guessed it! It was bad. Now, I do not have any problems (I didn't bother replacing the motherboard, ended up using a spare PC). Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queues: member order vs. defines in queues.conf
Hi, I'm still pulling my hair out getting my queues setup in 1.2.13. I went in to implement my custom roundrobinreset strategy (mentioned in a post by me here: http://lists.digium.com/pipermail/asterisk-users/2006-October/170713.html and a similar issue is addressed by the developers back in May: http://lists.digium.com/pipermail/asterisk-dev/2006-May/020916.html and I got it to basically work, that is do what people either claim that roundrobin will do *or* that some unspecified strategy will do if one uses penalties on members (note, I and at least a few other people on the list have attempted to make that approach work with no success -- so either it never worked, doesn't work in 1.2.13 or requires some extra configuration that we all missed). There also appears to be some minor (very annoying until one discovers the workaround) bug w/ making changes to members in a queue and then trying to use 'reload' or 'reload app_queue.so' to redefine the queues. If all you're doing is adding a new member, that new member seems to get stuck on the *front* of the queue (as the first avail. member) regardless of its actual position in queues.conf. The only workaround I've discovered is to totally restart Asterisk. I'm pretty sure that used to work fine. Probably has something to do with the new dynamic queue member/agent adding features. On a related note, it looks like the order of the members is reversed vs. how it's defined in queues.conf. I'm pretty sure that was *not* the case in earlier versions (at least in 0.9.x anyway). Anyone else note these (or better yet, care about or have any answers to the questions raised in my earlier posts?) Thanks, John Lawler ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ask users.conf
Hi Alls,In Asterisk-1.4 there is new config file, users.conf, but i don't know how mechanism between users.conf and sip/iax.conf, usually i add new user in sip.conf, but when i try use asterisk-gui, it write to users.conf and when i type sip list peer on asterisk console, there is no user that i create with asterisk-gui. Please give me some explanation coz i am newbie..Thanks-- Regards,mrdlnf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto record a call?
I have a debugging scenario where I wish to record the entire call. The call is establish via a .call file. I can't seem to get Monitor to do anything. My dialplan looks like this: [dialout] exten = s,1,DigitTimeout,1 exten = s,n,ResponseTimeout,10 exten = s,n,Answer exten = s,n,Monitor(wav,/tmp/test) . . . The file test.wav never shows up. Am I doing something wrong, or possibly there is a better way to accomplish this? Thanks, MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Reg errors? Other anomalies? Check thosecapacitors!
Always take your wedding ring off when working inside the box!! -- -- Steven http://www.glimasoutheast.org Doug Crompton [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] The motherboard's capacitor? What is that? Since there are probably a hundred or more caps on the MB, how did you determine that? Was it burned? Other than that, without making either capacitance or noise tests I can't imagine how you would make that assumption. Doug On Wed, 8 Nov 2006, Ronald Lewis wrote: Three months ago, I was experiencing all sorts of issues with my Asterisk box maintaining a connection to multiple trunks, etc. I also experienced various timing issues as well. In addition, Asterisk would sometimes take almost a minute to fully load and register its SIP and IAX trunks. Puzzled, I recompiled several times. No result. I checked my hardware. Didn't find anything. However, I did overlook one thing: * The motherboard's capacitor! Yep, you guessed it! It was bad. Now, I do not have any problems (I didn't bother replacing the motherboard, ended up using a spare PC). Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton* * Richboro, PA 18954* * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unknown caller id problem
Hi
I have a * box with TE110P. When call comes in via ISDN without caller
id information, asterisk sets the caller id as Unknown. Is there any
way to change this? I've tried below but only works for calls with
caller id.
$AGI-set_callerid('74442932');
Thanks
Jay
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[asterisk-users] DID billing with a2billing
Can anyone tell me what I have to do to get DID billing to word with a2billing. I am thing it may be context -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: I need (some) help in configuring PAP2.
On 2006-11-08 06:26:45 -0800, [EMAIL PROTECTED] said: Hello, I need (some) help in configuring PAP2. Try looking in sip.conf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] talking caller ID
Christian wrote:<[EMAIL PROTECTED]>From: "Christian" [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Wed, 08 Nov 2006 20:10:02 +0100Subject: [asterisk-users] talking caller ID Hi all,Lets say I have my incoming calls transfered to my mobile phone. When a call comes in, Asterisk will answer the call and ask the caller to hold the line while the call is being transfered.I know how to do this, but i dont want the caller to hear me answer the mobile phone. They can hear some music on hold. When I answer Asterisk will read the callerID to me and I can then decide if this call is important or not. If I press one on the mobile phone it will be connected, other wise it will be transfered to my voicemail. I think this can be done through some macro, but not sure how to do this.All the best and
thanks,ChristianChristian, this would be fairly straight forward. Take a look at the follow-me / ringgroup implementation of freepbx 2.2 (currently beta 2). It does call confirmation almost as you describe ('you have an incoming call, press 1 to accept, 2 to reject) while the phone rings or plays music to the caller. It would be rather trivial to tweak the dialplan that plays that message and play the callerid of the incoming call. p
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Re: [asterisk-users] DTMF Corruption Problem
Hey, Justin: Justin Tunney wrote: Asterisk People, I'm currently using Asterisk and with a SIP voip provider and I'm having problems where DTMF input in my IVR app is getting corrupted intermittently. For example, if someone enters 1025, it may come though correctly as 1025, or it may come trough as 10025, or 100255. DTMF digits will just double up. This doesn't happen all the time. Asterisk will just pick times to not be very friendly with DTMF, and other times it will just work flawlessly. I'm having virtually the same problem with Asterisk -- the SendDTMF() function. In my case, I hear the first digit, then none of the other digits -- or only some. It's been suggested that it's a channel issue, but I'm having the problem with an IAX channel, and if so, it would mean that there is DTMF issue in both the SIP and IAX channel drivers. That seems pretty unlikely. As in your case, it is highly inconsistent. Depending on when it is run, I'll hear different digits at different durations. Whatever the case, it's not usable. (DTMF grief seems to be painfully common with Asterisk.) It would help to get additional feedback from other users. I've posted a bug. The bug URL is: http://bugs.digium.com/view.php?id=8293 -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
