Re: [asterisk-users] Question about Presence and IM feature
15 jan 2010 kl. 08.23 skrev Yuji Kondo: I have two questions for Asterisk feature. 1. Can Asterisk support presence feature ? Asterisk is a telephony PBX and supports presence subscriptions for extension states - if a phone line is busy or not, over a few different SIP presence formats, like SIP dialog-info and SIMPLE. Is latest version supporting PUBLISH method No. 2. Can Asterisk provide Instant Messaging feature as server ? During a phone call, yes. Not otherwise. If you need a full-blown presence server, you should look at Kamailio/OpenSER at http://www.kamailio.org Best regards, /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginners Guide to setting up a Call Centre
Ok this has Probably been put to bed several time but never mind. Elastix, Trixbox, or AsterixNow, or DIY (ie Ubuntu or whatever installed with OpenPBX, Asterix etc by hand) I've got a new server to run Asterix on and want to get it working quickly and yet be configurable in the future with out having to reisntall and start again regally. Currently no VoIP hardware but that will come once I prove the concept. I guess Oh the machine does not have a CD Rom Drive so a network/USB install would be nice.. But I guess I can open the case and plug one in for installation if I must! (Says he who has just installed Ubuntu over the network to check the computer works!) Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing issue
Ishfaq Malik wrote:
Ishfaq Malik wrote:
Hi
We run a hosted VoIP service for multiple customers off the same server
and I'm having an odd issue with just one customer in particular. We're
using realtime in a MySQL DB and this is their dialplan
*** 1. row ***
context: pcsu-Identifier
exten: s
priority: 1
app: Answer
appdata:
*** 2. row ***
context: pcsu-Identifier
exten: s
priority: 2
app: Wait
appdata: 2
*** 3. row ***
context: pcsu-Identifier
exten: s
priority: 3
app: Set
appdata: CALLERID(num)=${CALLERID(num)}
*** 4. row ***
context: pcsu-Identifier
exten: s
priority: 4
app: GotoIfTime
appdata: 08:30-17:30|mon-fri|*|*?pcsu-Identifier-work|s|1
*** 5. row ***
context: pcsu-Identifier
exten: s
priority: 5
app: Playback
appdata: pcsu-voicemail-file
*** 6. row ***
context: pcsu-Identifier
exten: s
priority: 6
app: Voicemail
appdata: 2...@pcsu-local|s
*** 7. row ***
context: pcsu-Identifier
exten: s
priority: 8
app: Hangup
appdata:
*** 8. row ***
context: pcsu-Identifier-work
exten: s
priority: 1
app: Dial
appdata:
SIP/ukgeonum...@carrierSIP/ukgeonum...@carrierSIP/PCSU200SIP/PCSU201SIP/PCSU202SIP/PCSU203SIP/PCSU204SIP/PCSU205SIP/PCSU206|15
*** 9. row ***
context: pcsu-Identifier-work
exten: s
priority: 2
app: Dial
appdata:
SIP/ukgeonum...@carrierSIP/ukgeonum...@carrierSIP/ukgeonum...@carrierSIP/PCSU200SIP/PCSU201SIP/PCSU202SIP/PCSU203SIP/PCSU204SIP/PCSU205SIP/PCSU206|20
*** 10. row ***
context: pcsu-Identifier-work
exten: s
priority: 3
app: Playback
appdata: pcsu-voicemail-file
*** 11. row ***
context: pcsu-Identifier-work
exten: s
priority: 4
app: Voicemail
appdata: 2...@pcsu-local|s
*** 12. row ***
context: pcsu-Identifier-work
exten: s
priority: 5
app: Hangup
appdata:
I know how daft it looks but they insisted on ringing real UK geographic
numbers in the same step as SIP extensions. A while back I changed the
initial Answer step to NoOp as the Answer step was distorting our CDR
and I hadn't realised that Answer wasn't implicitly required. After I
did this the caller stopped hearing a ringing tone when ringing into
this dial plan. When I put the Answer step back in instead of the NoOp
the caller could hear the ringing tone when dialling in again.
I've tried replacing the Answer with Ringing but I still got silence
while the extensions and numbers were ringing.
Any thoughts on this would be helpful and I will be trying to replicate
this on out test system.
Thanks in advance
Ish
I should also add, we have no problems with the caller hearing ringing
with any of the other dial plans on this server even though they start
with NoOp and not Answer
Ish
Fixed it by using an explicit r option in the dial steps
Ish
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Software Developer
PackNet Ltd
Office: 0161 660 3062
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[asterisk-users] Realtime queue not work
hi, all
i try to confiture realtime queue, but not work, details as below:
Insert into queue_table(name)value('95040654321');
INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun',
'95040654321', 'SIP/1001', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321',
'SIP/1002', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Zhang Jianming',
'95040654321', 'SIP/1003', 2, 1);
but when i dial 95040654321 and press extension 1. error happens:
-- Executing Queue(SIP/1003-, 950406543211)
[Jan 15 03:18:57] WARNING[16626]: app_queue.c:4223 queue_exec: Unable
to join queue '950406543211'
== Auto fallthrough, channel 'SIP/1003-' status is 'UNKNOWN'
DO you know what's wrong?
Thank you!
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Sucan
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Re: [asterisk-users] how to strip + from the caller-ID
Szasz Szabolcs wrote:
Hi,
How can I strip + from the front of the caller ID?
I have tried this:
exten = s/_+X.,1,Set(CALLERID(name)=${CALLERID(name):1})
Hi,
what about:
exten = _+[1-9].,1,Dial(Local/${EXTEN:1...@context
to proceed your call without leading + in your dialplan?
Cheers
Joern
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Re: [asterisk-users] Fax Detection on SIP
- Paul Scott p...@cpanel.net wrote: Yeah sounds like you wanna use NVFaxDetect it would allow you to add something like exten = fax,1,Swift(number has changed); to your inbound call part of your dialplan On Jan 14, 2010, at 11:45 AM, Juan C. Villa wrote: Could you use NVFaxDetect? On Thu, 2010-01-14 at 17:35 +, --[ UxBoD ]-- wrote: Hi, We have a issue where one of our clients is receiving a high volume of calls from automated fax machines and passing through their context which means all phones get rung. I am looking for a way to detect the fax tone, on answer, and route it to a extension/macro where a announcement would tell them the correct number to fax to. The inbound calls are across SIP so am not able to use DADHI. Would this scenario be perfect for Fax for Asterisk ? All help greatfully appreciated. The problem is that NVFaxDetect does not appear to work with Asterisk 1.6.X :( Looking for something that is actively supported. Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10/100 voip phones and gigabit connection
On 01/15/10 17:54, randall wrote: hi all, i noticed that a lot of VOIP phones have a double network interface allowing you to use only 1 LAN cable for both the phone and your desktop, a really nice feature that can save a lot of cable, but most are 10/100 connections while i have a gigabit network. Off course there are phones available with a Gigabit connection but these are at least 3 to 4 times as expensive. another option would be to have both desktop and voip phone each a dedicated line ( basically having 2 seperate networks ), already have these in place from the old/current situation but i was hoping to clear some cables. does anybody know of another solution to this or is my conclusion above simply all the choice there is? You've hit the nail on the head. A VoIP phone with two network ports is probably best thought of as a two port switch. Like any switch, if you connect a gigabit NIC to a 10/100 switch, you'll end up with a 100 megabit connection. The only way to get a gigabit connection to your PC is via a phone that has gigabit ports, or have a separate cable back to the switch. Best practice is usually to segregate phone and PC networks anyway - it helps avoid degradation of VoIP quality when the LAN becomes heavily loaded. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Detection on SIP
- --[ UxBoD ]-- ux...@splatnix.net wrote: - Paul Scott p...@cpanel.net wrote: Yeah sounds like you wanna use NVFaxDetect it would allow you to add something like exten = fax,1,Swift(number has changed); to your inbound call part of your dialplan On Jan 14, 2010, at 11:45 AM, Juan C. Villa wrote: Could you use NVFaxDetect? On Thu, 2010-01-14 at 17:35 +, --[ UxBoD ]-- wrote: Hi, We have a issue where one of our clients is receiving a high volume of calls from automated fax machines and passing through their context which means all phones get rung. I am looking for a way to detect the fax tone, on answer, and route it to a extension/macro where a announcement would tell them the correct number to fax to. The inbound calls are across SIP so am not able to use DADHI. Would this scenario be perfect for Fax for Asterisk ? All help greatfully appreciated. The problem is that NVFaxDetect does not appear to work with Asterisk 1.6.X :( Looking for something that is actively supported. Well perhaps I should have read the sip.conf ;) Okay so I am able to set faxdetect=yes and then I presume just create a extension in the target context: exten = fax,1,Play(sorry-wrong-fax-number) exten = fax,n,Hangup() -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] jitterbuffer and PLC
Hi, I have a question about jitterbuffer and PLC.
I use Asterisk 1.6.2.0 and 1.6.0.20 or older.
I use uLaw.
My system map:
=
[ asterisk 2 ] -- # LOSS # -- # A # -- [ asterisk 1 ] -- # B # -- [ X-lite ]
=
I use two asterisk server.
'asterisk 2' do Playback(some voice file)
'asterisk 1' do jitter.
'asterisk 1' and 'asterisk 2' has trunked by sip or iax2.
X-lite call 3003 to asterisk 1,
and asterisk 1 call 3000 to asterisk 2.
On the map,
at LOSS I caused packet loss 5%,
and at A and B I captured packet.
I thought number of packets at B would be same number from asterisk 2.
And I thought number of packets at A would be 95% from asterisk 2.
But the result is different!
At A and B is the same number as 95%.
No PLC ,No interpolations has made.
But on CLI,messages was like that jitter and PLC work right .
Like this:
===
-- Local/3...@extd-651d;1 answered SIP/id-0001
-- fixed jitterbuffer created on channel Local/3...@extd-651d;1
VvvvLvvvLLvv
vvLvvLvvv
vvLvL
L
vLLvvLvvv
vvLvvLvLv
vLL-- Hungup 'IAX2/ip-address:4569-3645'
== Spawn extension (extd, 3000, 2) exited non-zero on 'Local/3...@extd-651d;2'
-- fixed jitterbuffer destroyed on channel Local/3...@extd-651d;1
== Spawn extension (extd, 3003, 2) exited non-zero on 'SIP/id-0001'
===
Why ? why no interpolations ?
when sip trunk and iax trunk,same result.
I don't know how to do. So please help me.
How to do to work correct.
Or Asterisk has not yet have jitter and PLC ,hasn't it?
In 'asterisk 1' ,
==
write on sip.conf = jbenable=yes ,and , jbimpl=adaptive
write on iax.conf = jitterbuffer=yes ,and, trunktimestamps=yes
write on codecs.conf = genericplc = true
and on extensions.conf
when use sip trunk =
---
exten = 3003,1,Dial(Local/3...@extd/nj)
exten = 3000,1,Set(CALLERID(num)=some id)
exten = 3000,2,Dial(SIP/${EXTEN}@some exten in sip.conf,120,T)
exten = 3000,3,Congestion
when use iax trunk =
---
exten = 3003,1,Dial(Local/3...@extd/nj)
exten = 3000,1,Set(CALLERID(num)=some id)
exten = 3000,2,Dial(IAX2/id:pass@asterisk 2 ip-address/${EXTEN},120,T)
exten = 3000,3,Congestion
==
In 'asterisk 2' ,
==
write on extensions.conf
=
---
exten = 3000,1,Answer()
exten = 3000,2,Wait(1)
exten = 3000,3,Playback(aaa)
exten = 3000,4,Congestion
---
==
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Re: [asterisk-users] 10/100 voip phones and gigabit connection
On 01/15/2010 12:41 PM, Rob Hillis wrote: On 01/15/10 17:54, randall wrote: hi all, i noticed that a lot of VOIP phones have a double network interface allowing you to use only 1 LAN cable for both the phone and your desktop, a really nice feature that can save a lot of cable, but most are 10/100 connections while i have a gigabit network. Off course there are phones available with a Gigabit connection but these are at least 3 to 4 times as expensive. another option would be to have both desktop and voip phone each a dedicated line ( basically having 2 seperate networks ), already have these in place from the old/current situation but i was hoping to clear some cables. does anybody know of another solution to this or is my conclusion above simply all the choice there is? You've hit the nail on the head. A VoIP phone with two network ports is probably best thought of as a two port switch. Like any switch, if you connect a gigabit NIC to a 10/100 switch, you'll end up with a 100 megabit connection. The only way to get a gigabit connection to your PC is via a phone that has gigabit ports, or have a separate cable back to the switch. Best practice is usually to segregate phone and PC networks anyway - it helps avoid degradation of VoIP quality when the LAN becomes heavily loaded. i'll folow the best practice then in that case. thanks for the confirmation Rob, Randall -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Inbound South America numbers
Hi, is someone able to provide inbound DID for South America, at least Bolivia, Colombia, Panama and Venezuela. Please contact me of list, thanks Regards -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10/100 voip phones and gigabit connection
- Original Message - From: randall To: asterisk-users@lists.digium.com Sent: Friday, January 15, 2010 7:54 AM Subject: [asterisk-users] 10/100 voip phones and gigabit connection hi all, just subscribed to the list and first mail, nice to be here. Hopefully i'm in the right place for this question since i'm planning a little VOIP implementation at the moment and ran in to something while going through the shopping list. i noticed that a lot of VOIP phones have a double network interface allowing you to use only 1 LAN cable for both the phone and your desktop, a really nice feature that can save a lot of cable, but most are 10/100 connections while i have a gigabit network. Off course there are phones available with a Gigabit connection but these are at least 3 to 4 times as expensive. In a pinch, the cheapest 1Gbit switch I could find is 17 Eur with 5 ports. Leif -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10/100 voip phones and gigabit connection
On 01/15/2010 02:00 PM, Leif Neland wrote: - Original Message - *From:* randall mailto:rand...@songshu.org *To:* asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Sent:* Friday, January 15, 2010 7:54 AM *Subject:* [asterisk-users] 10/100 voip phones and gigabit connection hi all, just subscribed to the list and first mail, nice to be here. Hopefully i'm in the right place for this question since i'm planning a little VOIP implementation at the moment and ran in to something while going through the shopping list. i noticed that a lot of VOIP phones have a double network interface allowing you to use only 1 LAN cable for both the phone and your desktop, a really nice feature that can save a lot of cable, but most are 10/100 connections while i have a gigabit network. Off course there are phones available with a Gigabit connection but these are at least 3 to 4 times as expensive. In a pinch, the cheapest 1Gbit switch I could find is 17 Eur with 5 ports. Leif its not the network switch that i'm worried about, its the build in switch of the phones with the double network card -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime queue not work
- Original Message -
From: Zhang Shukun
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, January 15, 2010 11:48 AM
Subject: [asterisk-users] Realtime queue not work
hi, all
i try to confiture realtime queue, but not work, details as below:
Insert into queue_table(name)value('95040654321');
INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun',
'95040654321', 'SIP/1001', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321',
'SIP/1002', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Zhang Jianming',
'95040654321', 'SIP/1003', 2, 1);
but when i dial 95040654321 and press extension 1. error happens:
-- Executing Queue(SIP/1003-, 950406543211)
[Jan 15 03:18:57] WARNING[16626]: app_queue.c:4223 queue_exec: Unable
to join queue '950406543211'
== Auto fallthrough, channel 'SIP/1003-' status is 'UNKNOWN'
No golden answers, but something to try.
queue names can not be just numbers? I'd try calling the queue q95040654321.
Does show queues show the queue? Don't know if that's supposed to work on
realtime queues.
Leif
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Re: [asterisk-users] 10/100 voip phones and gigabit connection
- Original Message - From: randall To: asterisk-users@lists.digium.com Sent: Friday, January 15, 2010 2:11 PM Subject: Re: [asterisk-users] 10/100 voip phones and gigabit connection On 01/15/2010 02:00 PM, Leif Neland wrote: - Original Message - From: randall To: asterisk-users@lists.digium.com Sent: Friday, January 15, 2010 7:54 AM Subject: [asterisk-users] 10/100 voip phones and gigabit connection list. i noticed that a lot of VOIP phones have a double network interface allowing you to use only 1 LAN cable for both the phone and your desktop, a really nice feature that can save a lot of cable, but most are 10/100 connections while i have a gigabit network. Off course there are phones available with a Gigabit connection but these are at least 3 to 4 times as expensive. In a pinch, the cheapest 1Gbit switch I could find is 17 Eur with 5 ports. Leif its not the network switch that i'm worried about, its the build in switch of the phones with the double network card Sure. My point was just that IF you only got one connection in the wall, its cheaper to get a switch than getting a phone with dual 1Gbit ports. Leif -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime queue not work
2010/1/15 Leif Neland le...@neland.dk:
- Original Message -
From: Zhang Shukun
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, January 15, 2010 11:48 AM
Subject: [asterisk-users] Realtime queue not work
hi, all
i try to confiture realtime queue, but not work, details as below:
Insert into queue_table(name)value('95040654321');
INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun',
'95040654321', 'SIP/1001', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321',
'SIP/1002', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Zhang Jianming',
'95040654321', 'SIP/1003', 2, 1);
but when i dial 95040654321 and press extension 1. error happens:
-- Executing Queue(SIP/1003-, 950406543211)
[Jan 15 03:18:57] WARNING[16626]: app_queue.c:4223 queue_exec: Unable
to join queue '950406543211'
== Auto fallthrough, channel 'SIP/1003-' status is 'UNKNOWN'
No golden answers, but something to try.
queue names can not be just numbers? I'd try calling the queue
q95040654321.
Thank you for reply. i've try a95040654321 as the queue name but not
work either.
there was the same error in the cli.
Does show queues show the queue? Don't know if that's supposed to work on
realtime queues.
Leif
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Sucan
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[asterisk-users] Logs problem of queue_log-mysql
Hello! I'm trying to registers events of queues in /var/log/asterisk/queue_log and Mysql database .I have configured realtime queue_log on MySQL and works well, but /var/log/asterisk/queue_log file is empty, since you're not registering events of queues. Removing extconfig.conf configurations (queue_log = mysql,general), /var/log/asterisk/queue_log works well, events logs on /var/log/asterisk/queue_log . With extconfig.conf configurations no events logs on /var/log/asterisk/queue_log. What happens?? My asterisk version is 1.6.1.11. addons 1.6.1.2 res_mysql.conf [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = userX dbpass = passX dbport = 3306 dbsock = /tmp/mysql.sock -- extconfig.conf [settings] queue_log = mysql,general logger.conf [general] queue_log = yes queue_log_name = queue_log Thanks, Best regards!! Cristian Arguello. __ Información de ESET NOD32 Antivirus, versión de la base de firmas de virus 4765 (20100112) __ ESET NOD32 Antivirus ha comprobado este mensaje. http://www.eset.com __ Información de ESET NOD32 Antivirus, versión de la base de firmas de virus 4771 (20100114) __ ESET NOD32 Antivirus ha comprobado este mensaje. http://www.eset.com __ Información de ESET NOD32 Antivirus, versión de la base de firmas de virus 4771 (20100114) __ ESET NOD32 Antivirus ha comprobado este mensaje. http://www.eset.com __ Información de ESET NOD32 Antivirus, versión de la base de firmas de virus 4774 (20100115) __ ESET NOD32 Antivirus ha comprobado este mensaje. http://www.eset.com __ Información de ESET NOD32 Antivirus, versión de la base de firmas de virus 4774 (20100115) __ ESET NOD32 Antivirus ha comprobado este mensaje. http://www.eset.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime queue not work
Leif Neland wrote:
- Original Message -
*From:* Zhang Shukun mailto:bit...@gmail.com
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com
*Sent:* Friday, January 15, 2010 11:48 AM
*Subject:* [asterisk-users] Realtime queue not work
hi, all
i try to confiture realtime queue, but not work, details as below:
Insert into queue_table(name)value('95040654321');
INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun',
'95040654321', 'SIP/1001', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321',
'SIP/1002', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Zhang Jianming',
'95040654321', 'SIP/1003', 2, 1);
but when i dial 95040654321 and press extension 1. error happens:
-- Executing Queue(SIP/1003-, 950406543211)
[Jan 15 03:18:57] WARNING[16626]: app_queue.c:4223 queue_exec: Unable
to join queue '950406543211'
== Auto fallthrough, channel 'SIP/1003-' status is 'UNKNOWN'
No golden answers, but something to try.
queue names can not be just numbers? I'd try calling the queue
q95040654321.
Does show queues show the queue? Don't know if that's supposed to
work on realtime queues.
Leif
Yes, from my experience - 'queue show' will show realtime queues.
'show queues' technically is deprecated in 1.4, but should give the same
results.
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Re: [asterisk-users] 10/100 voip phones and gigabit connection
On 01/15/2010 02:19 PM, Leif Neland wrote: - Original Message - *From:* randall mailto:rand...@songshu.org *To:* asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Sent:* Friday, January 15, 2010 2:11 PM *Subject:* Re: [asterisk-users] 10/100 voip phones and gigabit connection On 01/15/2010 02:00 PM, Leif Neland wrote: - Original Message - *From:* randall mailto:rand...@songshu.org *To:* asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Sent:* Friday, January 15, 2010 7:54 AM *Subject:* [asterisk-users] 10/100 voip phones and gigabit connection list. i noticed that a lot of VOIP phones have a double network interface allowing you to use only 1 LAN cable for both the phone and your desktop, a really nice feature that can save a lot of cable, but most are 10/100 connections while i have a gigabit network. Off course there are phones available with a Gigabit connection but these are at least 3 to 4 times as expensive. In a pinch, the cheapest 1Gbit switch I could find is 17 Eur with 5 ports. Leif its not the network switch that i'm worried about, its the build in switch of the phones with the double network card Sure. My point was just that IF you only got one connection in the wall, its cheaper to get a switch than getting a phone with dual 1Gbit ports. Leif OK, point taken. but i have 6xisdn2 and already 2x24 gigabit switches (will need to replace one with a PoE version ) these connections include both desktops and current phones. i was just hoping to cut back the amount of cabling with 50%, and when i found out that most phones with 10/100/1000 connection cost about 250,- euro's a piece instead of 90,- for a decent version with 10/100 it was a real bummer, it would mean about doubling my budget. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10/100 voip phones and gigabit connection
On Fri, 15 Jan 2010, randall wrote: Sure. My point was just that IF you only got one connection in the wall, its cheaper to get a switch than getting a phone with dual 1Gbit ports. Leif OK, point taken. but i have 6xisdn2 and already 2x24 gigabit switches (will need to replace one with a PoE version ) these connections include both desktops and current phones. i was just hoping to cut back the amount of cabling with 50%, and when i found out that most phones with 10/100/1000 connection cost about 250,- euro's a piece instead of 90,- for a decent version with 10/100 it was a real bummer, it would mean about doubling my budget. I'm not sure you get it - he is saying you can eliminate the extra cable run to the desk, and place a small 5 port gigabit switch under the desk and drive both your PC and the phone from it. Total cost per desk - 90 + 17 euros. Significantly less than 250 euros for a dual gigabit port phone. No change to your switching infrastructure in your machine room. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime queue not work
Zhang Shukun wrote:
2010/1/15 Leif Neland le...@neland.dk:
- Original Message -
From: Zhang Shukun
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, January 15, 2010 11:48 AM
Subject: [asterisk-users] Realtime queue not work
hi, all
i try to confiture realtime queue, but not work, details as below:
Insert into queue_table(name)value('95040654321');
INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun',
'95040654321', 'SIP/1001', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321',
'SIP/1002', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Zhang Jianming',
'95040654321', 'SIP/1003', 2, 1);
but when i dial 95040654321 and press extension 1. error happens:
-- Executing Queue(SIP/1003-, 950406543211)
[Jan 15 03:18:57] WARNING[16626]: app_queue.c:4223 queue_exec: Unable
to join queue '950406543211'
== Auto fallthrough, channel 'SIP/1003-' status is 'UNKNOWN'
No golden answers, but something to try.
queue names can not be just numbers? I'd try calling the queue
q95040654321.
Thank you for reply. i've try a95040654321 as the queue name but not
work either.
there was the same error in the cli.
Does show queues show the queue? Don't know if that's supposed to work on
realtime queues.
Leif
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Which version of Asterisk are you using? Can you paste ( use
pastebin.com please ) your extconfig.conf and res_mysql.conf (or
res_odbc.conf)?
Something must not be configured right.
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Re: [asterisk-users] Realtime queue not work
2010/1/15 Robert Broyles bahj...@gmail.com:
Leif Neland wrote:
- Original Message -
*From:* Zhang Shukun mailto:bit...@gmail.com
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com
*Sent:* Friday, January 15, 2010 11:48 AM
*Subject:* [asterisk-users] Realtime queue not work
hi, all
i try to confiture realtime queue, but not work, details as below:
Insert into queue_table(name)value('95040654321');
INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun',
'95040654321', 'SIP/1001', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321',
'SIP/1002', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Zhang Jianming',
'95040654321', 'SIP/1003', 2, 1);
but when i dial 95040654321 and press extension 1. error happens:
-- Executing Queue(SIP/1003-, 950406543211)
[Jan 15 03:18:57] WARNING[16626]: app_queue.c:4223 queue_exec: Unable
to join queue '950406543211'
== Auto fallthrough, channel 'SIP/1003-' status is 'UNKNOWN'
No golden answers, but something to try.
queue names can not be just numbers? I'd try calling the queue
q95040654321.
Does show queues show the queue? Don't know if that's supposed to
work on realtime queues.
Leif
Yes, from my experience - 'queue show' will show realtime queues.
'show queues' technically is deprecated in 1.4, but should give the same
results.
Thank you! as i am at home now.
i can't test if 'queue show' will list the queue 'a95040654321'
I will test it in the future.
but i guess from the warning message:
Unable to join queue '950406543211'
is this mean the queue has not been set up ?
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Best regards,
Sucan
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Re: [asterisk-users] 10/100 voip phones and gigabit connection
On 01/15/2010 02:54 PM, Jeff LaCoursiere wrote: On Fri, 15 Jan 2010, randall wrote: Sure. My point was just that IF you only got one connection in the wall, its cheaper to get a switch than getting a phone with dual 1Gbit ports. Leif OK, point taken. but i have 6xisdn2 and already 2x24 gigabit switches (will need to replace one with a PoE version ) these connections include both desktops and current phones. i was just hoping to cut back the amount of cabling with 50%, and when i found out that most phones with 10/100/1000 connection cost about 250,- euro's a piece instead of 90,- for a decent version with 10/100 it was a real bummer, it would mean about doubling my budget. I'm not sure you get it - he is saying you can eliminate the extra cable run to the desk, and place a small 5 port gigabit switch under the desk and drive both your PC and the phone from it. Total cost per desk - 90 + 17 euros. Significantly less than 250 euros for a dual gigabit port phone. No change to your switching infrastructure in your machine room. j i did get it, its a good idea itself and i considered doing this a few years back, but as you can read from my reply i already have the separate cabling lying around and my old phones that i need to replace are plugged in there. Since i have these already it would mean adding extra switches and untangling the huge amounts of cluttered wires under the desks, and i'm not sure if thats worth the trouble since it usually is a little dusty there and the ladies in the office always tend to get a little nervous when i stay down there too long ;) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime queue not work
2010/1/15 Robert Broyles bahj...@gmail.com:
Zhang Shukun wrote:
2010/1/15 Leif Neland le...@neland.dk:
- Original Message -
From: Zhang Shukun
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, January 15, 2010 11:48 AM
Subject: [asterisk-users] Realtime queue not work
hi, all
i try to confiture realtime queue, but not work, details as below:
Insert into queue_table(name)value('95040654321');
INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun',
'95040654321', 'SIP/1001', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321',
'SIP/1002', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Zhang Jianming',
'95040654321', 'SIP/1003', 2, 1);
but when i dial 95040654321 and press extension 1. error happens:
-- Executing Queue(SIP/1003-, 950406543211)
[Jan 15 03:18:57] WARNING[16626]: app_queue.c:4223 queue_exec: Unable
to join queue '950406543211'
== Auto fallthrough, channel 'SIP/1003-' status is 'UNKNOWN'
No golden answers, but something to try.
queue names can not be just numbers? I'd try calling the queue
q95040654321.
Thank you for reply. i've try a95040654321 as the queue name but not
work either.
there was the same error in the cli.
Does show queues show the queue? Don't know if that's supposed to work on
realtime queues.
Leif
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Which version of Asterisk are you using? Can you paste ( use
pastebin.com please ) your extconfig.conf and res_mysql.conf (or
res_odbc.conf)?
Thanks.
I use Asterisk version 1.4.28.
in extconfig.conf related to queue is queues and queue_members:
--
queues = mysql,asterisk,queue_table
queue_members = mysql,asterisk,queue_member_table
--
res_mysql.conf file is configured right i think, because i have
sipusers, sippeers, and
extensions realtime work properly.
Something must not be configured right.
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Best regards,
Sucan
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Re: [asterisk-users] Beginners Guide to setting up a Call Centre
2010/1/12 Jeff LaCoursiere j...@jeff.net That is so not true. FreePBX has hooks in a million places to do custom dialplan stuff - I do it all the time. I also link in custom AGI/AMI applications, custom provisioning, custom LCR, and am even working with one customer that has mastered making FreePBX multi-tenant. If you want to get your hands dirty there is plenty of dirt underneath FreePBX. On the other hand, if you want a simple setup that is easily managed, the GUI is fantastic and saves a LOT of time. And if you are a PHP programmer you can easily modify the operation of any part of it. Preach it brother. We take the same approach and have never had any difficulty integrating our customisations into the FreePBX dialplan. The common structure makes it EASIER for my techies to work on systems that we built and support. On asterisk-users its traditional to be hard core and raw-dialplan and look down on those who have projects to deliver and are happy to have the help. I'm not the insecure - each of you writing your raw dialplans runs some of my code every time you run Asterisk. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginners Guide to setting up a Call Centre
2010/1/15 Peter Childs pchi...@bcs.org Elastix, Trixbox, or AsterixNow, or DIY (ie Ubuntu or whatever installed with OpenPBX, Asterix etc by hand) I've got a new server to run Asterix on and want to get it working quickly and yet be configurable in the future with out having to reisntall and start again regally. Currently no VoIP hardware but that will come once I prove the concept. I guess Oh the machine does not have a CD Rom Drive so a network/USB install would be nice.. But I guess I can open the case and plug one in for installation if I must! (Says he who has just installed Ubuntu over the network to check the computer works!) Decide if you are going to be a zealot for your preferred approach - Ubuntu and all that - or if you want a solution that works without tons of extra work. If you wisely decide that you want the latter, then get Elastix and install it. Buy QueueMetrics and install on your Elastix build. Start running your inbound call centre. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] : Asterik with out registration.
HI All, Can you please help me with the Dail plan for the following case: I want to use Asterik as a B2B,Transport proxy. I dont want to do regisration of UAC or UAS. All I will do is the dail plan update with the routing information. is it possible to do. Asterik will be used between 2 proxies . I know the IP address and all details of the proxies on both sides. with out registering how can I do the routing through the Asterik Please help me with the dail plan. Thanks, Aditya -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting Answered Stations instead of Group in cdr?
I have a dialplan entry that takes a did, and sends it to a group of stations Dial(Sip/ExtSip/ExtSip/Ext) etc. However, cdr only shows dst = 5000 (given) and lastdata shows the dial context, however I see no cdr entry for who actually answered the phone. , I can see dstchannel as SIP/-RandomHexNumbers but its not in an easy format such as dst/src.. Any easy way to get that in say the userfield? or should I just do some tsql and parse out the sip stations from dstchannel? (note, if nobody answers the call, dstchannel shows the last station in the dial context?? Which is incorrect) Any Ideas? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginners Guide to setting up a Call Centre
Stephen Davies wrote: Decide if you are going to be a zealot for your preferred approach That's a little harsh, wouldn't you say? Do whatever your most comfortable with. But, to call me and those like me a zealot, for offering advice that was asked for is a little off, in my opinion. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.29 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.4.29. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.29 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Fix to Monitor which previously assumed the file to write to did not contain pathing. (Closes issue #16377, #16376. Reported by bcnit. Patched by dant. * Propertly set T.38 attributes and don't return before T.38 ports are configured when T.38 is found but no audio stream is found. (Closes issue #16318. Reported by bird_of_Luck. Tested by vrban, mihaill. Patched by vrban, mnicholson.) * Avoid crashes with large numbers of MeetMe conferences. (Closes issue #16509. Reported by Kashif Raza. Tested, Patched by seanbright.) * Change in 'sip show channels' display format allowing more digits for CID. (Closes issue #16459. Reported, Patched by Rzadzins. * Revise documentation on disposition values to the actual values used. (Closes issue #16289. Reported by wdoekes.) A summary of changes in this release can be found in the release summary: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.4.29-summary.txt For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.29 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.0.21 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.0.21. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.0.21 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Fix to Monitor which previously assumed the file to write to did not contain pathing. (Closes issue #16377, #16376. Reported by bcnit. Patched by dant. * If EXEC only gets a single argument, don't crash when the second is used. (Closes issue #16504. Reported by bklang. Patched by tilghman.) * Avoid a crash with large numbers of MeetMe conferences. (Closes issue #16509. Reported by Kashif Raza. Tested, Patched by seanbright.) * Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces (for Solaris 10). (Patched by seanbright.) * Allow REMAINDER to function properly in expressions. (Closes issue #16427. Reported, Patched by wdoekes.) * Shut down the SIP session timers more gracefully, in order to prevent a possible crash. (Reported, Tested by corruptor. Patched by tilghman.) * Fix channel name comparison for Bridge() application. (Closes issue #16528. Reported, Patched by telecos82.) A summary of changes in this release can be found in the release summary: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0.21-summary.txt For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.21 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.1.13 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.1.13. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.1.13 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Restarts busydetector (if enabled) when DTMF is received after call is bridged (Closes issue #16389. Reported, Tested, Patched by alecdavis.) * Send parking lot announcement to the channel which parked the call, not the park-ee. (Closes issue #16234. Reported, Tested by yeshuawatso. Patched by tilghman.) * When the field is blank, don't warn about the field being unable to be coerced just skip the column. (Closes http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html) Reported by Nic Colledge on the -dev list.) * Don't queue frames to channels that have no means to process them. (Closes issue #15609. Reported, Tested by aragon. Patched by tilghman.) * Fixes holdtime playback issue in app_queue. (Closes issue #16168. Reported, Patched by nickilo. Tested by wonderg, nickilo.) A summary of changes in this release can be found in the release summary: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.1.13-summary.txt For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.13 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.2.1 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.2.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.1 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * CLI 'queue show' formatting fix. (Closes issue #16078. Reported by RoadKill. Tested by dvossel. Patched by ppyy.) * Fix misreverting from 177158. (Closes issue #15725. Reported, Tested by shanermn. Patched by dimas.) * Fixes subscriptions being lost after 'module reload'. (Closes issue #16093. Reported by jlaroff. Patched by dvossel.) * app_queue segfaults if realtime field uniqueid is NULL (Closes issue #16385. Reported, Tested, Patched by haakon.) * Fix to Monitor which previously assumed the file to write to did not contain pathing. (Closes issue #16377, #16376. Reported by bcnit. Patched by dant. A summary of changes in this release can be found in the release summary: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.2.1-summary.txt For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.1 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginners Guide to setting up a Call Centre
2010/1/15 Doug Lytle supp...@drdos.info Decide if you are going to be a zealot for your preferred approach That's a little harsh, wouldn't you say? Do whatever your most comfortable with. But, to call me and those like me a zealot, for offering advice that was asked for is a little off, in my opinion. Hi Doug, Maybe I read too much into the original poster's question, and I didn't mean to be harsh. But I used to get called in often here in South Africa to sites where the usual way wasn't good enough for someone so they'd put the whole system together the way they thought it should be done and in the process bumped into all the subtle gotchas that are mostly worked out in the standard builds. Then discovered that its harder than they thought it would be and PBX users are ungrateful b*ggers sometimes and they've walked away. Our efforts to recover these installs are always twice the work because they are tainted by what went before. But we hate to see failed Asterisk projects so we try to get them right. If your objective is to run a simple inbound call centre and get good metrics into the bargain then a FreePBX-based ISO-install (Elastic, AsteriskNow, Trixbox-CE, whathaveyou) plus Queuemetrics will have you up an running in short order. Build from the bare metal using your-own-install-of-your-preferred-distro plus raw Asterisk plus dialplan from scratch plus DIY reportage and you'll be working away after a month and cursing Asterisk. Once you're an expert then you may indeed be able to do a better job for your application than the all-in-one distros. But not first time. So apologies to the poster if I read too much into the question, but this is the sort of situation I thought of. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10/100 voip phones and gigabit connection
On Fri, Jan 15, 2010 at 1:54 AM, randall rand...@songshu.org wrote: does anybody know of another solution to this or is my conclusion above simply all the choice there is? So let me get this straight. You're planning on buying multiple Gigabit, PoE switches, and you're quibbling over the price of running parallel data cable? The gigabit PoE switches are not cheap, at least if you're buying enterprise switches that actually deliver real gigabit, with full cross-sectional bandwidth. The cable isn't very much money, and if you double-wire now, you're ready when you have twice as many employees in the same space. Next, you don't say what this office is like, but I'm going to let you in on a little secret. Most people in an office rarely spike to a full 100Mbit connection. Do some bandwidth monitoring on your network and you'll discover that. A gigabit ethernet phone is a nice thing to have, but it's more a marketing thing than an actual necessity. Anybody that can afford a gigabit ethernet switching phone and true gigabit ethernet PoE backend can afford a second wire to every desk. Please let me know the use case if you find people can't be happy with a 100Mbit connection for the typical Windoze office environment. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI and Analogue lines (UK)
Have an intersting issue whem migrating a site from Zap on 1.3 to DAHDI on 1.4.. Nothing special about the hardware - older TDM400 card, 2 red modules fitted... Both channels work fine under 1.2/Zaptel. With 1.4/DAHDI both channels still work OK, but only for one line - the 2nd line causes it to refuse to dial-out no matter which port it's plugged into. The Lines are bog-standard BT analogue lines and we're about 2Km from the exchange. Both sound good to me and dial out OK with a test phone connected to them, but only one will dial-out via the PBX. This is what I see: [Jan 1 05:14:14] WARNING[1200]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) And yet the line isn't busy or congested - nothing's using it. The output of dsx*CLI dahdi show status Description Alarms IRQbpviol CRC4 Wildcard TDM400P REV E/F Board 5 OK 0 0 0 is fine, as is: dsx*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudodefaultdefault 1incoming default 2incoming default So I'm a bit stuck. Why doesn't DAHDI like that particular line? What does it do to it that Zap didn't? Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginners Guide to setting up a Call Centre
Most important thing is to PLAN your solution out.. flowcharts, understanding where calls go, etc. Project planning, and good ideas on how the calls should be handled, and coming up with testing scenarios, to make sure everything flows correctly. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Davies Sent: Friday, January 15, 2010 10:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Beginners Guide to setting up a Call Centre 2010/1/15 Doug Lytle supp...@drdos.info Decide if you are going to be a zealot for your preferred approach That's a little harsh, wouldn't you say? Do whatever your most comfortable with. But, to call me and those like me a zealot, for offering advice that was asked for is a little off, in my opinion. Hi Doug, Maybe I read too much into the original poster's question, and I didn't mean to be harsh. But I used to get called in often here in South Africa to sites where the usual way wasn't good enough for someone so they'd put the whole system together the way they thought it should be done and in the process bumped into all the subtle gotchas that are mostly worked out in the standard builds. Then discovered that its harder than they thought it would be and PBX users are ungrateful b*ggers sometimes and they've walked away. Our efforts to recover these installs are always twice the work because they are tainted by what went before. But we hate to see failed Asterisk projects so we try to get them right. If your objective is to run a simple inbound call centre and get good metrics into the bargain then a FreePBX-based ISO-install (Elastic, AsteriskNow, Trixbox-CE, whathaveyou) plus Queuemetrics will have you up an running in short order. Build from the bare metal using your-own-install-of-your-preferred-distro plus raw Asterisk plus dialplan from scratch plus DIY reportage and you'll be working away after a month and cursing Asterisk. Once you're an expert then you may indeed be able to do a better job for your application than the all-in-one distros. But not first time. So apologies to the poster if I read too much into the question, but this is the sort of situation I thought of. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Changing ring cadence on FXS lines
Is there a way I can change the ring cadence on FXS lines on a system using a Digium Wildcard TDM2400 card? I recently deployed a new phone system, and the customer has a few POTS phones in areas where they don't have data network services, so we're using the FXS lines to provide dialtone at those outbuildings. The old phone system would ring those phones with a short ring-short ring-pause cadence, which sounds louder to the users than Asterisk's default long ring-pause cadence. I tried setting a cadence line in chan_dahdi.conf and restarting Asterisk, and typing dahdi show cadences in the CLI after the restart showed my custom cadence, but the phones were still ringing long ring-pause. Can someone point me in the direction of what I'm doing wrong? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10/100 voip phones and gigabit connection
On 01/15/2010 05:01 PM, David Backeberg wrote: On Fri, Jan 15, 2010 at 1:54 AM, randallrand...@songshu.org wrote: does anybody know of another solution to this or is my conclusion above simply all the choice there is? So let me get this straight. You're planning on buying multiple Gigabit, PoE switches, no, only 1 would be enough, think i will get this one in the link for the phones seperately http://www.salland.eu/product/350730 and you're quibbling over the price yes, i like to quibble over price, i'm dutch ;) The gigabit PoE switches are not cheap, i noticed at least if you're buying enterprise switches that actually deliver real gigabit, with full cross-sectional bandwidth. The cable isn't very much money, and if you double-wire now, you're ready when you have twice as many employees in the same space. i already have 3 cat5e wires running to each workplace, at the moment 2 are occupied with respectively 1 for a desktop and 1 for the old phone Next, you don't say what this office is like, but I'm going to let you in on a little secret. Most people in an office rarely spike to a full 100Mbit connection. Do some bandwidth monitoring on your network and you'll discover that. we use a lot of email which is IMAP based, for normal text like this email it will not be a problem but we are send huge picture attachments, when using 100mbit it can get real sluggish at times especially when you are in a hurry to forward them, plus i like to have /home directories mounted on the server. a little extra never hurts. A gigabit ethernet phone is a nice thing to have, but it's more a marketing thing than an actual necessity. i don't care for the phone to have gigabit connection, its about the desktops not losing gigabit connection Anybody that can afford a gigabit ethernet switching phone and true gigabit ethernet PoE backend can afford a second wire to every desk. its not the wires at the desk i wanted to get rid off, i was hoping to use less in the server room, from the patch panel to the switch. Please let me know the use case if you find people can't be happy with a 100Mbit connection for the typical Windoze office environment. windoze? people still use that? ;) we mostley have Xubuntu based desktops running here, not that it matters or is absolutely necessary to have, but its a terrible thing to loose if you are used to it, also i use the LAN here sometimes for cluster testing. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium Asterisk World at ITEXPO - Yahoo keynote update
I don't know how many of you are going to be at ITEXPO/Digium Asterisk World in Miami next week - I hope to see as many of you as possible, though. There has been an interesting change in the line-up for the show, that I think bears mentioning here since it possibly will help quite a few of you in your discussions about getting Asterisk into your company. We've had the good fortune to have a last-minute keynote addition at the show, which is going to be Jeremy Wadhams from Yahoo. He's going to be talking about a Fortune 500 implementation of Asterisk across their entire network, and why they made the decision to move to Open- Source telephony for the core of their voice network. Keynote: 3:30 - 4:00 PM on Wednesday - Jeremy Wadhams, Yahoo It's not yet on the ITEXPO site, since we only confirmed this yesterday with Jeremy, who is really being a great sport for doing this on short notice. Having Yahoo giving a keynote on their multi-thousand seat installation is great news for the Asterisk community in general, and is great news specifically for other Enterprise managers who have been looking for that all-important set of examples of public users of a technology. (Enterprise tends to move in herds.) There are large numbers of Enterprise users of Asterisk, but it's always great to get a name-brand company who can be offered as a proponent of Asterisk. Again, hope to see all of you there in Miami next week! http://www.tmcnet.com/voip/conference/ JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] info for Busy for incoming internal call but not for exterrnal
Hi all, I've an asterisk ver 1.4.22. As in object I have an extension beloning to a queue. I need that for an external incoming call, the extension recieve the call waiting signal/tone, whereas for internal incoming call, the extension appear busy. Is it possible? Could someone let me know the right way to do that? thanks a lot in advance lorenzo -- Chi vive sperando muore cagando ... Lo Russo isoletta dell'Egeo che non conta un cazzo, 1941 ... sono anche un autore -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
Hi Guys, Other than than yum repository (which fails when installing freepbx with it) are there any automated install scripts out there that would install Asterisk 1.6 or 1.4 onto a CentOS LAMP system? If the script install FreePBX that would be a BONUS. Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE410P generates only 1 interrupt
I am having a problem with configuring TE410P card. I am using C2SBX+ motherboard from supermicro with fc11 os After installing the dahdi drivers and running the command /etc/init.d/dahdi start, the output of /proc/interrupts shows only 2 interrupts generated (I have enabled only 2 spans in the config file). It stops generating interrupts at this point. Any clue as to why TE410P stops generating interrupts? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
On Fri, Jan 15, 2010 at 12:27 PM, Bruce Nik brucev...@gmail.com wrote: Hi Guys, Other than than yum repository (which fails when installing freepbx with it) are there any automated install scripts out there that would install Asterisk 1.6 or 1.4 onto a CentOS LAMP system? If the script install FreePBX that would be a BONUS. Thanks, Bruce Do you like 'kitchen sink' installs? I can't think of any way to decide on an asterisk configuration that out-of-the-box would be right for everybody... as in, fax support? g729 licenses? whether or not to build against DAHDI? You get the idea. The only way I can think to do it would to be to build in a lot of stuff that most people would never want in their asterisk, which would then result in having to restart asterisk because you need a software update to a package that is a dependency for a part of asterisk you don't use anyway. Anybody who was using asterisk in a serious production environment would probably prefer the control of having most of what they don't want compiled out. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
Provided there is no comprehensive install guides (or is there?) yes I would like to see an easy install script which can install it all. On Fri, Jan 15, 2010 at 12:27 PM, Bruce Nik brucev...@gmail.com wrote: Hi Guys, Other than than yum repository (which fails when installing freepbx with it) are there any automated install scripts out there that would install Asterisk 1.6 or 1.4 onto a CentOS LAMP system? If the script install FreePBX that would be a BONUS. Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
On Fri, Jan 15, 2010 at 3:15 PM, Bruce Nik brucev...@gmail.com wrote: Provided there is no comprehensive install guides (or is there?) yes I would like to see an easy install script which can install it all. tar xvzf ./configure make (optional, do a 'make menuconfig') make install But the problem is that there are steps before the configure you need if you want support for more than barebones asterisk. Nobody knows what you personally need except you. Maybe I'm the only one who doesn't think it's so bad to build from source. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10/100 voip phones and gigabit connection
On Fri, 2010-01-15 at 14:11 +0100, randall wrote: its not the network switch that i'm worried about, its the build in switch of the phones with the double network card -- Hi, Don't think you'll find phone's with an internally gbit switch. As for voip it is not needed. If you connect your pc with GB-lan card to an dual-ported ip-phone, you and up with an 100Mbps lan connection to your pc. Only way to avoid that, is to insert a cheap second lan-card in your pc, and connect your phone to the second lan, so your pc will act as an switch, instead of your phone... hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10/100 voip phones and gigabit connection
If the phone is first, then it slightly limits the PC and rebooting the phone causes loss of contact with the PC. If the PC is first you have to have dual ports on it (a few bucks of hardware, plus configuration costs), then rebooting the PC causes the phone to loose contact with the world. Not good if you are on the phone and need to reboot. Two separate feeds would work best, but cost more. Dual wall jacks with green-PC and blue-Phone jacks could then be used. The phone will be on the desk, the PC may be under it. Two jacks, two cables. If there were a standard for two Ethernet connections in a cable... that could work, but might interfere with Power Over Ethernet. I wouldn't want to be like Bill Gates saying 640K memory is enough for anyone circa-197?, but isn't two 100 meg connections enough for any single desk? The phone doesn't really need more than 10 meg. An advantage of the separate net for the phone, is that it would make POE easy for the phones, and eliminate a lot of wall warts under every desk. Plug in the phone and it works. YMMV. Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
Here is the 1.4.x version on centos 5 walk through. http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Nik Sent: Friday, January 15, 2010 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4 Provided there is no comprehensive install guides (or is there?) yes I would like to see an easy install script which can install it all. On Fri, Jan 15, 2010 at 12:27 PM, Bruce Nik brucev...@gmail.com wrote: Hi Guys, Other than than yum repository (which fails when installing freepbx with it) are there any automated install scripts out there that would install Asterisk 1.6 or 1.4 onto a CentOS LAMP system? If the script install FreePBX that would be a BONUS. Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI and Analogue lines (UK)
On Fri, Jan 15, 2010 at 04:06:54PM +, Gordon Henderson wrote: Have an intersting issue whem migrating a site from Zap on 1.3 to DAHDI on 1.4.. Nothing special about the hardware - older TDM400 card, 2 red modules fitted... Both channels work fine under 1.2/Zaptel. With 1.4/DAHDI both channels still work OK, but only for one line - the 2nd line causes it to refuse to dial-out no matter which port it's plugged into. The Lines are bog-standard BT analogue lines and we're about 2Km from the exchange. Both sound good to me and dial out OK with a test phone connected to them, but only one will dial-out via the PBX. This is what I see: [Jan 1 05:14:14] WARNING[1200]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) And yet the line isn't busy or congested - nothing's using it. The output of dsx*CLI dahdi show status Description Alarms IRQbpviol CRC4 Wildcard TDM400P REV E/F Board 5 OK 0 0 0 is fine, as is: dsx*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudodefaultdefault 1incoming default 2incoming default So I'm a bit stuck. Why doesn't DAHDI like that particular line? What does it do to it that Zap didn't? What version of Zaptel? What is the value of 'InAlarm' from 'dahdi show channel 2' ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI and Analogue lines (UK)
On Sat, 16 Jan 2010, Tzafrir Cohen wrote: On Fri, Jan 15, 2010 at 04:06:54PM +, Gordon Henderson wrote: Have an intersting issue whem migrating a site from Zap on 1.3 to DAHDI on 1.4.. Nothing special about the hardware - older TDM400 card, 2 red modules fitted... Both channels work fine under 1.2/Zaptel. With 1.4/DAHDI both channels still work OK, but only for one line - the 2nd line causes it to refuse to dial-out no matter which port it's plugged into. The Lines are bog-standard BT analogue lines and we're about 2Km from the exchange. Both sound good to me and dial out OK with a test phone connected to them, but only one will dial-out via the PBX. This is what I see: [Jan 1 05:14:14] WARNING[1200]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) And yet the line isn't busy or congested - nothing's using it. The output of dsx*CLI dahdi show status Description Alarms IRQbpviol CRC4 Wildcard TDM400P REV E/F Board 5 OK 0 0 0 is fine, as is: dsx*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudodefaultdefault 1incoming default 2incoming default So I'm a bit stuck. Why doesn't DAHDI like that particular line? What does it do to it that Zap didn't? What version of Zaptel? Oldish - Zaptel Version: 1.2.23 What is the value of 'InAlarm' from 'dahdi show channel 2' ? InAlarm: 1 That's not good, is it... Doesn't explain why an analogue phone connected to the line works OK though - or can it indicate another sort of fault, or is it just too fussy? The line itself is their FAX line, although I'm not using it for FAXes - just as a second outgoing call line (I have it arranged to innore incoming calls - which are detected) There is also another phone on the line, so 3 devices including the asterisk box, however I got the same result with it plugged directly into the master socket with nothing else connected. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Detection on SIP
I have NVFaxDetect working 100% with Asterisk 1.6. Check out this article in my blog on details of how I got it working: http://cloudsconnected.com/?p=57 Good luck! On Fri, 2010-01-15 at 11:28 +, --[ UxBoD ]-- wrote: - Paul Scott p...@cpanel.net wrote: Yeah sounds like you wanna use NVFaxDetect it would allow you to add something like exten = fax,1,Swift(number has changed); to your inbound call part of your dialplan On Jan 14, 2010, at 11:45 AM, Juan C. Villa wrote: Could you use NVFaxDetect? On Thu, 2010-01-14 at 17:35 +, --[ UxBoD ]-- wrote: Hi, We have a issue where one of our clients is receiving a high volume of calls from automated fax machines and passing through their context which means all phones get rung. I am looking for a way to detect the fax tone, on answer, and route it to a extension/macro where a announcement would tell them the correct number to fax to. The inbound calls are across SIP so am not able to use DADHI. Would this scenario be perfect for Fax for Asterisk ? All help greatfully appreciated. The problem is that NVFaxDetect does not appear to work with Asterisk 1.6.X :( Looking for something that is actively supported. Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo on Polycom phones
We are using Polycom 550 and 650 phones and OSLEC echo cancellation software with Asterisk. Occasionally, we get echo on our PRI phone calls. The echo is always from our voice echoing back to us. How can I fix this echo? I have tried installing the VPMADT032 module on our TE121 card, but that made it worse. Thank you! This is my chan_dahdi.conf: [channels] context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no faxdetect=no echotraining=800 rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 This is part of my sip.cfg file: volume voice.volume.persist.handset=1 voice.volume.persist.headset=1 voice.volume.persist.handsfree=1/ gains voice.gain.rx.analog.handset=0 voice.gain.rx.analog.headset=0 voice.gain.rx.analog.chassis=0 voice.gain.rx.analog.chassis.IP_300=-6 voice.gain.rx.analog.chassis.IP_4000=3 voice.gain.rx.analog.chassis.IP_430=0 voice.gain.rx.analog.chassis.IP_650=0 voice.gain.rx.analog.chassis.IP_601=6 voice.gain.rx.analog.ringer=0 voice.gain.rx.analog.ringer.IP_300=-6 voice.gain.rx.analog.ringer.IP_4000=3 voice.gain.rx.analog.ringer.IP_430=0 voice.gain.rx.analog.ringer.IP_650=0 voice.gain.rx.analog.ringer.IP_601=6 voice.gain.rx.digital.handset=-15 voice.gain.rx.digital.headset=-21 voice.gain.rx.digital.chassis=0 voice.gain.rx.digital.chassis.IP_4000=0 voice.gain.rx.digital.chassis.IP_430=0 voice.gain.rx.digital.chassis.IP_650=6 voice.gain.rx.digital.chassis.IP_601=0 voice.gain.rx.digital.ringer=-21 voice.gain.rx.digital.ringer.IP_4000=-21 voice.gain.rx.digital.ringer.IP_430=-21 voice.gain.rx.digital.ringer.IP_650=-12 voice.gain.rx.digital.ringer.IP_601=-21 voice.gain.rx.analog.handset.sidetone=-14 voice.gain.rx.analog.headset.sidetone=-24 voice.gain.tx.analog.handset=12 voice.gain.tx.analog.headset=3 voice.gain.tx.analog.chassis=3 voice.gain.tx.analog.chassis.IP_300=0 voice.gain.tx.analog.chassis.IP_4000=3 voice.gain.tx.analog.chassis.IP_430=42 voice.gain.tx.analog.chassis.IP_650=36 voice.gain.tx.analog.chassis.IP_601=0 voice.gain.tx.digital.handset=0 voice.gain.tx.digital.headset=0 voice.gain.tx.digital.chassis=3 voice.gain.tx.digital.chassis.IP_4000=0 voice.gain.tx.digital.chassis.IP_430=-3 voice.gain.tx.digital.chassis.IP_650=0 voice.gain.tx.digital.chassis.IP_601=6 voice.gain.tx.analog.preamp.handset=14 voice.gain.tx.analog.preamp.headset=23 voice.gain.tx.analog.preamp.chassis=32 voice.gain.tx.analog.preamp.chassis.IP_430=32 voice.gain.tx.analog.preamp.chassis.IP_601=32/ AEC voice.aec.hs.enable=0 voice.aec.hs.lowFreqCutOff=100 voice.aec.hs.highFreqCutOff=7000 voice.aec.hs.erlTab_0_300=-24 voice.aec.hs.erlTab_300_600=-24 voice.aec.hs.erlTab_600_1500=-24 voice.aec.hs.erlTab_1500_3500=-24 voice.aec.hs.erlTab_3500_7000=-24 voice.aec.hd.enable=0 voice.aec.hd.lowFreqCutOff=100 voice.aec.hd.highFreqCutOff=7000 voice.aec.hd.erlTab_0_300=-24 voice.aec.hd.erlTab_300_600=-24 voice.aec.hd.erlTab_600_1500=-24 voice.aec.hd.erlTab_1500_3500=-24 voice.aec.hd.erlTab_3500_7000=-24 voice.aec.hf.enable=1 voice.aec.hf.lowFreqCutOff=100 voice.aec.hf.highFreqCutOff=7000 voice.aec.hf.erlTab_0_300=-6 voice.aec.hf.erlTab_300_600=-6 voice.aec.hf.erlTab_600_1500=-6 voice.aec.hf.erlTab_1500_3500=-6 voice.aec.hf.erlTab_3500_7000=-6/ AES voice.aes.hs.enable=0 voice.aes.hs.duplexBalance=7 voice.aes.hd.enable=0 voice.aes.hd.duplexBalance=0 voice.aes.hf.enable=1 voice.aes.hf.duplexBalance.0=9 voice.aes.hf.duplexBalance.1=8 voice.aes.hf.duplexBalance.2=7 voice.aes.hf.duplexBalance.3=6 voice.aes.hf.duplexBalance.4=5 voice.aes.hf.duplexBalance.5=4 voice.aes.hf.duplexBalance.6=3 voice.aes.hf.duplexBalance.7=2 voice.aes.hf.duplexBalance.8=1 voice.aes.hf.duplexBalance.IP_4000.0=10 voice.aes.hf.duplexBalance.IP_4000.1=9 voice.aes.hf.duplexBalance.IP_4000.2=8 voice.aes.hf.duplexBalance.IP_4000.3=7 voice.aes.hf.duplexBalance.IP_4000.4=6 voice.aes.hf.duplexBalance.IP_4000.5=5 voice.aes.hf.duplexBalance.IP_4000.6=4 voice.aes.hf.duplexBalance.IP_4000.7=3 voice.aes.hf.duplexBalance.IP_4000.8=2/ NS voice.ns.hs.enable=0 voice.ns.hs.signalAttn=-6 voice.ns.hs.silenceAttn=-9 voice.ns.hd.enable=0 voice.ns.hd.signalAttn=0 voice.ns.hd.silenceAttn=0 voice.ns.hf.enable=1 voice.ns.hf.signalAttn=-6 voice.ns.hf.silenceAttn=-9 voice.ns.hf.IP_4000.enable=1 voice.ns.hf.IP_4000.signalAttn=-6 voice.ns.hf.IP_4000.silenceAttn=-9/ AGC voice.agc.hs.enable=0 voice.agc.hd.enable=0 voice.agc.hf.enable=0/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginners Guide to setting up a Call Centre
Peter Childs wrote: Ok this has Probably been put to bed several time but never mind. Elastix, Trixbox, or AsterixNow, or DIY (ie Ubuntu or whatever installed with OpenPBX, Asterix etc by hand) I've got a new server to run Asterix on and want to get it working quickly and yet be configurable in the future with out having to reisntall and start again regally. Currently no VoIP hardware but that will come once I prove the concept. I guess Oh the machine does not have a CD Rom Drive so a network/USB install would be nice.. But I guess I can open the case and plug one in for installation if I must! (Says he who has just installed Ubuntu over the network to check the computer works!) Peter. In regards to a CD/DVD drive, I have a small ISP farm for web/email hosting. I stopped putting cd/dvd readers in the servers about 2 years ago. All the new motherboards out there support booting from a USB drive, so why bother? Get one good DVD drive and put it in a case with a USB adapter in it and just plug it in when you need it. Lyle Giese LCR Computer Services, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing ring cadence on FXS lines
Noah I. Engelberth wrote: Is there a way I can change the ring cadence on FXS lines on a system using a Digium Wildcard TDM2400 card? I recently deployed a new phone system, and the customer has a few POTS phones in areas where they don't have data network services, so we're using the FXS lines to provide dialtone at those outbuildings. The old phone system would ring those phones with a short ring-short ring-pause cadence, which sounds louder to the users than Asterisk's default long ring-pause cadence. I tried setting a cadence line in chan_dahdi.conf and restarting Asterisk, and typing dahdi show cadences in the CLI after the restart showed my custom cadence, but the phones were still ringing long ring-pause. Can someone point me in the direction of what I'm doing wrong? http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels Lyle Giese LCR Computer Services, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo on Polycom phones
hin lee wrote: We are using Polycom 550 and 650 phones and OSLEC echo cancellation software with Asterisk. Occasionally, we get echo on our PRI phone calls. The echo is always from our voice echoing back to us. How can I fix this echo? I have tried installing the VPMADT032 module on our TE121 card, but that made it worse. rxgain=0.0 txgain=0.0 Try reducing your transmit gains (txgain=-4.0 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10/100 voip phones and gigabit connection
On Fri, 15 Jan 2010, Hans Witvliet wrote: If you connect your pc with GB-lan card to an dual-ported ip-phone, you and up with an 100Mbps lan connection to your pc. Only way to avoid that, is to insert a cheap second lan-card in your pc, and connect your phone to the second lan, so your pc will act as an switch, instead of your phone... I'm curious - how have you managed to connect a second LAN card and have it bridge your (presumably onboard) ethernet? Does Windows have such capability? But I guess the OP was running XUbuntu, and though relatively complicated I guess you could get it to do that. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10/100 voip phones and gigabit connection
Windows, yes, but used to be through 3rd party software. Doubt this has changed as Windows has no focus on any useful network anything. Linux, yes, and it's definitely not complicated. Probably take 2 minutes to setup if you already had bridge utils installed, maybe 5 if you had to install the package first. Regardless, I would say this wouldn't be the best solution for the reasons someone already mentioned - you form some dependency between the phone and the PC. 2 cables is definitely the best, followed by a cheap gig switch at each desk. Also, someone was mentioning if you could run 2 ethernet connections through one cable. This works with 10/100 (as only 2 pairs are used, so you can wire 2 pairs to one jack, and the other 2 pairs in the cat5 to the other jack), but doesn't work with gigabit, as it uses all 4 pairs. Even if you just consider 10/100, this is a nifty hack in a pinch, but I seriously doubt anyone does this in a professional install. Cable isn't all that expensive when it comes right down to it. Andrew On Fri, Jan 15, 2010 at 6:38 PM, Jeff LaCoursiere j...@jeff.net wrote: On Fri, 15 Jan 2010, Hans Witvliet wrote: If you connect your pc with GB-lan card to an dual-ported ip-phone, you and up with an 100Mbps lan connection to your pc. Only way to avoid that, is to insert a cheap second lan-card in your pc, and connect your phone to the second lan, so your pc will act as an switch, instead of your phone... I'm curious - how have you managed to connect a second LAN card and have it bridge your (presumably onboard) ethernet? Does Windows have such capability? But I guess the OP was running XUbuntu, and though relatively complicated I guess you could get it to do that. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime cached values
Hello, Does the cached values for realtime peers expire automatically? I have rtcachefriends=yes in sip.conf. When the peer registers for the first time it is cached. After the first registration if I modify the peer in the database the new values are not used until I do a 'sip prune realtime peer' or 'sip reload'. Is it possible to specify an expiry time for the cached values so that asterisk does a database look-up again. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginners Guide to setting up a Call Centre
Looking at all the docs I can find Asterisks looks like it should be able to do the job and a whole lot more. This is for a small call centre so ideally we want all the features of an average call centre, ACD, Call Recording, Queue's etc etc. Any pointers on how to get started would be most helpful. Peter. --- (sorry this is so long) Peter, I figured that I would chime in, as I run IT and am a managing partner of a small call center based on Asterisk and I think that my experience will be helpful (hate to beat a dead horse)... Asterisk can definitely do what you need, so I am not going to talk about that any further. I wouldn't waste my time with anything else. I would strongly recommend either of the two following methods to get started, with the deciding factors being time and money. There are lots of factors that will sway this argument, such as the complexity of your workflow, CTI needs, etc., but those time and money are the biggies. You also have to carefully weigh your support requirements, uptime, and your desire to manage a phone system. Asterisk doesn't have to take that much work once it's installed and tuned, but it will require some maintenance. You will need to evaluate whether or not you want to take on that maintenance role or whether you want to pay to have it done for you. Method 1: A professional installation by a Digium Certified Asterisk Professional. It will cost you some money, how much depends on your needs and how clearly you articulate them. There are lots of great people out there that can help you get EXACTLY what you want and design a system that will grow with your business. Call Digium for recommendations, or reply to this with your contact info and we can talk off list (I'm not trying to sell anything, but I have some people that I can recommend). This can be a great option for a solid Asterisk system with good support and reliable operation with little maintenance. There's a couple different approaches to this method- managed and developed with support. Managed is where the team that developed the dialplan and asterisk environment for you manages the system for you as well for a recurring support fee. Drawbacks to this method: A. You will have to find a good vendor that will charge fairly and deliver on their SLA (always get an SLA with enforceable penalties). This isn't that tough, but it's important. B. The recurring support costs can eat into your budget quickly C. This will take some time to develop properly, and for simple environments it may be overkill. D. Adds/changes/ and deletes can be costly as well. This can be mitigating by communicating the need to accommodate staff turnover with a user maintainable system. Method 2: Get a distro, install it, be dialing in about 8 hours or less (the route that I took when we started). This method is by far and away the easiest, cheapest, get-it-up-and-running-consequences-be-damned method. You will take less time, effort and money to get going like this than any other way I know of. If your call flow is simple to moderately complex, this is the way to go in my opinion. The FreePBX distros (Trixbox, AsteriskNOW [I think], Elastix, etc) all are very well put together, and will do everything that you listed in your original message and then some. Of the distro's, I would probably either go with AsteriskNOW or, if you are up for a little more setup work, FreePBX on it's own. Drawbacks to this method: A. I can't speak for others, but I found that the configuration engines have their limitations when it comes to call centers. They simply weren't designed to do some of the specific things that we needed to do as we grew. This doesn't mean that they wont do everything you need though, each case is unique. They were fine for us in the beginning, but as our business grew so did our specific needs, and we outgrew these solutions. There is nothing wrong with that if you understand from the outset that you may have needs that aren't met in the future. These distros have to factor in the needs of their respective communities, and what may be good for one organization might not be good for others. B. Troubleshooting issues can be more complex as you start to understand Asterisk and increase your level of sophistication. I had a hard time troubleshooting FreePBX until I understood it's dialplan more, and it made troubleshooting complicated as I didn't fully understand the call flow through it's dialplan. The more you work with it, the easier it gets, but there can be a learning curve. C. Integration with other vendor's products can sometimes be a challenge if they don't already support your configuration GUI. D. You cannot, no matter what anyone tells you (I know this to be cold, hard fact), modify the built in dialplan of FreePBX. When you upgrade, or even
Re: [asterisk-users] Fax Detection on SIP
2010/1/15 --[ UxBoD ]-- ux...@splatnix.net - --[ UxBoD ]-- ux...@splatnix.net wrote: - Paul Scott p...@cpanel.net wrote: Yeah sounds like you wanna use NVFaxDetect it would allow you to add something like exten = fax,1,Swift(number has changed); to your inbound call part of your dialplan On Jan 14, 2010, at 11:45 AM, Juan C. Villa wrote: Could you use NVFaxDetect? On Thu, 2010-01-14 at 17:35 +, --[ UxBoD ]-- wrote: Hi, We have a issue where one of our clients is receiving a high volume of calls from automated fax machines and passing through their context which means all phones get rung. I am looking for a way to detect the fax tone, on answer, and route it to a extension/macro where a announcement would tell them the correct number to fax to. The inbound calls are across SIP so am not able to use DADHI. Would this scenario be perfect for Fax for Asterisk ? All help greatfully appreciated. The problem is that NVFaxDetect does not appear to work with Asterisk 1.6.X :( Looking for something that is actively supported. Well perhaps I should have read the sip.conf ;) Okay so I am able to set faxdetect=yes and then I presume just create a extension in the target context: exten = fax,1,Play(sorry-wrong-fax-number) exten = fax,n,Hangup() What about the answering phone (the one that received the call that was later detected as a fax call) ? To my knowledge, when asterisk is jumping into fax priority, the answering phone would hear a busy tone. Is there a way to let this phone hear a nice prompt saying you're currently receiving a fax) ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
