[asterisk-users] Listed agents in queue not ringing
Hi All,
I am having a problem when we call inbound the ivr picks up send caller to
the queue but does not forward the calls to the listed agents, however if we
use the call groups instead of queues it rings to the listed agents in group
Here are the default settings
include=DID_suhaib_timeinterval_inbound
include=DID_suhaib_timeinterval_AllTheTime
include=DID_suhaib_timeinterval_OfficeHours
include=DID_suhaib_timeinterval_AfterHours
include=DID_suhaib_timeinterval_Weekend
include=DID_suhaib_timeinterval_EarlyHours
exten=0,1,
exten=o,1,Goto(default,6043,1)
exten=6800,1,VoiceMailMain(${CALLERID(num)}...@default)
Here is the office hour interval settings
exten=__X.,1,Goto(voicemenu-custom-2|s|1)
exten=_XX,1,Goto(queues|6525|1)
Here are the queues settings
exten=6500,1,Queue(${EXTEN})
exten=6501,1,Queue(${EXTEN})
exten=6502,1,Queue(${EXTEN})
exten=6503,1,Queue(${EXTEN})
exten=6509,1,Queue(${EXTEN})
exten=6900,1,agentlogin()
exten=6950,1,agentcallbacklogin()
Here is the queue settings
fullname=TechPC
strategy=ringall
timeout=180
wrapuptime=15
autofill=yes
autopause=no
joinempty=yes
leavewhenempty=no
reportholdtime=yes
maxlen=0
musicclass=default
member=Agent/6029
member=Agent/6038
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[asterisk-users] Help: DTMF problem
Hi, I have polycom 301 IP phones most of them especially when I call a direct line with extensions, I cannot dial an extension. It does not recognize my key inputs. If the number is an 800 number it seems to work fine. I have used dtmfmode=inband with my sip trunks and my extensions as rfc2833. Any suggestions will really be appreciated. Thanks in advance Jarga Jallow Technical Support Engineer 2985 S. Hwy. 360 Grand Praire, Texas 75052 Direct: 972-206-1212 ext# 29 Mobile: 214-669-9046 Fax:972-999-4113 Toll Free: 1-877-801-5511 ext 34 Toll Free: 1-877-926-2288 http://www.2mcctv.com/ www.2mcctv.com http://www.2mcctv.com/ image001.gifimage002.jpgimage003.gif___ -- 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] Help: dtmf mode
Hi, I am having trouble making a selection when I call a number and need to make a selection to go to an extension with my polycom phones 301. Anybody have an idea how to fix this problem? Thanks in advance. Jarga Jallow Technical Support Engineer 2985 S. Hwy. 360 Grand Praire, Texas 75052 Direct: 972-206-1212 ext# 29 Mobile: 214-669-9046 Fax:972-999-4113 Toll Free: 1-877-801-5511 ext 34 Toll Free: 1-877-926-2288 http://www.2mcctv.com/ www.2mcctv.com http://www.2mcctv.com/ image001.jpgimage002.gifimage003.gif___ -- 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] Help: dtmf mode
Sip.conf : ; Note: If your SIP devices are behind a NAT and your Asterisk ; server isn't, try adding nat=1 to each peer definition to ; solve translation problems. [general] bindport = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw ; If you need to answer unauthenticated calls, you should change this ; next line to 'from-trunk', rather than 'from-sip-external'. ; You'll know this is happening if when you call in you get a message ; saying The number you have dialed is not in service. Please check the ; number and try again. context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown tos=0x68 ; #, in this configuration file, is NOT A COMMENT. This is exactly ; how it should be. #include sip_nat.conf #include sip_registrations_custom.conf #include sip_registrations.conf #include sip_custom.conf #include sip_additional.conf I am calling other external phones, I think they PSTN destinations. Jarga Jallow Technical Support Engineer 2985 S. Hwy. 360 Grand Praire, Texas 75052 Direct: 972-206-1212 ext# 29 Mobile: 214-669-9046 Fax:972-999-4113 Toll Free: 1-877-801-5511 ext 34 Toll Free: 1-877-926-2288 www.2mcctv.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Johnson Sent: Thursday, January 24, 2008 2:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help: dtmf mode Please post your sip.conf entry for your phone and also describe your calling path. Are you having a problem with internal calls (e.g.: to voicemailmain) on the same switch, or are you referring to calls to PSTN destinations via pots/pri/sip/? Also, which versions of Asterisk, Zaptel, linux, etc. are you using? S. On Jan 24, 2008 12:43 PM, Jarga Jallow [EMAIL PROTECTED] wrote: Hi, I am having trouble making a selection when I call a number and need to make a selection to go to an extension with my polycom phones 301. Anybody have an idea how to fix this problem? Thanks in advance. Jarga Jallow Technical Support Engineer 2985 S. Hwy. 360 Grand Praire, Texas 75052 Direct: 972-206-1212 ext# 29 Mobile: 214-669-9046 Fax:972-999-4113 Toll Free: 1-877-801-5511 ext 34 Toll Free: 1-877-926-2288 www.2mcctv.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 ___ -- 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
Re: [asterisk-users] Help: dtmf mode
On the polycom manual it says for g729 use rfc1890. I did that but sometimes it works sometimes it doesn't. Am not sure why. Jarga Jallow Technical Support Engineer 2985 S. Hwy. 360 Grand Praire, Texas 75052 Direct: 972-206-1212 ext# 29 Mobile: 214-669-9046 Fax:972-999-4113 Toll Free: 1-877-801-5511 ext 34 Toll Free: 1-877-926-2288 www.2mcctv.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Thursday, January 24, 2008 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help: dtmf mode My polycoms all have dtmfmode=rfc2833 and they work fine on both asterisk's IVRs and external ones brought to me from the PSTN: [120] type=friend context=internalaugmented secret=a_secret host=dynamic *dtmfmode=rfc2833* Moj Jarga Jallow wrote: Hi, I am having trouble making a selection when I call a number and need to make a selection to go to an extension with my polycom phones 301. Anybody have an idea how to fix this problem? Thanks in advance. Jarga Jallow Technical Support Engineer 2985 S. Hwy. 360 Grand Praire, Texas 75052 Direct: 972-206-1212 ext# 29 Mobile: 214-669-9046 Fax:972-999-4113 Toll Free: 1-877-801-5511 ext 34 Toll Free: 1-877-926-2288 www.2mcctv.com http://www.2mcctv.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 ___ -- 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] Help Dial extention
I have a Linksys sipura phone which does not dial ext 26 only, every other ext works. When I dial ext 26 it show to:0 instead. Does anybody know how to fix this? Thanks in advance. Jarga Jallow image001.jpg___ --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] Help with Polycom 320
I am having trouble configuring my Polycom 320 IP phone. When I dial an extension it seems like am calling from outside. Also on the phone menu it says not registered. Does anybody know how to fix this? Thanks in advance Jarga Jallow Technical Support Engineer 2985 S. Hwy. 360 Grand Praire, Texas 75052 Direct: 972-206-1212 ext# 29 Mobile: 214-669-9046 Fax:972-999-4113 Toll Free: 1-877-801-5511 ext 34 Toll Free: 1-877-926-2288 http://www.2mcctv.com/ www.2mcctv.com http://www.2mcctv.com/ image002.jpgimage003.gifimage004.gif___ --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 Help
I have them working now, but I do not understand what you mean. Can you elaborate more? Thanks Jarga Jallow Technical Support Engineer 2985 S. Hwy. 360 Grand Praire, Texas 75052 Direct: 972-206-1212 ext# 29 Mobile: 214-669-9046 Fax:972-999-4113 Toll Free: 1-877-801-5511 ext 34 Toll Free: 1-877-926-2288 www.2mcctv.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, November 08, 2007 1:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Help On Nov 6, 2007 2:25 PM, Jarga Jallow [EMAIL PROTECTED] wrote: Under asterisk info: Sip registry 12/12 76.xxx.xxx.xxx D N 5066 UNREACHABLE 11/11 76.xxx.xxx.xxx D N 5064 UNREACHABLE 10/10 76.xxx.xxx.xxx D N 5062 UNREACHABLE All these IP phones are behind NAT. What could be the problem? You aren't supposed to be registering to your IP phones you should have the IP phones registering against your 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 ___ --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] Help: Asterisk info
I am getting this error under system info: File Line Command Message common_functions.php 314 file_exists(/proc/scsi/scsi) the file does not exist on your machine Does anybody know how to fix this? Thank you in advance Jarga image001.jpg___ --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 Help
Under asterisk info: Sip registry 12/12 76.xxx.xxx.xxx D N 5066 UNREACHABLE 11/11 76.xxx.xxx.xxx D N 5064 UNREACHABLE 10/10 76.xxx.xxx.xxx D N 5062 UNREACHABLE All these IP phones are behind NAT. What could be the problem? Thanks in advance. Jarga image002.jpg___ --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] Help: Static and dropped calls
Does anybody know why am getting a lot of static and sometimes dropped calls from my asterisk server. Vitelity is my number provider if it matters. Thank you Jarga Jallow image001.jpg___ --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] Help
I need help with my grand stream GXP2000 phones they keep freezing randomly. Any ideas? Jarga image001.jpg___ --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] Help
GXP2000 Firmware/sofware version is 1.1.4.18 Jarga -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris 'Xenon' Hanson Sent: Thursday, November 01, 2007 11:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help Jarga Jallow wrote: I need help with my grand stream GXP2000 phones they keep freezing randomly. Any ideas? What firmware revision? Want to buy a used one from me? I'm trying to standardize on Sipura 841s, and I have one GXP2000. Jarga -- Chris 'Xenon' Hanson | Xenon @ 3D Nature | http://www.3DNature.com/ I set the wheels in motion, turn up all the machines, activate the programs, and run behind the scenes. I set the clouds in motion, turn up light and sound, activate the window, and watch the world go 'round. -Prime Mover, Rush. ___ --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
