[asterisk-users] Hardware Compatibility HP Proliant - Sangoma PCI Express
Hi to all, I am in the process of setup a new asterisk server, I think in the HP Proliant ML350 G6 Server (aprox. 100 SIP Users), and Sangoma A102DE Card. The specs of the Proliant (HP PART 487932-001) about PCI are the next. 1 ( 1 ) x PCI Express 2.0 x16 ( x8 mode ) , 1 ( 1 ) x PCI Express 2.0 x8 ( x8 mode ) , 4 ( 3 ) x PCI Express 2.0 x8 ( x4 mode ) The question is, if the card is compatible with the PCI slots in the server? And. If there is a known issue with this combination? Thanks a lot. Ricardo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Got SIP response 603 decline, then the call hang up
Hi to all, I have a strange behavior in my asterisk server.
I have a queue for 5 agents, the calls enter the queue an go to the agents
normally, but if I need to transfer or dial directly to an agent extension
that is already in a call, the pbx hung up the actual call (not the
transferred call).
This is what I see in the log.
Called 103
-- Agent/103 is ringing
-- SIP/103-0e89 is ringing
-- Got SIP response 603 Decline back from 192.168.215.104// (104
is the IP of SIP/103)
== Spawn extension (cola-radio2, s, 4) exited non-zero on 'DAHDI/6-1'
-- SIP/103-0e89 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [...@agentdialfromq:2]
Hangup(Local/1...@agentdialfromq-eb28,2, ) in new stack
== Spawn extension (AgentDialFromQ, 103, 2) exited non-zero on
'Local/1...@agentdialfromq-eb28,2'
-- Hungup 'DAHDI/6-1'
As you can see when dialing to SIP/103 I got 603 and the actual call hung
up.
This is my queues.conf and agents.conf
[general]
;monitor-type=MixMonitor
;##
persistentmembers=yes
autofill=yes
joinempty = strict
leavewhenempty = strict
;##
[cola-radio]
musicclass = default
joinempty = strict
leavewhenempty = strict
;###
reportholdtime = no
ringinuse = no
strategy = rrmemory
timeout=15
retry=0
wrapuptime=1
maxlen=6
servicelevel = 60
memberdelay = 0
timeoutrestart = no
;###
announce=beep
announce-frequency = 30
announce-holdtime = yes
periodic-announce-frequency=1
;periodic-announce=cu_periodic_announce
;periodic-announce=/var/lib/asterisk/cus_sounds/cu_periodic_announce
context = ivr-cola-radio
;monitor-format = gsm
;monitor-type = MixMonitor
;monitor-join = yes
;Queue Members
member = Agent/101
member = Agent/103
member = Agent/104
member = Agent/105
member = Agent/106
member = Agent/109
;member = Agent/110
member = Agent/111
member = Agent/112
member = Agent/113
member = Agent/114
member = Agent/115
member = Agent/116
member = Agent/117
member = Agent/118
member = Agent/119
member = Agent/120
AGENTS.CONF
[agents]
; Enable recording calls addressed to agents. It's turned off by default.
recordagentcalls=yes
;
; The format to be used to record the calls: wav, gsm, wav49.
; By default its wav.
recordformat=wav
;
; The text to be added to the name of the recording. Allows forming a url
link.
;urlprefix=http://localhost/calls/
;
; The optional directory to save the conversations in. The default is
; /var/spool/asterisk/monitor
savecallsin=/var/spool/asterisk/monitor/Qcabina
ackcall=no
persistentagents=yes
;musiconhold=default
;###
autologoffunavail=yes
wrapuptime=1000
;###
agent = 101,,Operador 1
agent = 103,,Operador 3
agent = 104,,Operador 4
agent = 105,,Operador 5
agent = 106,,Operador 6
;agent = 107,,Operador 7
;agent = 108,,Operador 8
agent = 109,,Operador 9
;agent = 110,,Operador 10
agent = 111,,Operador 11
agent = 112,,Operador 12
agent = 113,,Operador 13
agent = 114,,Operador 14
agent = 115,,Operador 15
agent = 116,,Operador 16
agent = 117,,Operador 17
agent = 118,,Operador 18
agent = 119,,Operador 19
agent = 120,,Operador 20
AND THE INTERESTING PART IN DIALPLAN
-To log into the queue
exten =
*402,1,AgentCallBackLogin(${CALLERID(num)}||${CALLERID(num)}...@agentdialfromq)
-the AgentDialFronQ context
[AgentDialFromQ]
exten =_1XX,1,Dial(SIP/${EXTEN},,tTr)
exten =_1XX,n,Hangup
What can be the problem?
Thanks for any help.
Ricardo
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A101DE with Dell PE 2850
Thanks a lot for that info Christian. Date: Tue, 8 Dec 2009 20:14:02 +0100 From: Christian Victor christ...@victormedia.de Subject: Re: [asterisk-users] Sangoma A101DE with Dell PE 2850 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 9b9941b90912081114w3db968f9ke2b4ce2d15622...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 2009/12/8 Ricardo Melendez rmelen...@utep.com.mx: First I see at sangoma page that A101DE is PCI-Express? (I think ?x1 for the size of the connector) Yes, it is PCIe x1. There is an A101D wich is PCI(-X). for PCI Express one x4 lane width one x8 lane width I can connect the card to any of the slots?, or only to PCI-Express Slots? (is compatible the card with x4 and x8 PCI-Express slots?) Yes, the A101DE runs in PCIe x4 or x8 and the A101D will run in PCI or PCI-X Christian ___ -- 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] Sangoma A101DE with Dell PE 2850
Hi friends, I am about to install an asterisk server using a Sangoma A101DE over a Dell PE 2850 Server but I have doubts about PCI requirements. First I see at sangoma page that A101DE is PCI-Express (I think x1 for the size of the connector) And the specs for the PE 2850 is For PCI-X one 3.3-V, 64-bit, 100-MHz or three 3.3-V, 64 bit, 133MHz for PCI Express one x4 lane width one x8 lane width I can connect the card to any of the slots?, or only to PCI-Express Slots? (is compatible the card with x4 and x8 PCI-Express slots?) Thanks in advance. Atte. Ricardo ___ -- 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] Flashing Cisco 7941 to SIP
Hi to All, I am trying to flash to SIP image one Cisco 7941 IP Phone to work with Asterisk, I have searched the internet and find some instructions but I need the firmware SIP Image to complete the flash. Can anyone help me with the SIP image for Cisco 7941? The image name is cmterm-7941_7961-sip.8-5-2.cop Thanks in advance. Ricardo ___ -- 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] Messaging System
Hi to All, I need to implement an automatic telephone messaging system that works like this: -the system generates the call based on mysql records or any database -when the client answer the phone, the Asterisk PBX playback a recorded message -when finish, hang up the channel. Only for voice messages not SMS. Exists some application based on Asterisk that makes this, or any code to implement in dialplan Thanks in advance. Ricardo ___ -- 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 and Sangoma Cards PCI express compatibility
Hi to All, I dont know much about PCI express slots in newer Servers, my doubt is if the Digium and Sangoma PCI express cards, are compatible with the x8 PCI express slots that come in the HP Proliant ML150 G5 server. Thanks in advance. Ricardo ___ -- 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 and Sangoma Cards PCI express compatibility
Hi to All, I dont know much about PCI express slots in newer Servers, my doubt is if the Digium and Sangoma PCI express cards, are compatible with the x8 PCI express slots that come in the HP Proliant ML150 G5 server. Thanks in advance. Ricardo ___ -- 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] Inbound/Outbound undesired behavior
Hi to all, I need some help, I have an Asterisk Server in a small call center, for inbound calls I setup a Queue in queues.conf and their respective Agents in agents.conf, but when an Agent is calling out and a call is coming from PSTN the call is send to that agents which have a call in progress. How I can fix this in order to have only one call at a time. I think in limitonpeer and call-limit but the documentation says that the sip user can have 2 calls with this parameters (1 for inbound and 1 for Outbound and this is the behavior I don't want). Thanks in advance. Ricardo MR ___ -- 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] Softphone Framework or Libraries
Hi to all, I have a project for Customer Relationship Management interfaced with asterisk, I need send the CallerID to my application (via http or tcp/ip), When the phone rings I need to launch a pop-up windows to the Call Center Agent to display customer info, do you know a framework/libraries to make this, if is possible a softphone embedded into html page for the same function. I need to choice one to suit it to my needs Thanks in advance. Ricardo Melendez ___ -- 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] Write Asterisk CDR MySQL records to multiple servers
Hi to all, I actually have an asterisk server configured to write CDR mysql records in the same machine (localhost), but I want to write this records to another machine also in mysql at the same time, It is possible? It means that I want save the records in both machines. Thanks in advance. Ricardo Melendez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Custom Filename for Incoming Agent Calls
Hi, to all, I have configured 3 Inbound/outbound agents queues, I record
Outgoing calls with custom filename like
outgoing-${callerid(num)}-${EXTEN}-${TIMESTAMP}.gsm
but I need to record Incoming calls and asterisk by default add 13 digits
number to inbound recordings like Agent-001-1298375678-890.gsm, how I can
customize this filename recordings?
Thanks in advance.
Ricardo Melendez
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connect Asterisk PBX to Traditional PBX and retain functionality
Hi to All, I have a PBX (MAINPBX) from a Telecomm Provider, which have the feature to transfer calls (Incoming call - Answer - FLASH - Dial Number to transfer - Answer - FLASH+4) and the call is transferred, but I have the need to implement an internal ACD using Asterisk as the PBX, the trunks connected to my Asterisk FXO ports are Extensions of my MAINPBX (ex., 5437, 5440 etc), all features work fine, but I have the need to make asterisk act as a normal telephone when transferring calls, I need to release the line (FXO port in my Asterisk) and make the transfer via the MAINPBX feature. Otherwise I will use 2 lines of my Asterisk PBX to make the transfer and it reduce the incoming lines available for my ACD. It's possible send the commands FLASH, FLASH+4 using the incoming line to my MAINPBX via Asterisk like a normal telephone? Thanks in Advance. Ricardo Melendez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cascading Asterisk PBX
Hi to All, I have a PBX (MAINPBX) from a Telecomm Provider, which have the feature to transfer calls (Incoming call - Answer - FLASH - Dial Number to transfer - Answer - FLASH+4) and the call is transferred, but I have the need to implement an internal ACD using Asterisk as the PBX, the trunks connected to my Asterisk FXO ports are Extensions of my MAINPBX (ex., 5437, 5440 etc), all features work fine, but I have the need to make asterisk act as a normal telephone when transferring calls, I need to release the line (FXO port in my Asterisk) and make the transfer via the MAINPBX feature. Otherwise I will use 2 lines of my Asterisk PBX to make the transfer and it reduce the incoming lines available for my ACD. It's possible send the commands FLASH, FLASH+4 using the incoming line to my MAINPBX via Asterisk like a normal telephone? Thanks in Advance. Ricardo Melendez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Soundpoint (NO LINE)
Hi, I have just configure a Soundpoint 550 to work with Asterisk, it appear Registered to the asterisk server, and appear in asterisk console with SIP SHOW PEERS, and can receive calls, but when I try to dial, it launch a tone as if not line to dial, also can not stream audio to the other end when answer the received call. I have read the documentation but is very confuse. This phone have a Outbound Proxy (where I put the asterisk IP and port 5060) And 4 lines (where I configure the username/password and the Servers) Anyone can help me. Thanks in Advance. Atte. Ricardo Melendez ___ --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] Click to Talk Web Applications with Asterisk
Hi, I would like to develop a click to talk app to interface with asterisk, anyone know about some SDK/frameworks to implement this. Regards. Ricardo Meléndez Rosales ___ --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] PCI: UNABLE TO HANDLE 64-bit ADDRESS SPACE FOR
HI to All, I have an issue, when I connect a TDM04B and a TE212P on my server (Intel Entry Server board S3000AH) and boot RHEL4 the system launch this message: PCI: Unable to handle 64-bit address space for I think the problem is the TE212P because the TDM04B is detected as a Tiger Jet Network Modem. I have installed both cards in a PCI-X 64 bits Ing. Ricardo Meléndez Rosales Analista de Sistemas Transporte Escolar y de Personal S.A. de C.V. Tel.- 53404779 Movil 0445591911976 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] UNICALL MFC/R2 + Asterisk 1.4
Help I need to install asterisk 1.4.X using unicall, somebody can tell me which are the correct versions of spandsp, libunicall, libmfcr2, libsupertone, to install with asterisk 1.4, I have installed a prepatched version, but I need to know which are the correct releases Thanks in advance. Richard ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Big trouble with zap lines
Hi, I recently install asterisk 1.4 over fedora 4, I configured 12 zap channels like this In zaptel.conf fxsks=1-4 fxsks=5-8 fxsks=9-12 loadzone=us defaultzone=us in zapata.conf in channels section context=incoming signalling=fxs_ks channel = 1-4 channel = 5-8 channel = 9-12 when i run ztcfg -vv show 12 channels correctly configured whe i run zap show channels in asterisk console this show 12 channels correctly configured when i call to a zap channel like this Dial(zap/1/somenumber,15,r) in the console appear that asterisk is dialing trought this channel to this somenumber but in the line the call never go out nor in, the same happens when dial from outside, the line is ringing until the normal timeout. the PSTN lines used work normally whit normal hardphones (PSTN) zaptel, asterisk, zttool and ztcfg all never send any error message. What could be the problem?? Could be a damaged wildcard My card is wctdm2400p with 12 fxs ports in 3 modules thanks in advance ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Timestamp in recorded calls filename
Hi, I need to add the timestamp to the recorded call filename, I use this
variable ${TIMESTAMP} in the Monitor() function, but when I look for this
call, the TIMESTAMP is missing in the filename.
I try to export this as a environment variable but nothing changes.
Any help is welcome, thanks.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
