Re: [asterisk-users] How to enable inbound CLI forX-Lite/Asterisk phone.
Hi,
It is not showing any CLI information even after I have placed that
NoOp(${CALLERID(all)}) function for debugging. Following message was
displayed in debug:
Executing [EMAIL PROTECTED]:3]NoOp(ZAP/6-1,)
What should I do since it is critical to have the callee number.
Please give some more hints.
thanks
Syed Nasruddin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phil
Reynolds
Sent: Friday, October 10, 2008 3:10 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to enable inbound CLI
forX-Lite/Asterisk phone.
Quoting Syed Nasruddin [EMAIL PROTECTED]:
I am using asterisk 1.4.18. I am using it for inbound only call
center.
The SIP phones are X-Lite. Right now when a call is proxied by
Asterisk
to X-Lite the agent only sees asterisk written on its CLI screen. I
want
the agents to be able to view the callees number. Is there any way to
make this happen.
CLI showing as asterisk can indicate absent or withheld number. If
asterisk has it, it should pass it on to X-Lite without any special
settings.
Check to see if asterisk has CLI for the call by putting it in a NoOp in
the dialplan - NoOp(${CALLERID(all)}) would do. Watch asterisk with
verbose set to at least 3.
--
Phil Reynolds
mail: [EMAIL PROTECTED]
Web: http://www.tinsleyviaduct.com/phil/
Waltham 66, Emley Moor 69, Droitwich 79, Windows 95
This message was sent using IMP, the Internet Messaging Program.
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Re: [asterisk-users] How to enable inboundCLI forX-Lite/Asterisk phone.
How do we adjust zaptel and asterisk for CLI??. Is there some variable
to be set??.. kindly explain keeping in view your country settings this
will give me some hint.thanks
Syed Nasruddin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phil
Reynolds
Sent: Monday, October 13, 2008 7:47 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to enable inboundCLI
forX-Lite/Asterisk phone.
Quoting Syed Nasruddin [EMAIL PROTECTED]:
Hi,
It is not showing any CLI information even after I have placed that
NoOp(${CALLERID(all)}) function for debugging. Following message was
displayed in debug:
Executing [EMAIL PROTECTED]:3]NoOp(ZAP/6-1,)
What should I do since it is critical to have the callee number.
First off, is CLI information being presented on that line. If so, you
need to adjust zaptel and asterisk so that they see it - this is a
country-specific matter - what works for one may well not work for
another.
If CLI is not being presented on that line, you need to have that
enabled.
--
Phil Reynolds
mail: [EMAIL PROTECTED]
Web: http://www.tinsleyviaduct.com/phil/
Waltham 66, Emley Moor 69, Droitwich 79, Windows 95
This message was sent using IMP, the Internet Messaging Program.
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To UNSUBSCRIBE or update options visit:
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[asterisk-users] How to enable inbound CLI for X-Lite/Asterisk phone.
Hi, I am using asterisk 1.4.18. I am using it for inbound only call center. The SIP phones are X-Lite. Right now when a call is proxied by Asterisk to X-Lite the agent only sees asterisk written on its CLI screen. I want the agents to be able to view the callees number. Is there any way to make this happen. Regards Syed Nasruddin ___ -- 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] MixMonitor-Saving Recorded file with AgentId.
Hi, Is there any way of achieving what I have mentioned in my previous email. Scenario: I am recording all calls in queue. I want to save file in a way that I can identify the agent for whom the recording ahs been made. The saved file name should have something related to agent id or anything that I could relate to. Please give suggestions. I have checked agent.conf it dosent give much help on this thing. thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Syed Nasruddin Sent: Wednesday, September 03, 2008 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] MixMonitor-Saving Recorded file with AgentId. Hi, I am using asterisk 1.4.18. I am using Queues and recording all the calls to agents by using MixMonitor. There are 4 agents. I want to save recorded files with AgentId so that I can access recorded files of specific agent. e.g Agent Id.gsm please give hint abt it. thanks Syed Nasruddin ___ -- 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] MixMonitor-Saving Recorded file with AgentId.
Hi, I am using asterisk 1.4.18. I am using Queues and recording all the calls to agents by using MixMonitor. There are 4 agents. I want to save recorded files with AgentId so that I can access recorded files of specific agent. e.g Agent Id.gsm please give hint abt it. thanks Syed Nasruddin ___ -- 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
Re: [asterisk-users] Queue Penalties not working properly
I am using asterisk 1.4.18. I cant at this stage upgrade to any latest version. Linear strategy for queues is not in asterisk 1.4.18. I have to use ringall instead. Is it possible Disabling call-waiting for my agents only?? While other sip users have call waiting functionality. regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robin Rodriguez Sent: Tuesday, August 05, 2008 11:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly Syed Nasruddin wrote: Hi, Actully the way I want the penalties functionality to behave it is not doing it accordingly. I am right now using ringall. Set penalty 1 for one agent and 2 for secnd agent. All the calls come in and go to first agent#1 having penalty one. But the second call also go to agent#1 and start waiting for it to be free rather it should have gone to penalty two agent#2 I have added call-limit=1 for bot sip accounts. And started the services. Still find the status wrong. nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins Sent: Tuesday, August 05, 2008 7:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly On Tue, Aug 5, 2008 at 5:27 PM, Syed Nasruddin [EMAIL PROTECTED] wrote: Cannot i use ringall strategy with penalties??? Will rrmemory will fullfil my requirement?? rrmemory isn't ringall, it won't ring all members. But yes - you can use ringall with penalties. My requirements: 1. 10 Call Center Agents. 2. All the calls coming in will ALWAYS be routed to specific 5 agents, firstly. 3. IF ALL the first 5 agents are busy then ONLY then the call will be routed to next 5 Agents. Moreover why my queue status shows my agent as NOT IN USE while in fact it is busy answering the call?? What you are seeing is caused by status NOT IN USE. You have to set call-limit in sip.conf for all your phones, to any value, so that device states work correctly, and queue can know that those phones are busy. Now you probably can see in CLI that queue is sending second call to first agent(s). Regards, Atis Thanks Syed nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, August 05, 2008 5:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly On Tue, Aug 5, 2008 at 8:47 AM, Robin Rodriguez [EMAIL PROTECTED] wrote: Syed Nasruddin wrote: Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it ends up with agent 1, when secnd call comes it continue wait in queue and doesn't go to agent 2 and when agent one is free it goes to this agent. I have set penalties in queue.conf. I have monitered my queue and witnessed that my agent1 status shows Not In Use and Agent 2 also same status is this the reason behind this. I have copied my queue show results below.please help . how do I change this stauts problem callcenter*CLI queue show myqueue has 0 calls (max unlimited) in 'ringall' strategy (14s holdtime), W:0, C:2, A:0, SL:0.0% within 0s Members: SIP/1001 with penalty 1 (Not in use) has taken 2 calls (last was 2233 secs ago) SIP/1000 with penalty 2 (Not in use) has taken no calls yet No Callers Syed nasr You need to use the linear queue strategy, it is in 1.6 or there is a backport to 1.4 -- Robin Rodriguez VoIP/Telecom Engineer Atlantic.net 1-800-211-9496 Robin, round robin ___ -- 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 ___ -- 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 Very carefully reread the descriptions on penalties and queue strategies on voip-info.org, the first time I
[asterisk-users] Queue Penalties not working properly
Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it ends up with agent 1, when secnd call comes it continue wait in queue and doesn't go to agent 2 and when agent one is free it goes to this agent. I have set penalties in queue.conf. I have monitered my queue and witnessed that my agent1 status shows Not In Use and Agent 2 also same status is this the reason behind this. I have copied my queue show results below.please help . how do I change this stauts problem callcenter*CLI queue show myqueue has 0 calls (max unlimited) in 'ringall' strategy (14s holdtime), W:0, C:2, A:0, SL:0.0% within 0s Members: SIP/1001 with penalty 1 (Not in use) has taken 2 calls (last was 2233 secs ago) SIP/1000 with penalty 2 (Not in use) has taken no calls yet No Callers Syed nasr ___ -- 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
Re: [asterisk-users] Queue Penalties not working properly
Cannot i use ringall strategy with penalties??? Will rrmemory will fullfil my requirement?? My requirements: 1. 10 Call Center Agents. 2. All the calls coming in will ALWAYS be routed to specific 5 agents, firstly. 3. IF ALL the first 5 agents are busy then ONLY then the call will be routed to next 5 Agents. Moreover why my queue status shows my agent as NOT IN USE while in fact it is busy answering the call?? Thanks Syed nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, August 05, 2008 5:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly On Tue, Aug 5, 2008 at 8:47 AM, Robin Rodriguez [EMAIL PROTECTED] wrote: Syed Nasruddin wrote: Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it ends up with agent 1, when secnd call comes it continue wait in queue and doesn't go to agent 2 and when agent one is free it goes to this agent. I have set penalties in queue.conf. I have monitered my queue and witnessed that my agent1 status shows Not In Use and Agent 2 also same status is this the reason behind this. I have copied my queue show results below.please help . how do I change this stauts problem callcenter*CLI queue show myqueue has 0 calls (max unlimited) in 'ringall' strategy (14s holdtime), W:0, C:2, A:0, SL:0.0% within 0s Members: SIP/1001 with penalty 1 (Not in use) has taken 2 calls (last was 2233 secs ago) SIP/1000 with penalty 2 (Not in use) has taken no calls yet No Callers Syed nasr You need to use the linear queue strategy, it is in 1.6 or there is a backport to 1.4 -- Robin Rodriguez VoIP/Telecom Engineer Atlantic.net 1-800-211-9496 Robin, round robin ___ -- 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 ___ -- 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
Re: [asterisk-users] Queue Penalties not working properly
Hi, Actully the way I want the penalties functionality to behave it is not doing it accordingly. I am right now using ringall. Set penalty 1 for one agent and 2 for secnd agent. All the calls come in and go to first agent#1 having penalty one. But the second call also go to agent#1 and start waiting for it to be free rather it should have gone to penalty two agent#2 I have added call-limit=1 for bot sip accounts. And started the services. Still find the status wrong. nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins Sent: Tuesday, August 05, 2008 7:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly On Tue, Aug 5, 2008 at 5:27 PM, Syed Nasruddin [EMAIL PROTECTED] wrote: Cannot i use ringall strategy with penalties??? Will rrmemory will fullfil my requirement?? rrmemory isn't ringall, it won't ring all members. But yes - you can use ringall with penalties. My requirements: 1. 10 Call Center Agents. 2. All the calls coming in will ALWAYS be routed to specific 5 agents, firstly. 3. IF ALL the first 5 agents are busy then ONLY then the call will be routed to next 5 Agents. Moreover why my queue status shows my agent as NOT IN USE while in fact it is busy answering the call?? What you are seeing is caused by status NOT IN USE. You have to set call-limit in sip.conf for all your phones, to any value, so that device states work correctly, and queue can know that those phones are busy. Now you probably can see in CLI that queue is sending second call to first agent(s). Regards, Atis Thanks Syed nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, August 05, 2008 5:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly On Tue, Aug 5, 2008 at 8:47 AM, Robin Rodriguez [EMAIL PROTECTED] wrote: Syed Nasruddin wrote: Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it ends up with agent 1, when secnd call comes it continue wait in queue and doesn't go to agent 2 and when agent one is free it goes to this agent. I have set penalties in queue.conf. I have monitered my queue and witnessed that my agent1 status shows Not In Use and Agent 2 also same status is this the reason behind this. I have copied my queue show results below.please help . how do I change this stauts problem callcenter*CLI queue show myqueue has 0 calls (max unlimited) in 'ringall' strategy (14s holdtime), W:0, C:2, A:0, SL:0.0% within 0s Members: SIP/1001 with penalty 1 (Not in use) has taken 2 calls (last was 2233 secs ago) SIP/1000 with penalty 2 (Not in use) has taken no calls yet No Callers Syed nasr You need to use the linear queue strategy, it is in 1.6 or there is a backport to 1.4 -- Robin Rodriguez VoIP/Telecom Engineer Atlantic.net 1-800-211-9496 Robin, round robin ___ -- 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 ___ -- 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 -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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 ___ -- 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
Re: [asterisk-users] Asterisk Queues problem- URGENT
Hi all, Okay I have solved the problem. Actually the asterisk detected 24 Port FXO and numbered its ports. Since it has previously detcted ports 1-4 for FXS and ports 58 for FXO for my initial 8-port card. When I installed 24 port second card it numbered the new fxo ports from 9-32. uptill now fair enough. Now problem was when I physically inserted lines in to the Patch Panel 24 port of the new card I inserted the lines from port 1 - 10 (right now only ten lines added). The problem was solved by reinserting the lines in the patch panel from 9-18 since the ports from 1-8 are already detected by asterisk for the previous card. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Syed Nasruddin Sent: Monday, August 04, 2008 11:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Queues problem- URGENT Hi, Can anyone help me on this. I am really stuck.again defining the problem briefly.: 1. Second New card TDM240P added to machine. 2. Only FXO modules i.e 24 FXO. 3. Asterisk detected all the ports successfully and when I run module reload chan_zap.so it list allthe FXO ports correctly. 4. when I can on any of the newly added lines there is a clear ring on the orginators phone while no activity detetcted by asterisk. It just keep quiet. It looks like call is not being detected by the card to my asterisk. 5. 4 port FXO card which was previously installed is functioning properly only this new added card is causing problem. 6. I have 12 new lines and only one of the lines is generating below mentioned logs in asterisk: == Starting post polarity CID detection on channel 18 -- Starting simple switch on 'Zap/18-1' [Aug 4 11:09:29] NOTICE[12255]: chan_zap.c:6169 ss_thread: Got event 17 (Polarity Reversal)... [Aug 4 11:09:31] WARNING[12255]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/18-1' == Starting post polarity CID detection on channel 17 -- Starting simple switch on 'Zap/17-1' [Aug 4 11:09:35] ERROR[12256]: callerid.c:564 callerid_feed: fsk_serie made mylen 0 (-1) [Aug 4 11:09:35] WARNING[12256]: chan_zap.c:6202 ss_thread: CallerID feed failed: Success [Aug 4 11:09:35] WARNING[12256]: chan_zap.c:6215 ss_thread: CallerID returned with error on channel 'Zap/17-1' [Aug 4 11:09:37] WARNING[12256]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/17-1' Can anyone decipher this code??? What is happening?? Please give me some cluess to work on. In my Zapata.conf I have following two lines related to above logs: Cidsignalling= v23 Cidstart = polarity Please help./ Syed nasr (MONDAY 04/08/2008) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Syed Nasruddin Sent: Friday, August 01, 2008 8:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Queues problem Thanks, Yes that was the problem I have added joinempty=yes. It is now working,. Right now another critical problem has come up which I have mentioned in my previous email. I am copying the problem here again: was initially running only with one TDM800P card having 4FXO and 4 FXS port then I later added another 24 port FXO card. So now in total I have now 32 FXO ports for in coming calls. Card was successfully integerated and all the ports were detected by asterisk. Just few minutes back the POT lines were also ready and now I am getting additional errors which I am pasting here. starting simple switch on 'Zap/17-1'[Aug 1 19:00:26] ERROR[3416]: callerid.c:564 callerid_feed: fsk_s erie made mylen 0 (-1)[Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6202 ss_thread: Caller ID feed failed: Success [Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6215 ss_thread: Caller ID returned with error on channel 'Zap/17-1' [Aug 1 19:00:28] WARNING[3416]: chan_zap.c:6232 ss_thread: CID ti med out waiting for ring. Exiting simple switch-- Hungup 'Zap/17-1' -- Saved useragent X-Lite release 1002tx stamp 29712 for pee r 1001[Aug 1 19:18:29] NOTICE[3162]: chan_zap.c:6678 handle_init_event: Alarm cleared on channel 17 == Starting post polarity CID detection on channel 17-- Starting simple switch on 'Zap/17-1' [Aug 1 19:18:29] NOTICE[3582]: chan_zap.c:6169 ss_thread: Got event 4 (Alarm)... [Aug 1 19:18:31] WARNING[3582]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch Hungup 'Zap/17-1' Please help on this urgent. I cant upgrade right now since I am not confident abt upgrade procedure and any other problems occuring after that. This is my only production machine. thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Michelson Sent: Friday, August 01, 2008 7:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re
[asterisk-users] Customized Queuing Strategy
Hi All, I have this Call Center requirement to be implenetd: 1. 10 Call Center Agents. 2. All the calls coming in will ALWAYS be routed to specific 5 agents, firstly. 4. IF ALL the first 5 agents are busy then ONLY then the call will be routed to next 5 Agents. Kindly suggest some good easy strategy to handle this. thanks ___ -- 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
Re: [asterisk-users] Customized Queuing Strategy
Hi Thanks ALL for reply, If I use cascading queue will it do the trick?? The only problem is (as mentioned in below example) if a call enters testq and get answered then after hungup at the agent end only will the call will again enter the next queue which is testq2 as in this example.?? Moreover if I keep penalty 1 for all the first 5 agents and penalty 2 or higher for all the next 5 agents and implement ringall strategy will it do the same effect?? exten = 1589,1,Answer exten = 1589,2,Ringing exten = 1589,3,Wait(2) exten = 1589,4,Queue(testq|t|||45) exten = 1589,5,Queue(testq2|t|||45) exten = 1589,6,Hangup thanks in advance. Syed nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Monday, August 04, 2008 1:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Customized Queuing Strategy Syed Nasruddin wrote: 1. 10 Call Center Agents. 2. All the calls coming in will ALWAYS be routed to specific 5 agents, firstly. 4. IF ALL the first 5 agents are busy then ONLY then the call will be routed to next 5 Agents. Set up two queues. Call Queue() on the first queue - corresponding to #1 - with a rather strict timeout. Fall back on the second queue. More sophisticated strategies require either the modification of the source code for app_queue, or custom queue implementation in AGI. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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 ___ -- 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
Re: [asterisk-users] Customized Queuing Strategy
Dear Atis,
I am running in to syntax problem. Sorry only beginner level experience
of conditional checking:
exten = 1589,1,Answer
exten = 1589,2,Ringing
exten = 1589,3,Wait(2)
exten = 1589,4,Queue(testq|t|||45)
if (${QUEUESTATUS=) Hangup(); since I want to hangup if the caller
has already been catered by an agent and the caller hasnt hanged up, so
what status value should I look for. Moreover syntax of above
conditional statement is complete or something missin:
if (${QUEUESTATUS=) Hangup();
if above condition fails then the control must move to below lines
rather then getting hanged up.
exten = 1589,5,Queue(testq2|t|||45)
exten = 1589,6,Hangup
Thanks
Syed nasr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis
Lezdins
Sent: Monday, August 04, 2008 2:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Customized Queuing Strategy
Sorry for previous blank answer :)
On Mon, Aug 4, 2008 at 1:20 PM, Syed Nasruddin [EMAIL PROTECTED]
wrote:
Hi Thanks ALL for reply,
If I use cascading queue will it do the trick?? The only problem is
(as
mentioned in below example) if a call enters testq and get answered
then
after hungup at the agent end only will the call will again enter the
next queue which is testq2 as in this example.??
Check the QUEUESTATUS variable:
http://www.voip-info.org/wiki-Asterisk+cmd+Queue
Moreover if I keep penalty 1 for all the first 5 agents and penalty 2
or
higher for all the next 5 agents and implement ringall strategy will
it
do the same effect??
Yes
exten = 1589,1,Answer
exten = 1589,2,Ringing
exten = 1589,3,Wait(2)
exten = 1589,4,Queue(testq|t|||45)
if (${QUEUESTATUS=) Hangup();
exten = 1589,5,Queue(testq2|t|||45)
exten = 1589,6,Hangup
Regards,
Atis
thanks in advance.
Syed nasr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Balashov
Sent: Monday, August 04, 2008 1:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Customized Queuing Strategy
Syed Nasruddin wrote:
1. 10 Call Center Agents.
2. All the calls coming in will ALWAYS be routed to specific 5
agents,
firstly.
4. IF ALL the first 5 agents are busy then ONLY then the call will be
routed to next 5 Agents.
Set up two queues. Call Queue() on the first queue - corresponding to
#1 - with a rather strict timeout. Fall back on the second queue.
More sophisticated strategies require either the modification of the
source code for app_queue, or custom queue implementation in AGI.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Customized Queuing Strategy
Thanks Atis and steve.
I think I will have it running tomorrow.
Thanks a lot.
Syed Nasruddin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Monday, August 04, 2008 6:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Customized Queuing Strategy
On Mon, Aug 4, 2008 at 9:24 AM, Atis Lezdins [EMAIL PROTECTED] wrote:
On Mon, Aug 4, 2008 at 2:59 PM, Syed Nasruddin [EMAIL PROTECTED]
wrote:
Dear Atis,
I am running in to syntax problem. Sorry only beginner level
experience
of conditional checking:
Yes, sorry for that, i just wrote it quickly and didn't checked
expression. Also, i didn't wrote in .conf format, as it's been a long
time since i wrote that.
exten = 1589,1,Answer
exten = 1589,2,Ringing
exten = 1589,3,Wait(2)
exten = 1589,4,Queue(testq|t|||45)
if (${QUEUESTATUS=) Hangup(); since I want to hangup if the
caller
has already been catered by an agent and the caller hasnt hanged up,
so
what status value should I look for. Moreover syntax of above
conditional statement is complete or something missin:
Exactly, if call has been handled by agent, QUEUESTATUS will be empty.
Otherwise it will be LEAVEUNAVAIL or something like that (not empty)
if (${QUEUESTATUS=) Hangup();
if above condition fails then the control must move to below lines
rather then getting hanged up.
ok, i'll try:
exten = 1589,5,GotoIf($[${QUEUESTATUS}=]?exit)
exten = 1589,5,Queue(testq2|t|||45)
rename to priority 6
exten = 1589,6,Hangup
rename to priority 7 and add label exit:
exten = 1589,7(exit),Hangup
But as said before, you can also use penalties of members. Next
penalty is only chosen if nobody with smallest penalty can't be
dialed. Plus, there will also be advantage that if you dial member for
15 seconds, and at first there is noone with penalty 1 available -
queue will call somebody with penalty 2. Then, if dialed member(s)
don't answer, queue will again try somebody with penalty 1 first.
Regards,
Atis
Plus, if/when you grow, you can use AMI to dynamically change
penalties. This can be a great advantage for business logic.
Thanks,
Steve T
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Re: [asterisk-users] Asterisk Queues problem- URGENT
Hi, Can anyone help me on this. I am really stuck.again defining the problem briefly.: 1. Second New card TDM240P added to machine. 2. Only FXO modules i.e 24 FXO. 3. Asterisk detected all the ports successfully and when I run module reload chan_zap.so it list allthe FXO ports correctly. 4. when I can on any of the newly added lines there is a clear ring on the orginators phone while no activity detetcted by asterisk. It just keep quiet. It looks like call is not being detected by the card to my asterisk. 5. 4 port FXO card which was previously installed is functioning properly only this new added card is causing problem. 6. I have 12 new lines and only one of the lines is generating below mentioned logs in asterisk: == Starting post polarity CID detection on channel 18 -- Starting simple switch on 'Zap/18-1' [Aug 4 11:09:29] NOTICE[12255]: chan_zap.c:6169 ss_thread: Got event 17 (Polarity Reversal)... [Aug 4 11:09:31] WARNING[12255]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/18-1' == Starting post polarity CID detection on channel 17 -- Starting simple switch on 'Zap/17-1' [Aug 4 11:09:35] ERROR[12256]: callerid.c:564 callerid_feed: fsk_serie made mylen 0 (-1) [Aug 4 11:09:35] WARNING[12256]: chan_zap.c:6202 ss_thread: CallerID feed failed: Success [Aug 4 11:09:35] WARNING[12256]: chan_zap.c:6215 ss_thread: CallerID returned with error on channel 'Zap/17-1' [Aug 4 11:09:37] WARNING[12256]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/17-1' Can anyone decipher this code??? What is happening?? Please give me some cluess to work on. In my Zapata.conf I have following two lines related to above logs: Cidsignalling= v23 Cidstart = polarity Please help./ Syed nasr (MONDAY 04/08/2008) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Syed Nasruddin Sent: Friday, August 01, 2008 8:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Queues problem Thanks, Yes that was the problem I have added joinempty=yes. It is now working,. Right now another critical problem has come up which I have mentioned in my previous email. I am copying the problem here again: was initially running only with one TDM800P card having 4FXO and 4 FXS port then I later added another 24 port FXO card. So now in total I have now 32 FXO ports for in coming calls. Card was successfully integerated and all the ports were detected by asterisk. Just few minutes back the POT lines were also ready and now I am getting additional errors which I am pasting here. starting simple switch on 'Zap/17-1'[Aug 1 19:00:26] ERROR[3416]: callerid.c:564 callerid_feed: fsk_s erie made mylen 0 (-1)[Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6202 ss_thread: Caller ID feed failed: Success [Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6215 ss_thread: Caller ID returned with error on channel 'Zap/17-1' [Aug 1 19:00:28] WARNING[3416]: chan_zap.c:6232 ss_thread: CID ti med out waiting for ring. Exiting simple switch-- Hungup 'Zap/17-1' -- Saved useragent X-Lite release 1002tx stamp 29712 for pee r 1001[Aug 1 19:18:29] NOTICE[3162]: chan_zap.c:6678 handle_init_event: Alarm cleared on channel 17 == Starting post polarity CID detection on channel 17-- Starting simple switch on 'Zap/17-1' [Aug 1 19:18:29] NOTICE[3582]: chan_zap.c:6169 ss_thread: Got event 4 (Alarm)... [Aug 1 19:18:31] WARNING[3582]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch Hungup 'Zap/17-1' Please help on this urgent. I cant upgrade right now since I am not confident abt upgrade procedure and any other problems occuring after that. This is my only production machine. thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Michelson Sent: Friday, August 01, 2008 7:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Queues problem Syed Nasruddin wrote: Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: _app_queue.c:3939 queue_exec: unable to join queue myqueue_ In extension file: Queue(myqueue|t|||120) And my agents are joining in following manner: Exten = 1001,1,AgentLogin(SIP/1001) Exten = 1000,1,AgentLogin(SIP/1000) One more thing my asterisk successfully captures the call , it plays music on hold but when it starts to push the call in queue it gives out this error. Any one help me out. It's a production machine. Thanks Syed nasr
[asterisk-users] Asterisk Queues problem
Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: app_queue.c:3939 queue_exec: unable to join queue myqueue In extension file: Queue(myqueue|t|||120) And my agents are joining in following manner: Exten = 1001,1,AgentLogin(SIP/1001) Exten = 1000,1,AgentLogin(SIP/1000) One more thing my asterisk successfully captures the call , it plays music on hold but when it starts to push the call in queue it gives out this error. Any one help me out. It's a production machine. Thanks Syed nasr ___ -- 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
Re: [asterisk-users] Asterisk Queues problem
Hi, I was initially running only with one TDM800P card having 4FXO and 4 FXS port then I later added another 24 port FXO card. So now in total I have now 32 FXO ports for in coming calls. Card was successfully integerated and all the ports were detected by asterisk. Just few minutes back the POT lines were also ready and now I am getting additional errors which I am pasting here. starting simple switch on 'Zap/17-1'[Aug 1 19:00:26] ERROR[3416]: callerid.c:564 callerid_feed: fsk_s erie made mylen 0 (-1)[Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6202 ss_thread: Caller ID feed failed: Success [Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6215 ss_thread: Caller ID returned with error on channel 'Zap/17-1' [Aug 1 19:00:28] WARNING[3416]: chan_zap.c:6232 ss_thread: CID ti med out waiting for ring. Exiting simple switch-- Hungup 'Zap/17-1' -- Saved useragent X-Lite release 1002tx stamp 29712 for pee r 1001[Aug 1 19:18:29] NOTICE[3162]: chan_zap.c:6678 handle_init_event: Alarm cleared on channel 17 == Starting post polarity CID detection on channel 17-- Starting simple switch on 'Zap/17-1' [Aug 1 19:18:29] NOTICE[3582]: chan_zap.c:6169 ss_thread: Got eve nt 4 (Alarm)... [Aug 1 19:18:31] WARNING[3582]: chan_zap.c:6232 ss_thread: CID ti med out waiting for ring. Exiting simple switch * Hungup 'Zap/17-1' Kindly give me a hint abt what is happening. And also why my agents are not getting in the queues. Thanks for quick reply. Syed nasr From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Syed Nasruddin Sent: Friday, August 01, 2008 6:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Queues problem Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: app_queue.c:3939 queue_exec: unable to join queue myqueue In extension file: Queue(myqueue|t|||120) And my agents are joining in following manner: Exten = 1001,1,AgentLogin(SIP/1001) Exten = 1000,1,AgentLogin(SIP/1000) One more thing my asterisk successfully captures the call , it plays music on hold but when it starts to push the call in queue it gives out this error. Any one help me out. It's a production machine. Thanks Syed nasr ___ -- 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
Re: [asterisk-users] Asterisk Queues problem
Thanks, Yes that was the problem I have added joinempty=yes. It is now working,. Right now another critical problem has come up which I have mentioned in my previous email. I am copying the problem here again: was initially running only with one TDM800P card having 4FXO and 4 FXS port then I later added another 24 port FXO card. So now in total I have now 32 FXO ports for in coming calls. Card was successfully integerated and all the ports were detected by asterisk. Just few minutes back the POT lines were also ready and now I am getting additional errors which I am pasting here. starting simple switch on 'Zap/17-1'[Aug 1 19:00:26] ERROR[3416]: callerid.c:564 callerid_feed: fsk_s erie made mylen 0 (-1)[Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6202 ss_thread: Caller ID feed failed: Success [Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6215 ss_thread: Caller ID returned with error on channel 'Zap/17-1' [Aug 1 19:00:28] WARNING[3416]: chan_zap.c:6232 ss_thread: CID ti med out waiting for ring. Exiting simple switch-- Hungup 'Zap/17-1' -- Saved useragent X-Lite release 1002tx stamp 29712 for pee r 1001[Aug 1 19:18:29] NOTICE[3162]: chan_zap.c:6678 handle_init_event: Alarm cleared on channel 17 == Starting post polarity CID detection on channel 17-- Starting simple switch on 'Zap/17-1' [Aug 1 19:18:29] NOTICE[3582]: chan_zap.c:6169 ss_thread: Got event 4 (Alarm)... [Aug 1 19:18:31] WARNING[3582]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch Hungup 'Zap/17-1' Please help on this urgent. I cant upgrade right now since I am not confident abt upgrade procedure and any other problems occuring after that. This is my only production machine. thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Michelson Sent: Friday, August 01, 2008 7:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Queues problem Syed Nasruddin wrote: Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: _app_queue.c:3939 queue_exec: unable to join queue myqueue_ In extension file: Queue(myqueue|t|||120) And my agents are joining in following manner: Exten = 1001,1,AgentLogin(SIP/1001) Exten = 1000,1,AgentLogin(SIP/1000) One more thing my asterisk successfully captures the call , it plays music on hold but when it starts to push the call in queue it gives out this error. Any one help me out. It's a production machine. Thanks Syed nasr When diagnosing this sort of issue, it is a good idea to check the value of QUEUESTATUS to see why the caller could not enter the queue. The most common reason for a caller to not join the queue is because joinempty=no is set in queues.conf (if you do not have joinempty set at all, then it defaults to no). This setting causes callers attempting to join a queue to not be able to if the queue is empty or if all the queue members are paused or have an invalid device state. Another possibility is that you have a maximum length set on the queue and so no more callers can join because the queue is full. My suggestion is to see what the QUEUESTATUS is. If the status is JOINEMPTY, then you can issue a queue show command on the CLI to see what the current states of your queue members are. It may be as easy to fix as setting joinempty=yes in queues.conf. If the status is something else, though, then a different fix may be in order instead. Mark Michelson ___ -- 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 ___ -- 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
Re: [asterisk-users] How to integerate 2 TDM cards on same machine.
Thanks Noah. It is now properly running. Thanks again regards Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Tuesday, July 15, 2008 9:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to integerate 2 TDM cards on same machine. Hi Syed - zttool shows that TDM800P is loaded first and TDM2401E is loaded second. now problem is ports are not being configured by asterisk. i have done following changes in two files zaptel.onf and zapata.conf. zaptel.conf loadzone=us, defaultzone=us, fxoks=1-4, fxsks=5-8, fxsks=9-32(or should this be fxoks???) zapata.conf signalling=fxoks channels =1-4 signalling=fxsks channels = 5-8 signalling=fxsks channels = 9-32 please see the bold lines. since FXO ports use FXS signalling so i used fxsks. is this right or wrong. are these changes have to be made in both the files as i have done or only in zaptel.conf waiting for information Almost there. Your zaptel.conf is correct (sorry I gave you the wrong signalling before). In zapata.conf, your signalling lines should look like: signalling=fxo_ks channels = 1-4 signalling=fxs_ks channels = 5-32 - Noah ___ -- 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 ___ -- 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] How to integerate 2 TDM cards on same machine.
Hi, I have been using single TDM800P card. It is a small card with 4FXO and 4FXS ports. I have been using it for sometime without any problem. I am using Asterisk 1.4.18.1. Now due to greater requirement to handle more calls our office has bought another larger card TDM2401E which has 24 FXO ports. I have installed it on the same machine. Would like to know following about its configuration. 1. Same Zaptel Driver will be used which is catering for my TDM800P card?? 2. My zaptel.conf has following current config: loadzone=us, defaultzone=us, fxoks=1-4, fxsks=5-8. What changes do I have to make to this file to accommodate 24 FXO ports of another card. 3. My Zapata.conf has following config: Signaling=fxo_ks . Channel = 1-4 Signaling=fxs_ks Channel = 5-8 Can you please tell me what changed to do with this file in order to accomdate 24-FXO port card. 4. Thanks for your help. Regards Syed Nasruddin ___ -- 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
Re: [asterisk-users] How to integerate 2 TDM cards on same machine.
thanks Noah, zttool shows that TDM800P is loaded first and TDM2401E is loaded second. now problem is ports are not being configured by asterisk. i have done following changes in two files zaptel.onf and zapata.conf. zaptel.conf loadzone=us, defaultzone=us, fxoks=1-4, fxsks=5-8, fxsks=9-32(or should this be fxoks???) zapata.conf signalling=fxoks channels =1-4 signalling=fxsks channels = 5-8 signalling=fxsks channels = 9-32 please see the bold lines. since FXO ports use FXS signalling so i used fxsks. is this right or wrong. are these changes have to be made in both the files as i have done or only in zaptel.conf waiting for information thanks syed nasruddin From: [EMAIL PROTECTED] on behalf of Noah Miller Sent: منگل 7/15/2008 12:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to integerate 2 TDM cards on same machine. Hi Syed - I have been using single TDM800P card. It is a small card with 4FXO and 4FXS ports. I have been using it for sometime without any problem. I am using Asterisk 1.4.18.1. Now due to greater requirement to handle more calls our office has bought another larger card TDM2401E which has 24 FXO ports. I have installed it on the same machine. Would like to know following about its configuration. Same Zaptel Driver will be used which is catering for my TDM800P card?? My zaptel.conf has following current config: loadzone=us, defaultzone=us, fxoks=1-4, fxsks=5-8. Just add in the extra channels: fxoks=9-32. Be sure to check the order the cards are loading with zttool. If the 2401E is loading first, it will actually be channels 1-24, and the 800 will be channels 25-32. Also, test to make sure your machine is capable of this setup. Each of these cards will generate 1000 interrupts per second. Most modern motherboards should be able to handle this, but some older ones may choke under this load. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ -- 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
Re: [asterisk-users] Call Center
Dear Sherwood,
Thanks.
Just three questions:
1. Will I be needing Apache or Asterk-stat will handle itself?
2. Are there How-tos for integerating asterisk-stat with asterisk?
3. My Recordings are being saved in the default folder i.e:
/var/spool/asterisk/monitor/ in .gsm format. When I wish to listen to a
particular recording I first convert it with SOX utility into .wav
format and then listen it. Will this also be automated so that when I
select a recording and try to listen it will be in right format.
Thanks again.
Syed Nasruddin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: Tuesday, June 17, 2008 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Center
Syed Nasruddin wrote:
Dear Sherwood,
I am also using Asterisk Call Center Setup in my office with voice
recording. The only thing I am unable to setup is web based call
recording (CDR) access. From your email I think you have configured
such
a thing can you please share with me how can I also setup this
solution.
I know how to run and install Apache. Don't know abt PostgreSQL.
However
can do it if you can define some steps.
And also how to integrate this all PostgreSQl+Apache+Web Based Links
to
Recordings. It will be a great help.
regards
Syed Nasruddin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: Tuesday, June 17, 2008 5:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Center
broadband Voice wrote:
Is anyone using Asterisk as a call center. I want to be able to set
it
up for my office line, when calls come in after 7:00pm Est want a
recording to says the office is closed and have about 5 phones that I
want to use as an agent. Can anyone share their implementation?
Thanks.
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There's a ton of us on here who have installations in call centers.
What
would you like to know?
I currently have a Dual AMD64 2.4Ghz (Dual Cores on each) with 4GB RAM
running a Tormenta 2 and a Digium 407. Two T1s going to a PRI, 12 FXO
channels in a Rhino modular channel bank (all on the Digium card), and
2
24 port adtran total access channel banks running on the Tormenta. The
Adtrans drive the 40 analog phones for the sales floor, and we have 25
SIP phones. All phone conversations are recording by Asterisk and are
converted from GSM to Speex post-call by speexenc. We also run
PostgreSQL and Apache on the same system to serve CDRs with links to
recordings.
Anything else you'd like to know?
Syed,
What I did for a quick and dirty solution was install asterisk-stats and
modify the source code to include a link to the unique filename of the
recording (I use ${UNIQUEID}). This has worked just fine for our 75 or
so phone setup :) IIRC we found asterisk-stats on voip-info.org. We
just used that instead of creating an in house CDR web app, since the
client just needed a basic interface to look up calls and pull the
recordings.
If you'd like more information just let me know.
--
Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL PROTECTED]
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Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution.
Thanks for the link. I think I will be using this product. Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gavin Henry Sent: Saturday, June 14, 2008 1:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. 2008/6/12 Syed Nasruddin [EMAIL PROTECTED]: HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. When the call ends, the recording should stop. Our clients use this for E1 Pri: http://www.voicetronix.com/logger.htm Not sure if there is a analogue solution. -- http://www.suretecsystems.com/services/openldap/ ___ -- 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] Call Center
Dear Sherwood, I am also using Asterisk Call Center Setup in my office with voice recording. The only thing I am unable to setup is web based call recording (CDR) access. From your email I think you have configured such a thing can you please share with me how can I also setup this solution. I know how to run and install Apache. Don't know abt PostgreSQL. However can do it if you can define some steps. And also how to integrate this all PostgreSQl+Apache+Web Based Links to Recordings. It will be a great help. regards Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Tuesday, June 17, 2008 5:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Center broadband Voice wrote: Is anyone using Asterisk as a call center. I want to be able to set it up for my office line, when calls come in after 7:00pm Est want a recording to says the office is closed and have about 5 phones that I want to use as an agent. Can anyone share their implementation? 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 There's a ton of us on here who have installations in call centers. What would you like to know? I currently have a Dual AMD64 2.4Ghz (Dual Cores on each) with 4GB RAM running a Tormenta 2 and a Digium 407. Two T1s going to a PRI, 12 FXO channels in a Rhino modular channel bank (all on the Digium card), and 2 24 port adtran total access channel banks running on the Tormenta. The Adtrans drive the 40 analog phones for the sales floor, and we have 25 SIP phones. All phone conversations are recording by Asterisk and are converted from GSM to Speex post-call by speexenc. We also run PostgreSQL and Apache on the same system to serve CDRs with links to recordings. Anything else you'd like to know? -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- 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] Using Asterisk Only as Voice RecordingSolution.
Dear PaulH, I have 5 PSTN Lines going into my legacy PBX. There is an active IVR present on legacy PBX which the client wants to keep. So what I have to do is: 1. Install 10 Port i.e 5-FXO/5-FXS card on asterisk machine. 2. Insert All those PSTN directly to my 5-Port FXO. 3. Take out 5-FXS Port lines and insert them into my legacy PBX. 4. Since as I mentioned previously that my client wants to keep its IVR intact on its Legacy system so I will not be handling IVR in my Asterisk Dialplan. 5. when the call arrives at asteriskwhat should I do?? Should I simply call Dial(FXS channel) or something else. Kindly provide some info regarding Step 5. Thanks Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Friday, June 13, 2008 9:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. Basically, you run the phone lines into the asterisk box, then out of the Asterisk system into the PABX. This works reasonably well, and gives you the option to migrate to a full asterisk setup in the future. PaulH Syed Nasruddin wrote: Thanks Steve, How I can use it Asterisk as Man In The Middle. Since we have to keep our Native PBX intact and functioning but only thing it doesn't handle is Voice Recording. I thought if I can get some Channel Variable or some system generated event regarding OFF-HOOK and ON-HOOK condition through Asterisk I will easily handle this requirement. It will be a great help if you can elaborate how I can use asterisk as man-in-the-middle configuration along with my current PBX. Thanks a lot for your prompt response Syed Nasruddin (CISSP) Assistant Manager - Systems National Commodity Exchange Limited 9th Floor, PIC Towers 32-A Lalazar Drive M.T. Khan Road Karachi Phone: 111623623 ext 217 Fax: 5611263 Web: www.ncel.com.pk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 12, 2008 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. On Thu, Jun 12, 2008 at 11:16 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. When the call ends, the recording should stop. Problem being faced by me is this that I am able to catch the call in my diaplan and initialize MixMonitor but since my diaplan never detects OFF-HOOK/ON_HOOK condition it continue to wait and finally hang-up while in actual the call is running through our PBX. Is there any channel variable or any other mechanism by which I can accomplish this task? Since i will not be using any Dial() or similar application I will be needing some kind of OFF-HOOK trigger/Event in my dialplan. Your help will be highly appreciated. regards Syed Nasruddin It may not be possible to do this in parallel the way you are trying now. In series should be a simple task. Just pass the call through Asterisk as the man in the middle, the dialplan will be very simple. Thanks, Steve T ___ -- 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 ___ -- 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] Using Asterisk Only as Voice RecordingSolution.
Thanks Steve, Sure, although I would have loved to see a pre-config dialplan:. Thanks for the tip. I think it will help me through. Best Regards Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, June 13, 2008 4:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. Step five: Profit ;-) I am not going to write your dialplan for you but here is a clue. http://www.voip-info.org/wiki/view/Asterisk+legacy+integration Of those various setups, you can extract what you need. Thanks, Steve T On Fri, Jun 13, 2008 at 8:05 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: Dear PaulH, I have 5 PSTN Lines going into my legacy PBX. There is an active IVR present on legacy PBX which the client wants to keep. So what I have to do is: 1. Install 10 Port i.e 5-FXO/5-FXS card on asterisk machine. 2. Insert All those PSTN directly to my 5-Port FXO. 3. Take out 5-FXS Port lines and insert them into my legacy PBX. 4. Since as I mentioned previously that my client wants to keep its IVR intact on its Legacy system so I will not be handling IVR in my Asterisk Dialplan. 5. when the call arrives at asteriskwhat should I do?? Should I simply call Dial(FXS channel) or something else. Kindly provide some info regarding Step 5. Thanks Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Friday, June 13, 2008 9:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. Basically, you run the phone lines into the asterisk box, then out of the Asterisk system into the PABX. This works reasonably well, and gives you the option to migrate to a full asterisk setup in the future. PaulH Syed Nasruddin wrote: Thanks Steve, How I can use it Asterisk as Man In The Middle. Since we have to keep our Native PBX intact and functioning but only thing it doesn't handle is Voice Recording. I thought if I can get some Channel Variable or some system generated event regarding OFF-HOOK and ON-HOOK condition through Asterisk I will easily handle this requirement. It will be a great help if you can elaborate how I can use asterisk as man-in-the-middle configuration along with my current PBX. Thanks a lot for your prompt response Syed Nasruddin (CISSP) Assistant Manager - Systems National Commodity Exchange Limited 9th Floor, PIC Towers 32-A Lalazar Drive M.T. Khan Road Karachi Phone: 111623623 ext 217 Fax: 5611263 Web: www.ncel.com.pk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 12, 2008 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. On Thu, Jun 12, 2008 at 11:16 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. When the call ends, the recording should stop. Problem being faced by me is this that I am able to catch the call in my diaplan and initialize MixMonitor but since my diaplan never detects OFF-HOOK/ON_HOOK condition it continue to wait and finally hang-up while in actual the call is running through our PBX. Is there any channel variable or any other mechanism by which I can accomplish this task? Since i will not be using any Dial() or similar application I will be needing some kind of OFF-HOOK trigger/Event in my dialplan. Your help will be highly appreciated. regards Syed Nasruddin It may not be possible to do this in parallel the way you are trying now. In series should be a simple task. Just pass the call through Asterisk as the man in the middle, the dialplan will be very simple. Thanks, Steve T ___ -- 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
[asterisk-users] Using Asterisk Only as Voice Recording Solution.
HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: 1. Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. 2. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. 3. When the call ends, the recording should stop. Problem being faced by me is this that I am able to catch the call in my diaplan and initialize MixMonitor but since my diaplan never detects OFF-HOOK/ON_HOOK condition it continue to wait and finally hang-up while in actual the call is running through our PBX. Is there any channel variable or any other mechanism by which I can accomplish this task? Since i will not be using any Dial() or similar application I will be needing some kind of OFF-HOOK trigger/Event in my dialplan. Your help will be highly appreciated. regards Syed Nasruddin ___ -- 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] Using Asterisk Only as Voice RecordingSolution.
Thanks Steve, How I can use it Asterisk as Man In The Middle. Since we have to keep our Native PBX intact and functioning but only thing it doesn't handle is Voice Recording. I thought if I can get some Channel Variable or some system generated event regarding OFF-HOOK and ON-HOOK condition through Asterisk I will easily handle this requirement. It will be a great help if you can elaborate how I can use asterisk as man-in-the-middle configuration along with my current PBX. Thanks a lot for your prompt response Syed Nasruddin (CISSP) Assistant Manager - Systems National Commodity Exchange Limited 9th Floor, PIC Towers 32-A Lalazar Drive M.T. Khan Road Karachi Phone: 111623623 ext 217 Fax: 5611263 Web: www.ncel.com.pk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 12, 2008 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. On Thu, Jun 12, 2008 at 11:16 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. When the call ends, the recording should stop. Problem being faced by me is this that I am able to catch the call in my diaplan and initialize MixMonitor but since my diaplan never detects OFF-HOOK/ON_HOOK condition it continue to wait and finally hang-up while in actual the call is running through our PBX. Is there any channel variable or any other mechanism by which I can accomplish this task? Since i will not be using any Dial() or similar application I will be needing some kind of OFF-HOOK trigger/Event in my dialplan. Your help will be highly appreciated. regards Syed Nasruddin It may not be possible to do this in parallel the way you are trying now. In series should be a simple task. Just pass the call through Asterisk as the man in the middle, the dialplan will be very simple. Thanks, Steve T ___ -- 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
