Re: [asterisk-users] Problems configuring a PRI...
Here is my configuration with Global Crossing. Hope this helps. Zaptel.co # Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" B8ZS/ESF ClockSource span=1,1,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 Zapata.conf mode=mixed signalling=pri_cpe context=incoming-att group=1 channel => 1-23 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Hoff Sent: Tuesday, June 10, 2008 5:23 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Problems configuring a PRI... I'm trying to get a Qwest PRI configured and working with my lab Asterisk server. They said that the switchtype is 5ess and the signaling is pri_cpe. My entries into zaptel.conf are: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us channels=1-23 And my entries in zapata.conf are: language=en context=telco-incoming switchtype=5ess signalling=pri_cpe rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no group = 1 switchtype = 5ess signalling = pri_cpe group = 1 channel => 1-23 I'm not able to make/receive calls, and the error I'm receiving is: [Jun 10 11:32:37] WARNING[31768]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! == Primary D-Channel on span 1 down Qwest says that the PRI is fine. I have a green light on the PRI card. Help! ___ Chris Hoff Telecommunications Administrator SEI LLC Voice +1 701 298 8865 Ext 2189 Mobile +1 701 361 5976 Fax +1 701 298 8860 Email [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
Re: [asterisk-users] MiixMonitor filename for queue calls.
I am using the following entry to define my filename exten => 8484,1,Set(MONITOR_FILENAME=QUEUE-NOC-${CALLERID(NUM)}-${STRFTIME(${EPOCH},G MT+8,%C%y%m%d%H%M)}) This will display QUEUE-NOC (Caller ID number) (and time stamp) I would also like to add the answering Agent ID to the file name. Any idea what this variable name is? Thank you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Smith Sent: Saturday, June 07, 2008 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MiixMonitor filename for queue calls. Hi Ed, Glad to see you figured out your problem. I'm not sure what the differences are between your config and mine, but maybe this will help others too. I add and remove my agents from the queue. So my agents.conf file is just the presistentagens=yes. Also I just run the command in the dial plan like below which saved mine items just fine. No configurations in the queue.conf file for the monitor type. exten => 852,n,MixMonitor(/mercury/recordings/holding/${UNIQUEID}.gsm|b|) From there, in the hangup extension, I run a php script to take the CDR record and the file (rename it of course to queue-extension-callerid-callid-timestamp.gsm), and place it into the agents folder and the database for our agents/supervisors to review or download them. Kevin Ed Nunez wrote: > > Can anyone give me input on the following issue? > > > > I have a queue with MixMonitor enabled. > > This is also enabled in agents.conf. > > On my extensions.conf, I am setting the monitor filename as fillows, > although I see the filename as desired in the console as I make my > test call, the system is only using the default file name to save the > mixmonitor file (agented + uniqueID) > > > > Agents.conf > > > > [general] > > persistentagents=yes > > > > [agents] > > maxlogintries=3 > > musiconhold => default > > updatecdr=yes > > recordagentcalls=yes > > recordformat=wav49 > > urlprefix=http://pbx.netoneint.com/calls/ > > savecallsin=/var/calls > > > > agent => 1000,1000,Ed Test1 > > agent => 1001,1001,Ed Test2 > > > > > > queues.conf > > > > [noi-noc] > > monitor-format = wav49 > > monitor-type = MixMonitor > > > > member => Agent/1001 > > member => Agent/1000 > > > > > > extensions.conf > > > > exten => > 8484,1,Set(MONITOR_FILENAME=QUEUE-NOC-${CALLERID(NUM)}-${STRFTIME(${EPOCH) > > exten => 8484,1,answer > > exten => 8484,2,Queue(noi-noc) > > > > > > Console output > > > > -- Executing [EMAIL PROTECTED]:1] Set("Zap/1-1", > "MONITOR_FILENAME=QUEUE-NOC-4073844200-Fri Jun 6 15:06:38 2008") in > new stack > > -- Executing [EMAIL PROTECTED]:2] Queue("Zap/1-1", "noi-noc") in > new stack > > -- Started music on hold, class 'default', on Zap/1-1 > > -- outgoing agentcall, to agent '1001', on > 'Local/[EMAIL PROTECTED],1' > > -- Called Agent/1001 > > -- Executing [EMAIL PROTECTED]:1] > Dial("Local/[EMAIL PROTECTED],2", "SIP/1658") in new stack > > -- Called 1658 > > -- SIP/1658-087e7610 is ringing > > -- Agent/1001 is ringing > > -- SIP/1658-087e7610 answered Local/[EMAIL PROTECTED],2 > > -- Agent/1001 answered Zap/1-1 > > -- Stopped music on hold on Zap/1-1 > > [Jun 6 15:06:40] WARNING[3976]: app_queue.c:3014 try_calling: The > device state of this queue member, Agent/1001, is still 'Not in Use' > when it probably should not be! Please check UPGRADE.txt for correct > configuration settings. > > == Begin MixMonitor Recording Zap/1-1 > > == Spawn extension (numberplan-custom-3, 1658, 1) exited non-zero on > 'Local/[EMAIL PROTECTED],2' > > == Spawn extension (incoming-att, 8484, 2) exited non-zero on 'Zap/1-1' > > -- Hungup 'Zap/1-1' > > == End MixMonitor Recording Zap/1-1 > > > > > > > > > > > > > > ___ > -- 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 -- Kevin Smith --- Mercury Network Technical Support Phone: 989.837.3790 Toll Free: 888.866.4638 www.mercury.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 ___ -- 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] MiixMonitor filename for queue calls.
I have found the answer to my question. For anyone intrested, the system was saving the file with my desired filename in the default /monitor sub-directory and was also saving a second copy of the file in the /calls sub-directory. I commented out the ;recordagentcalls=yes Line in agents.con and this stoped the system from recording the seconfd file in the /calls sub-directory. Hope this information may be usefull to someone. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nunez Sent: Friday, June 06, 2008 3:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'; [EMAIL PROTECTED] Subject: [asterisk-users] MiixMonitor filename for queue calls. Can anyone give me input on the following issue? I have a queue with MixMonitor enabled. This is also enabled in agents.conf. On my extensions.conf, I am setting the monitor filename as fillows, although I see the filename as desired in the console as I make my test call, the system is only using the default file name to save the mixmonitor file (agented + uniqueID) Agents.conf [general] persistentagents=yes [agents] maxlogintries=3 musiconhold => default updatecdr=yes recordagentcalls=yes recordformat=wav49 urlprefix=http://pbx.netoneint.com/calls/ savecallsin=/var/calls agent => 1000,1000,Ed Test1 agent => 1001,1001,Ed Test2 queues.conf [noi-noc] monitor-format = wav49 monitor-type = MixMonitor member => Agent/1001 member => Agent/1000 extensions.conf exten => 8484,1,Set(MONITOR_FILENAME=QUEUE-NOC-${CALLERID(NUM)}-${STRFTIME(${EPOCH) exten => 8484,1,answer exten => 8484,2,Queue(noi-noc) Console output -- Executing [EMAIL PROTECTED]:1] Set("Zap/1-1", "MONITOR_FILENAME=QUEUE-NOC-4073844200-Fri Jun 6 15:06:38 2008") in new stack -- Executing [EMAIL PROTECTED]:2] Queue("Zap/1-1", "noi-noc") in new stack -- Started music on hold, class 'default', on Zap/1-1 -- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1001 -- Executing [EMAIL PROTECTED]:1] Dial("Local/[EMAIL PROTECTED],2", "SIP/1658") in new stack -- Called 1658 -- SIP/1658-087e7610 is ringing -- Agent/1001 is ringing -- SIP/1658-087e7610 answered Local/[EMAIL PROTECTED],2 -- Agent/1001 answered Zap/1-1 -- Stopped music on hold on Zap/1-1 [Jun 6 15:06:40] WARNING[3976]: app_queue.c:3014 try_calling: The device state of this queue member, Agent/1001, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. == Begin MixMonitor Recording Zap/1-1 == Spawn extension (numberplan-custom-3, 1658, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' == Spawn extension (incoming-att, 8484, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' == End MixMonitor Recording Zap/1-1 ___ -- 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] MiixMonitor filename for queue calls.
Can anyone give me input on the following issue? I have a queue with MixMonitor enabled. This is also enabled in agents.conf. On my extensions.conf, I am setting the monitor filename as fillows, although I see the filename as desired in the console as I make my test call, the system is only using the default file name to save the mixmonitor file (agented + uniqueID) Agents.conf [general] persistentagents=yes [agents] maxlogintries=3 musiconhold => default updatecdr=yes recordagentcalls=yes recordformat=wav49 urlprefix=http://pbx.netoneint.com/calls/ savecallsin=/var/calls agent => 1000,1000,Ed Test1 agent => 1001,1001,Ed Test2 queues.conf [noi-noc] monitor-format = wav49 monitor-type = MixMonitor member => Agent/1001 member => Agent/1000 extensions.conf exten => 8484,1,Set(MONITOR_FILENAME=QUEUE-NOC-${CALLERID(NUM)}-${STRFTIME(${EPOCH) exten => 8484,1,answer exten => 8484,2,Queue(noi-noc) Console output -- Executing [EMAIL PROTECTED]:1] Set("Zap/1-1", "MONITOR_FILENAME=QUEUE-NOC-4073844200-Fri Jun 6 15:06:38 2008") in new stack -- Executing [EMAIL PROTECTED]:2] Queue("Zap/1-1", "noi-noc") in new stack -- Started music on hold, class 'default', on Zap/1-1 -- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1001 -- Executing [EMAIL PROTECTED]:1] Dial("Local/[EMAIL PROTECTED],2", "SIP/1658") in new stack -- Called 1658 -- SIP/1658-087e7610 is ringing -- Agent/1001 is ringing -- SIP/1658-087e7610 answered Local/[EMAIL PROTECTED],2 -- Agent/1001 answered Zap/1-1 -- Stopped music on hold on Zap/1-1 [Jun 6 15:06:40] WARNING[3976]: app_queue.c:3014 try_calling: The device state of this queue member, Agent/1001, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. == Begin MixMonitor Recording Zap/1-1 == Spawn extension (numberplan-custom-3, 1658, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' == Spawn extension (incoming-att, 8484, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' == End MixMonitor Recording Zap/1-1 ___ -- 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] sip show peers
Anyone has any good ideas on how to parse the CDR events and QUEUEs log events from AMI connection? Thank you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Friday, May 02, 2008 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip show peers On Friday 02 May 2008 12:35:48 Philipp Kempgen wrote: > Yeah, sometimes it would be helpful if asterisk -rx 'command' didn't > pretty print but instead fall back to an easily parseable output > format (like TSV with cslashes) if stdout isn't connected to a tty > (isatty()). The CLI is intended to be used by a human. If you want machine parseable output, I would suggest using AMI, as that's what it's meant for. -- Tilghman ___ -- 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] problems with zaptel and Udev
I had the same issue and updated my Zaptel drivers to version 1.4.17 and it's rebooting fine now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of robert boardman Sent: Sunday, January 13, 2008 12:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] problems with zaptel and Udev Hi I have had a Centos 5 installed with asterisk and zaptel for a couple of weeks, I had to reboot eh machine today, and when it rebooted it got stuck at "Starting udev" if I remove thew tdm400 it boots OK, but no zaptel has anyone seen this , and can offer any advice? Thanks Robb ___ -- 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] Asterisk ports and CentOS firewall
If I enable the firewall on my Server, which ports should I open for Asterisk to work properly. Is it enough to just open the SIP ports? ___ -- 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] install Asterisk-addons 1.4.2
I have Asterisk 1.4.5 and addons 1.4.1. Can anyone tell me if I can just install addons 1.4.2 on this system without re installing Asterisk? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of clive.chan(Alpha Trilogies Networks) Sent: Wednesday, June 20, 2007 9:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] install Asterisk-addons 1.4.2 Hi, I am trying to install the Asterisk-addons-1.4.2, and when I make install it prompt me such error messages make[1]: Entering directory `/usr/src/asterisk-addons/asterisk-ooh323c' cp .libs/libchan_h323.so.1.0.1 /usr/lib/asterisk/modules/chan_ooh323.so cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory make[1]: *** [install] Error 1 make[1]: Leaving directory `/usr/src/asterisk-addons/asterisk-ooh323c' make: *** [install] Error 2 How to solve it out? clive chan Alpha Trilogies Networks Sdn Bhd Tel : 04 - 647 288 Ext: 338 Tel : 04 - 647 2999 Mobile : 012 - 408 6376 email : [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
[asterisk-users] ChanSpy SIP
Has anyone succesfully tried using ChanSpy on SIP channels with the latest Asterisk 1.4? I tried ChanSpy(SIP/5060) to monitor SIP extension 5060 and the console displays, Monitoring Sip/5060, but I don't hear anything. I am able to monitor Zap channels.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] g729
Yes, that is correct. I am using mixmon and using wav49. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder Singh Sent: Wednesday, June 06, 2007 4:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 Are you trying to record the conversation as well ? On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote: > > > > > I installed a hardware g729 codec card in my asterisk, and I'm getting the > following error when calling from a g729 sip extension to a SIP trunk also > set to g729. The call goes through just fine, but these error messages keep > flying by until I disconnect the call. > > > > Any ideas? > > > > ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin > failed, dropping frame for spies > > Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: > Translation to slin failed, dropping frame for spies > > Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies: > Translation to slin failed, dropping frame for spies > > Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: > Translation to slin failed, dropping frame for spies > > Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies: > Translation to slin failed, dropping frame for spies > > Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies: > Translation to slin failed, dropping frame for spies > > Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies: > Translation to slin failed, dropping frame for spies > > Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: > Translation to slin failed, dropping frame for spies > > Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies: > Translation to slin failed, dropping frame for spies > > Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: > Translation to slin failed, dropping frame for spies > > Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies: > Translation to slin failed, dropping frame for spies > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users