[asterisk-users] FAX 2 mail configuration
Hello Im trying to configure Fax2Mail in my Asterisk 1.4.23.1 server, wich receievs the Faxes through a SIP trunk. I found a lot of solutions in voip-info.org So, I would like to know what's the best free Fax2Mail solution and if I really need to install Dahdi or Zaptel. Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt -- _ -- 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] ZTdummy
Hello I have a 4 span PRI board with Zaptel, and Im using it for a long time. In the last days I noticed that the result of zap show status show a ZTDUMMY but I never installed it: o*CLI zap show status Description Alarms IRQ bpviol CRC4 T4XXP (PCI) Card 0 Span 1OK 0 0 0 T4XXP (PCI) Card 0 Span 2OK 0 0 0 T4XXP (PCI) Card 0 Span 3UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 4UNCONFIGUR 0 0 0 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 What is that doing ? Also, some people say they cant send me calls through the ZAP trunk... could be because of this ZTDUMMY? Thanks Regards Joao Pereira -- _ -- 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] spandsp with asterisk 1.4.x
Em 17-03-2010 20:51, VinÃcius Fontes escreveu: - Joao Gomes Pereiragomespere...@startel.pt escreveu: Hello Im trying to receive FAXes with my Asterisk with rxfax command. To do that, Im trying to load the app_fax.so module but asterisk says: [Mar 17 20:06:04] WARNING[11907]: loader.c:359 load_dynamic_module: Error loading module 'app_fax.so': libspandsp.so.2: cannot open shared object file: No such file or directory [Mar 17 20:06:04] WARNING[11907]: loader.c:653 load_resource: Module 'app_fax.so' could not be loaded. But I do have libspandsp.so.2 # find / -name libspandsp.so.2 /usr/local/lib/libspandsp.so.2 And yes, /usr/local/lib is in my ld.so.conf: cat /etc/ld.so.conf include ld.so.conf.d/*.conf /etc/ld.so.conf.d/*.conf /usr/local/lib /usr/include /usr/local/include What could be missing? Thanks Regards Joao Pereira Sorry for kinda hijacking your topic, but where did you get the 1.4 app_fax.so backport from? I'm really interested on that. Yes, It was difficult to find... I dont have the page, but here is the wget: wget http://downloads.sourceforge.net/project/agx-ast-addons/precompiled-linux-spandsp-app-fax.tar.bz2 Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt -- _ -- 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] spandsp with asterisk 1.4.x
Em 17-03-2010 20:28, Doug Lytle escreveu:
Joao Gomes Pereira wrote:
What could be missing?
Running ldconfig as root
Thanks, thats it!!!
Now the module is loaded.
I just hope the FAX code works:
[macro-faxreceive]
exten = s,1,Set(FAXFILE=/var/spool/asterisk/fax/${CALLEDFAX}/${UNIQUEID})
exten = s,2,NoOP()
exten = s,3,NoOP()
exten = s,4,rxfax(${FAXFILE}.tif)
exten = s,103,Set(extmail...@startel.pt)
exten = s,104,Goto(4)
exten = s,105,Set(EXTNAME=Unknown)
exten = s,106,Goto(4)
exten = s,107,Set(EXTCOMPANY=Company)
exten = s,108,Goto(4)
Thanks again
Regards
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt
--
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[asterisk-users] spandsp with asterisk 1.4.x
Hello Im trying to receive FAXes with my Asterisk with rxfax command. To do that, Im trying to load the app_fax.so module but asterisk says: [Mar 17 20:06:04] WARNING[11907]: loader.c:359 load_dynamic_module: Error loading module 'app_fax.so': libspandsp.so.2: cannot open shared object file: No such file or directory [Mar 17 20:06:04] WARNING[11907]: loader.c:653 load_resource: Module 'app_fax.so' could not be loaded. But I do have libspandsp.so.2 # find / -name libspandsp.so.2 /usr/local/lib/libspandsp.so.2 And yes, /usr/local/lib is in my ld.so.conf: cat /etc/ld.so.conf include ld.so.conf.d/*.conf /etc/ld.so.conf.d/*.conf /usr/local/lib /usr/include /usr/local/include What could be missing? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt -- _ -- 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] install asterisk from binaries
Hello I'm trying to install Asterisk in a Linux server without compiler, yum or apt-get. Is this possible? Where can I find the pre compiled binaries? Thanks Regards Joao Pereira -- _ -- 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] asterisk peer uses 5060 to send and 5061 to receive
Hello Im configuring an asterisk peer, wich uses port 5060 to send and port 5061 to receive signaling. So, wich port should I put in my asterisk SIP trunk configuration? port = 5060 or port = 5061 ? Thanks Regards Joao Pereira -- _ -- 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] asterisk peer uses 5060 to send and 5061 to receive
Hello Im configuring an asterisk peer, wich uses port 5060 to send and port 5061 to receive signaling. So, wich port should I put in my asterisk SIP trunk configuration? port = 5060 or port = 5061 ? Thanks Regards Joao Pereira -- _ -- 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] ChanSpy doesn't hangs up
Hello I have a simple configuration to allow the admins to listen the agents calls: exten = _654,1,ChanSpy(Agent) exten = _654,2,Hangup() The problem is... even when the agents hung up... it seems the channels remain active: asterisk*CLI show channels SIP/211-b3042018 6...@default:1Up ChanSpy(Agent) SIP/211-b3fbf768 6...@default:1Up ChanSpy(Agent) SIP/211-b4d88940 6...@default:1Up ChanSpy(Agent) SIP/211-b52e6498 6...@default:1Up ChanSpy(Agent) SIP/211-b4bbfcc0 6...@default:1Up ChanSpy(Agent) SIP/211-b4ba9f88 6...@default:1Up ChanSpy(Agent) (the agent is using extension 211) I have more then 10 lines like these. Why do the ChanSpy calls dont hang up? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt -- _ -- 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 gets stuck
Hello I have a simple configuration to allow the admins listen to agent calls: exten = _654,1,ChanSpy(Agent) exten = _654,2,Hangup() The problem is... even when the agents hung up... it seems the channels remain active: okavango*CLI show channels SIP/211-b3042018 6...@default:1Up ChanSpy(Agent) SIP/211-b3fbf768 6...@default:1Up ChanSpy(Agent) SIP/211-b4d88940 6...@default:1Up ChanSpy(Agent) SIP/211-b52e6498 6...@default:1Up ChanSpy(Agent) SIP/211-b4bbfcc0 6...@default:1Up ChanSpy(Agent) SIP/211-b4ba9f88 6...@default:1Up ChanSpy(Agent) (the agent is using extension 211) I have more then 10 lines like these. Why do the ChanSpy calls dont hang up? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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 registers with private IP
Hello I'm trying to register an Asterisk working behind Nat. Here is the trunk: register=username:passw...@sip.startel.pt [startel] type=peer host=sip.startel.pt username=username fromuser=username secret=password qualify=yes disallow=all allow=ulaw allow=alaw allow=gsm insecure=very port=5060 nat=yes canreinvite=yes The problem is: Asterisk is registering with its internal IP (192.168.1.25), as you can see here: sip:s...@192.168.1.25 Q= Expires:: 81 Callid:: 480b40aa13ddd8707787b21a69656...@127.0.0.1 Cseq:: 103 User-agent:: Asterisk PBX How can I force Asterisk to register with its public IP? Is it possible to configure STUN in an Asterisk trunk? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk registers with private IP
Nice!!! Thanks a lot. Its not the case (because Im using a fixed IP), but if the IP where dynamic? Thanks Regards Joao Pereira Randy R wrote: externip=123.123.123.123 On Tue, Dec 1, 2009 at 4:32 PM, Joao Gomes Pereira gomespere...@startel.pt wrote: Hello I'm trying to register an Asterisk working behind Nat. Here is the trunk: register=username:passw...@sip.startel.pt [startel] type=peer host=sip.startel.pt username=username fromuser=username secret=password qualify=yes disallow=all allow=ulaw allow=alaw allow=gsm insecure=very port=5060 nat=yes canreinvite=yes The problem is: Asterisk is registering with its internal IP (192.168.1.25), as you can see here: sip:s...@192.168.1.25 Q= Expires:: 81 Callid:: 480b40aa13ddd8707787b21a69656...@127.0.0.1 Cseq:: 103 User-agent:: Asterisk PBX How can I force Asterisk to register with its public IP? Is it possible to configure STUN in an Asterisk trunk? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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 -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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 in Asterisk 1.2.24
Hello I have an old Asterisk where I need to listen to Agent calls. So I created this code: exten = _555,1,ChanSpy(Agent) exten = _555,n,Hangup() But I always get: 2009-10-22 16:00:38 WARNING[5695]: pbx.c:1720 pbx_extension_helper: No application 'ChanSpy' for extension (default, 555, 1) It seems that Asterisk doesn't have ChanSpy enabled... is this possible? Which Asterisk module do I have to enable? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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] ChanSpy in Asterisk 1.2.24
Thanks a lot The file App_chanspy was already in /usr/lib/asterisk/modules But I had in my modules.conf: noload = app_chanspy.conf Now I erased this line... but Asterisk still doesn't load this app_chanspy... Do I need to stop/start Asterisk? Or the reload is enough? Thanks Regards Joao Pereira Danny Nicholas wrote: App_chanspy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joao Gomes Pereira Sent: Thursday, October 22, 2009 10:08 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ChanSpy in Asterisk 1.2.24 Hello I have an old Asterisk where I need to listen to Agent calls. So I created this code: exten = _555,1,ChanSpy(Agent) exten = _555,n,Hangup() But I always get: 2009-10-22 16:00:38 WARNING[5695]: pbx.c:1720 pbx_extension_helper: No application 'ChanSpy' for extension (default, 555, 1) It seems that Asterisk doesn't have ChanSpy enabled... is this possible? Which Asterisk module do I have to enable? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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] ChanSpy in Asterisk 1.2.24
I had to restart Asterisk, and now the module is loaded. Thanks a lot for the help Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt Danny Nicholas wrote: Try module load app_chanspy.so from CLI. If that doesn't work, restart asterisk. -Original Message- From: Joao Gomes Pereira [mailto:gomespere...@startel.pt] Sent: Thursday, October 22, 2009 10:57 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ChanSpy in Asterisk 1.2.24 Thanks a lot The file App_chanspy was already in /usr/lib/asterisk/modules But I had in my modules.conf: noload = app_chanspy.conf Now I erased this line... but Asterisk still doesn't load this app_chanspy... Do I need to stop/start Asterisk? Or the reload is enough? Thanks Regards Joao Pereira Danny Nicholas wrote: App_chanspy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joao Gomes Pereira Sent: Thursday, October 22, 2009 10:08 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ChanSpy in Asterisk 1.2.24 Hello I have an old Asterisk where I need to listen to Agent calls. So I created this code: exten = _555,1,ChanSpy(Agent) exten = _555,n,Hangup() But I always get: 2009-10-22 16:00:38 WARNING[5695]: pbx.c:1720 pbx_extension_helper: No application 'ChanSpy' for extension (default, 555, 1) It seems that Asterisk doesn't have ChanSpy enabled... is this possible? Which Asterisk module do I have to enable? Thanks Regards Joao Pereira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi configuraion / error
channel (Default) (Echo Canceler: mg2) (Slaves: 124) 124 channels to configure. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) [r...@catumbela modules]# Any idea of what could be nmissing? Thanks a lot Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Dahdi configuraion / error
it looks like this: tail /etc/dahdi/modules # Autogenerated by /usr/sbin/dahdi_genconf (Dahdi::Config::Gen::Modules) on Wed Jun 24 12:41:26 2009 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. wct4xxp Danny Nicholas escreveu: No such device is sometimes an indication that /etc/init.d/dahdi start did not load the driver. What does /etc/dahdi/modules look like? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joao Gomes Pereira Sent: Thursday, September 03, 2009 11:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi configuraion / error Tzafrir Cohen escreveu: On Tue, Sep 01, 2009 at 08:53:04AM -0500, Danny Nicholas wrote: This may be dumb and/or obvious, but did you do these steps? 1. dahdi_genconf dahdi modules user to make sure all of the configuration files are up to standard this is an R23 connection, so I dont think genconfig will help. Also, I already had this working but not its not working... I dont know why (Which will default to generate a ccs configuration for it, rather than cas) Current configuration appears to be OK at first glance. 2. dahdi_cfg -vv to see if any obvious messages came up (causing the red condition(s) ) The information you have provided is useful, but it boils down to this (IMO) - RED is dead! Is there actually a cable plugged? Connecting it to a live system? Yes, it has a cable connected (in port 1) to a Telco, so Im configuerd as slave. Here is the dahdi_cfg -vv ( sorry for the long post ) [r...@catumbela modules]# dahdi_cfg -vv DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 08) Channel 09: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 11) Channel 12: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 12) Channel 13: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 13) Channel 14: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 14) Channel 15: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 15) Channel 16: D-channel (Default) (Echo Canceler: none) (Slaves: 16) Channel 17: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 17) Channel 18: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 18) Channel 19: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 19) Channel 20: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 20) Channel 21: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 21) Channel 22: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 22) Channel 23: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 23) Channel 24: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 24) Channel 25: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 25) Channel 26: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 26) Channel 27: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 27) Channel 28: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 28) Channel 29: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 29) Channel 30: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 30) Channel 31: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 31) Channel 32: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 32) Channel 33: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 33) Channel 34: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 34) Channel 35: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 35) Channel 36: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 36) Channel 37: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 37) Channel 38: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 38) Channel 39: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 39) Channel 40: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 40) Channel 41: Clear channel (Default) (Echo
Re: [asterisk-users] Dahdi configuraion / error
here are my logs when I start the dahdi driver: /etc/rc.d/init.d/dahdi start Sep 3 18:02:39 catumbela kernel: Found TE4XXP at base address fdcff000, remapped to f88a8000 Sep 3 18:02:39 catumbela kernel: TE4XXP version c01a0164, burst OFF Sep 3 18:02:39 catumbela kernel: FALC version: 0005, Board ID: 00 Sep 3 18:02:39 catumbela kernel: Reg 0: 0x2afea400 Sep 3 18:02:39 catumbela kernel: Reg 1: 0x2afea000 Sep 3 18:02:39 catumbela kernel: Reg 2: 0x Sep 3 18:02:39 catumbela kernel: Reg 3: 0x Sep 3 18:02:39 catumbela kernel: Reg 4: 0x Sep 3 18:02:39 catumbela kernel: Reg 5: 0x Sep 3 18:02:39 catumbela kernel: Reg 6: 0xc01a0164 Sep 3 18:02:39 catumbela kernel: Reg 7: 0x1f00 Sep 3 18:02:39 catumbela kernel: Reg 8: 0x010200ff Sep 3 18:02:39 catumbela kernel: Reg 9: 0x00fd Sep 3 18:02:39 catumbela kernel: Reg 10: 0x004a Sep 3 18:02:39 catumbela kernel: Found a Wildcard: Wildcard TE405P (2nd Gen) Sep 3 18:02:39 catumbela kernel: TE4XXP: Launching card: 0 Sep 3 18:02:39 catumbela kernel: TE4XXP: Setting up global serial parameters Sep 3 18:02:39 catumbela dahdi: wct4xxp: succeeded Sep 3 18:02:39 catumbela kernel: About to enter spanconfig! Sep 3 18:02:39 catumbela kernel: Done with spanconfig! Sep 3 18:02:39 catumbela kernel: About to enter startup! Sep 3 18:02:39 catumbela kernel: TE4XXP: Span 1 configured for CAS/HDB3 Sep 3 18:02:39 catumbela kernel: timing source auto card 0! Sep 3 18:02:39 catumbela kernel: wct4xxp: Setting yellow alarm on span 1 Sep 3 18:02:39 catumbela kernel: timing source auto card 0! Sep 3 18:02:39 catumbela kernel: SPAN 1: Primary Sync Source Sep 3 18:02:39 catumbela kernel: VPM400: Not Present Sep 3 18:02:39 catumbela kernel: VPM450: Not Present Sep 3 18:02:39 catumbela kernel: Completed startup! Sep 3 18:02:39 catumbela dahdi: Running dahdi_cfg: succeeded Joao Gomes Pereira escreveu: it looks like this: tail /etc/dahdi/modules # Autogenerated by /usr/sbin/dahdi_genconf (Dahdi::Config::Gen::Modules) on Wed Jun 24 12:41:26 2009 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. wct4xxp Danny Nicholas escreveu: No such device is sometimes an indication that /etc/init.d/dahdi start did not load the driver. What does /etc/dahdi/modules look like? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joao Gomes Pereira Sent: Thursday, September 03, 2009 11:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi configuraion / error Tzafrir Cohen escreveu: On Tue, Sep 01, 2009 at 08:53:04AM -0500, Danny Nicholas wrote: This may be dumb and/or obvious, but did you do these steps? 1. dahdi_genconf dahdi modules user to make sure all of the configuration files are up to standard this is an R23 connection, so I dont think genconfig will help. Also, I already had this working but not its not working... I dont know why (Which will default to generate a ccs configuration for it, rather than cas) Current configuration appears to be OK at first glance. 2. dahdi_cfg -vv to see if any obvious messages came up (causing the red condition(s) ) The information you have provided is useful, but it boils down to this (IMO) - RED is dead! Is there actually a cable plugged? Connecting it to a live system? Yes, it has a cable connected (in port 1) to a Telco, so Im configuerd as slave. Here is the dahdi_cfg -vv ( sorry for the long post ) [r...@catumbela modules]# dahdi_cfg -vv DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 08) Channel 09: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 11) Channel 12: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 12) Channel 13: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 13) Channel 14: Clear channel (Default) (Echo
Re: [asterisk-users] Dahdi configuraion / error
Here it is: [r...@catumbela ~]# lsmod|grep wct4xxp wct4xxp 242176 0 dahdi 197640 5 wct4xxp [r...@catumbela ~]# dmesg is in attach :) Danny Nicholas escreveu: Okay. What is the output of these commands? dmesg lsmod|grep wct4xxp -Original Message- From: Joao Gomes Pereira [mailto:gomespere...@startel.pt] Sent: Thursday, September 03, 2009 11:56 AM To: Danny Nicholas; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi configuraion / error it looks like this: tail /etc/dahdi/modules # Autogenerated by /usr/sbin/dahdi_genconf (Dahdi::Config::Gen::Modules) on Wed Jun 24 12:41:26 2009 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. wct4xxp Danny Nicholas escreveu: No such device is sometimes an indication that /etc/init.d/dahdi start did not load the driver. What does /etc/dahdi/modules look like? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joao Gomes Pereira Sent: Thursday, September 03, 2009 11:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi configuraion / error Tzafrir Cohen escreveu: On Tue, Sep 01, 2009 at 08:53:04AM -0500, Danny Nicholas wrote: This may be dumb and/or obvious, but did you do these steps? 1. dahdi_genconf dahdi modules user to make sure all of the configuration files are up to standard this is an R23 connection, so I dont think genconfig will help. Also, I already had this working but not its not working... I dont know why (Which will default to generate a ccs configuration for it, rather than cas) Current configuration appears to be OK at first glance. 2. dahdi_cfg -vv to see if any obvious messages came up (causing the red condition(s) ) The information you have provided is useful, but it boils down to this (IMO) - RED is dead! Is there actually a cable plugged? Connecting it to a live system? Yes, it has a cable connected (in port 1) to a Telco, so Im configuerd as slave. Here is the dahdi_cfg -vv ( sorry for the long post ) [r...@catumbela modules]# dahdi_cfg -vv DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 08) Channel 09: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 11) Channel 12: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 12) Channel 13: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 13) Channel 14: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 14) Channel 15: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 15) Channel 16: D-channel (Default) (Echo Canceler: none) (Slaves: 16) Channel 17: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 17) Channel 18: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 18) Channel 19: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 19) Channel 20: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 20) Channel 21: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 21) Channel 22: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 22) Channel 23: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 23) Channel 24: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 24) Channel 25: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 25) Channel 26: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 26) Channel 27: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 27) Channel 28: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 28) Channel 29: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 29) Channel 30: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 30) Channel 31: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 31) Channel 32: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 32) Channel 33: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 33) Channel 34: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 34) Channel 35
[asterisk-users] Dahdi configuraion / error
timing source auto card 0! VPM400: Not Present VPM450: Not Present Completed startup! lp: driver loaded but no devices found NET: Registered protocol family 10 Disabled Privacy Extensions on device c034f620(lo) IPv6 over IPv4 tunneling driver divert: not allocating divert_blk for non-ethernet device sit0 eth0: no IPv6 routers present Stopped TE4XXP, Turned off DMA TE4XXP: Disabling interrupts since there are no active spans dahdi: Telephony Interface Unloaded dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.2.0.2 Found TE4XXP at base address fdcff000, remapped to f88a8000 TE4XXP version c01a0164, burst OFF FALC version: 0005, Board ID: 00 Reg 0: 0x37248400 Reg 1: 0x37248000 Reg 2: 0x Reg 3: 0x Reg 4: 0x Reg 5: 0x Reg 6: 0xc01a0164 Reg 7: 0x1f00 Reg 8: 0x010200ff Reg 9: 0x00fd Reg 10: 0x004a Found a Wildcard: Wildcard TE405P (2nd Gen) TE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters About to enter spanconfig! Done with spanconfig! dahdi: Registered tone zone 0 (United States / North America) About to enter startup! TE4XXP: Span 1 configured for CAS/HDB3 wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 1: Primary Sync Source VPM400: Not Present VPM450: Not Present Completed startup! Stopped TE4XXP, Turned off DMA TE4XXP: Disabling interrupts since there are no active spans dahdi: Telephony Interface Unloaded dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.2.0.2 Found TE4XXP at base address fdcff000, remapped to f88a8000 TE4XXP version c01a0164, burst OFF FALC version: 0005, Board ID: 00 Reg 0: 0x36d84400 Reg 1: 0x36d84000 Reg 2: 0x Reg 3: 0x Reg 4: 0x Reg 5: 0x Reg 6: 0xc01a0164 Reg 7: 0x1f00 Reg 8: 0x010200ff Reg 9: 0x00fd Reg 10: 0x004a Found a Wildcard: Wildcard TE405P (2nd Gen) TE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters About to enter spanconfig! Done with spanconfig! dahdi: Registered tone zone 0 (United States / North America) About to enter startup! TE4XXP: Span 1 configured for CAS/HDB3 timing source auto card 0! wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 1: Primary Sync Source VPM400: Not Present VPM450: Not Present Completed startup! Stopped TE4XXP, Turned off DMA TE4XXP: Disabling interrupts since there are no active spans dahdi: Telephony Interface Unloaded dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.2.0.2 Found TE4XXP at base address fdcff000, remapped to f88a8000 TE4XXP version c01a0164, burst OFF FALC version: 0005, Board ID: 00 Reg 0: 0x35dbc400 Reg 1: 0x35dbc000 Reg 2: 0x Reg 3: 0x Reg 4: 0x Reg 5: 0x Reg 6: 0xc01a0164 Reg 7: 0x1f00 Reg 8: 0x010200ff Reg 9: 0x00fd Reg 10: 0x004a Found a Wildcard: Wildcard TE405P (2nd Gen) TE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters About to enter spanconfig! Done with spanconfig! dahdi: Registered tone zone 0 (United States / North America) About to enter startup! TE4XXP: Span 1 configured for CAS/HDB3 wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 1: Primary Sync Source VPM400: Not Present VPM450: Not Present Completed startup! DMESG Could this be a configuration issue or a hardware problem? Thanks regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] TE4XXP: Version Synchronization Error!
Hello to all I'm using asterisk 1.4 and dahdi. I had everything working fine, and I could place calls through my R2 channel. But now the channel is always RED and Im getting this error message: TE4XXP: Version Synchronization Error! Here is my chan_dahdi.conf-- [channels] language=en context=incomingr2 signalling=mfcr2 mfcr2_variant=ar switchtype=national mfcr2_get_ani_first=no mfcr2_max_ani=20 mfcr2_max_dnis=9 mfcr2_category=national_subscriber channel =1-15,17-30 here is my dahdi config: span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-30:1101 dchan=16 --- What could be the problem? Why was this working fine and now the channel is RED? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] TE4XXP: Version Synchronization Error!
Im also getting these errors: Aug 26 17:58:17 catumbela kernel: dahdi: Registered tone zone 0 (United States / North America) Aug 26 17:58:17 catumbela kernel: About to enter startup! Aug 26 17:58:17 catumbela kernel: TE4XXP: Version Synchronization Error! Aug 26 17:58:17 catumbela last message repeated 127 times Aug 26 17:58:17 catumbela kernel: TE4XXP: Span 1 configured for CAS/HDB3 Aug 26 17:58:17 catumbela kernel: TE4XXP: Version Synchronization Error! Aug 26 17:58:17 catumbela last message repeated 11 times Aug 26 17:58:17 catumbela dahdi: Running dahdi_cfg: succeeded Aug 26 17:58:18 catumbela kernel: TE4XXP: Version Synchronization Error! Aug 26 17:58:18 catumbela last message repeated 21 times Aug 26 17:58:18 catumbela kernel: wct4xxp: Setting yellow alarm on span 1 Aug 26 17:58:16 catumbela kernel: TE4XXP: Version Synchronization Error! Aug 26 17:58:16 catumbela last message repeated 67 times Aug 26 17:58:16 catumbela kernel: SPAN 1: Primary Sync Source Aug 26 17:58:16 catumbela kernel: TE4XXP: Version Synchronization Error! Aug 26 17:58:16 catumbela last message repeated 2 times Aug 26 17:58:16 catumbela kernel: VPM400: Not Present Aug 26 17:58:16 catumbela kernel: TE4XXP: Version Synchronization Error! Aug 26 17:58:16 catumbela last message repeated 37 times Aug 26 17:58:16 catumbela kernel: VPM450: Not Present Aug 26 17:58:16 catumbela kernel: TE4XXP: Version Synchronization Error! Aug 26 17:58:16 catumbela kernel: Completed startup! Aug 26 17:58:16 catumbela kernel: TE4XXP: Version Synchronization Error! What could be worng with my dahdi configuration? Thanks regards Joao Pereira Joao Gomes Pereira escreveu: Hello to all I'm using asterisk 1.4 and dahdi. I had everything working fine, and I could place calls through my R2 channel. But now the channel is always RED and Im getting this error message: TE4XXP: Version Synchronization Error! Here is my chan_dahdi.conf-- [channels] language=en context=incomingr2 signalling=mfcr2 mfcr2_variant=ar switchtype=national mfcr2_get_ani_first=no mfcr2_max_ani=20 mfcr2_max_dnis=9 mfcr2_category=national_subscriber channel =1-15,17-30 here is my dahdi config: span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-30:1101 dchan=16 --- What could be the problem? Why was this working fine and now the channel is RED? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] queue agents get stuck
Hello to all I have a queue where often my agents get stuck and cannot logoff. This is very bad, because agents cannot login again, and in Queuemetrics reports the agents appear to be online. How can I create a timeout to my agents and for the queue to kick them? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] AGI with queues status
Hello I'm trying to use an AGI that returns the queues status (numbers of available agents, etc ), but I'm having some problems with it (it's still very buggy). Is there any AGI repository with source code samples? Had anyone used an AGI to check queues and agents status? Thanks regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Asterisk and Kamailio NAT problem
Hello I'm using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=ulaw allow=alaw allow=gsm allow=g726 qualify=1000 username=my_username fromuser=my_username secret=password sip*CLI sip show peers Name/username HostDyn Nat ACL Port Status kamailio/my_username xxx.xxx.xxx.xxx 5060 OK (890 ms) Is there something missing in my SIP.CONF to improve the compatibility with Kamailio? How can I debug the RTP stream in Asterisk? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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 and Kamailio NAT problem
Hello Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=ulaw allow=alaw allow=gsm allow=g726 qualify=1000 username=my_username fromuser=my_username secret=password sip*CLI sip show peers Name/username HostDyn Nat ACL Port Status kamailio/my_username xxx.xxx.xxx.xxx 5060 OK (890 ms) Is there something missing in my SIP.CONF to improve the compatibility with Kamailio? How can I debug the RTP stream in Asterisk? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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] dinamic queue distribution
Hello I have 2 queues and I would like to send calls to queue_1 and queue_2 dynamically. For example: If I have 10 agents logged (2 in queue_1 and 8 in queue_2) I want 20% of the calls to be sent to queue_1 and 80% to queue_2 Is this possible? Is there a way I can see how many logged (or available) agents I have in a queue before sending a call? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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] x-lite settings to reach asterisk
Does your asterisk has a private or public IP? Is your X-Lite in your LAN or outside? If X-Lite is outside the Lan, you need to forward all traffic coming to your Lan in port 5060, to asterisks private IP. Activate SIP debug in asterisk CLI to check if the traffic is getting to asterisk. Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt Tom Poe wrote: Hello: I have the linux version 2.0 of x-lite downloaded. Does anyone know exactly what settings needed to reach the asterisk server on my home network? Internet -DSL transparent bridge -router -asterisk -softphone x-lite attempts to login and register, but times out. There must be some setting I'm missing. Any help appreciated. Tom ___ -- 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 -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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] queues load balancing
Hello I have 2 queues (queue_1 and queue_2 ) in my Asterisk, and I want to send 2/3 of the calls to queue_1 and 1/3 of the calls to queue_2 How can I do that load balancing in extensions.conf? I have something like this: exten = 123,1,Ringing exten = 123,2,Wait(1) exten = 123,3,Answer ; 2 in 3 calls go to queue_1 exten = 123,x,Queue(queue_1) ; 1 in 3 calls go to queue_2 exten = 123,x,Queue(queue_2) But how can I configure this call distribution? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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] queues load balancing
Thanks for the idea. I will try it this way: exten = 123,1,Ringing exten = 123,2,Wait(1) exten = 123,3,Answer exten = 123,4,Random(33:123,10) exten = 123,5,Queue(queue_1) exten = 123,6,Hangup exten = 123,10,Queue(queue_2) exten = 123,11,Hangup Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt Geraint Lee wrote: Take a look at: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Random You should be able to do what you want with this, it obviously won't take in to account the actual amount of people still in the queue (for example if someone hangs up while on hold). I'm sure there'd be a way of integrating this in to it using some different functions, but for a quick fix random will do just fine. Cheers 2009/7/20 Joao Gomes Pereira gomespere...@startel.pt mailto:gomespere...@startel.pt Hello I have 2 queues (queue_1 and queue_2 ) in my Asterisk, and I want to send 2/3 of the calls to queue_1 and 1/3 of the calls to queue_2 How can I do that load balancing in extensions.conf? I have something like this: exten = 123,1,Ringing exten = 123,2,Wait(1) exten = 123,3,Answer ; 2 in 3 calls go to queue_1 exten = 123,x,Queue(queue_1) ; 1 in 3 calls go to queue_2 exten = 123,x,Queue(queue_2) But how can I configure this call distribution? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt http://www.startel.pt +351 304500650 sip: gomespere...@startel.pt mailto:gomespere...@startel.pt ___ -- 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 -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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] Call recording in - out
Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf- [general] persistentmembers = no monitor-format=wav monitor-join=yes monitor-type=mixmonitor [queue_1] persistentmembers = no monitor-format=wav monitor-join=yes monitor-type=mixmonitor wrapuptime=3 timeout=15 strategy=roundrobin retry=5 member = Agent/600 member = Agent/601 agents.conf- [general] persistentagents=no [agents] updatecdr=no recordagentcalls=yes recordformat=wav monitor-join=yes savecallsin=/var/www/html/recordings/ custom_beep=beep group=1 wrapuptime=19 ackcall=no group=1 agent = 600,1234,Jose agent = 601,1234,Maria Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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 recording in - out
Hello I did as you told me, but the problem remains. Im using Asterisk 1.2.x and this is my config: queues.conf - [general] persistentmembers = no [queue_1] persistentmembers = no monitor-format=wav monitor-join=yes monitor-type=MixMonitor wrapuptime=3 timeout=15 strategy=roundrobin retry=5 queue-youarenext= queue-thereare= queue-thankyou= queue-callswaiting= member = Agent/600 member = Agent/601 agents.conf - [general] persistentagents=no [agents] updatecdr=no custom_beep=beep group=1 wrapuptime=19 ackcall=no musiconhold = music group=1 agent = 600,1234,Jose agent = 601,1234,Maria The calls are recordedbut always produces 2 separated files, with in and out. What could be missing? Do I need to create a line in crontab to mix the 2 files? Thanks regards Joao Pereira Kurian Thayil wrote: Hi, I had similar issue which happened when record option was mentioned in both agents.conf and queues.conf. When I commented the recordagentcalls option in agents.conf, it started to work. Mention the monitor option only in the queues.conf file. Do try. Regards, Kurian Thayil. On Sun, 2009-06-07 at 17:51 +0100, Joao Gomes Pereira wrote: Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf- [general] persistentmembers = no monitor-format=wav monitor-join=yes monitor-type=mixmonitor [queue_1] persistentmembers = no monitor-format=wav monitor-join=yes monitor-type=mixmonitor wrapuptime=3 timeout=15 strategy=roundrobin retry=5 member = Agent/600 member = Agent/601 agents.conf- [general] persistentagents=no [agents] updatecdr=no recordagentcalls=yes recordformat=wav monitor-join=yes savecallsin=/var/www/html/recordings/ custom_beep=beep group=1 wrapuptime=19 ackcall=no group=1 agent = 600,1234,Jose agent = 601,1234,Maria Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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
