Re: [asterisk-users] PSTN connectivity
Hi, Please see the sample. A ) Analog HardwareType Ports Action FXO Ports 1 Edithttp://192.168.1.134/admin/config.php?type=setupdisplay=dahdidahdi_form=analog_signallingports=fxo FXS Ports -- B) Analog FXO PortsPort 1: *Loop Start* Group: *0* Context: * from-analog* * C ) ZAP Trunk (DAHDI compatibility Mode)* Trunk Description: Outbound Caller ID:CID Options: Maximum Channels: Disable Trunk: Disable Monitor Trunk Failures: Enable Outgoing Dial Rules Dial Rules: 0471+NXX Dial Rules Wizards: Outbound Dial Prefix:Outgoing Settings Zap Identifier (trunk name): *D ) INBOUND route * Description: Extensions: 199 * E ) **OUTBOUND Route* Route Name: 9_outside Route CID: Override Extension CID Route Password: PIN Set: Emergency Dialing: Intra Company Route: Music On Hold? Dial Patterns 8|NXXNXX 8|NXX Dial patterns wizards*: * Trunk SequenceZAP/g0 0 * F ) In command Line I can see the following things * [root@astrisks ~]# *dahdi_cfg -vv* DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == Channel map: Channel 01: FXS Loopstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to none [root@astrisks ~]# *dahdi_scan* [1] active=yes alarms=OK description=Wildcard X100P Board 1 name=WCFXO/0 manufacturer=Digium devicetype=Wildcard X100P location=PCI Bus 02 Slot 02 basechan=1 totchans=1 irq=193 type=analog port=1,FXO *Asterisk CLI* *astrisks*CLI dahdi show status* Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard X100P Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) * output when i dialing to a local number* Connected to Asterisk 1.6.2.11 currently running on astrisks (pid = 2890) Verbosity is at least 3 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [s@from-internal:1] Macro(SIP/199-003a, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-003a' in macro 'hangupcall' == Spawn extension (from-internal, s, 1) exited non-zero on 'SIP/199-003a' -- Executing [h@from-internal:1] Macro(SIP/199-003a, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-003a' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/199-003a' On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind govoi...@gmail.com wrote: Some CLI logs will get you better help on the issue ! also paste the FXO configurations and how you configured it ! On Wed, Sep 28, 2011 at 2:11 PM, michael k mich...@inapp.com wrote: Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before connecting to the FXO card, i am getting a ringing. But i get a message like the number is out of order when i just connect the line to FXO card. Please some one help me to resolve his issue -- _ -- 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 -- _ -- 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
Re: [asterisk-users] PSTN connectivity
The Call at this point is not even looking for FXO/Dahdi/Zap.. See the CLI. there is some misconfiguration in FreePBX and your dialled number is not hitting any dial-able rule. See your FreePBX guide. On Thu, Sep 29, 2011 at 11:01 AM, michael k mich...@inapp.com wrote: Hi, Please see the sample. A ) Analog HardwareType Ports Action FXO Ports 1 Edithttp://192.168.1.134/admin/config.php?type=setupdisplay=dahdidahdi_form=analog_signallingports=fxo FXS Ports -- B) Analog FXO PortsPort 1: *Loop Start* Group: *0* Context: * from-analog* * C ) ZAP Trunk (DAHDI compatibility Mode)* Trunk Description: Outbound Caller ID:CID Options: Maximum Channels: Disable Trunk: Disable Monitor Trunk Failures: Enable Outgoing Dial Rules Dial Rules: 0471+NXX Dial Rules Wizards: Outbound Dial Prefix:Outgoing Settings Zap Identifier (trunk name): *D ) INBOUND route * Description: Extensions: 199 * E ) **OUTBOUND Route* Route Name: 9_outside Route CID: Override Extension CID Route Password: PIN Set: Emergency Dialing: Intra Company Route: Music On Hold? Dial Patterns 8|NXXNXX 8|NXX Dial patterns wizards*: * Trunk SequenceZAP/g0 0 * F ) In command Line I can see the following things * [root@astrisks ~]# *dahdi_cfg -vv* DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == Channel map: Channel 01: FXS Loopstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to none [root@astrisks ~]# *dahdi_scan* [1] active=yes alarms=OK description=Wildcard X100P Board 1 name=WCFXO/0 manufacturer=Digium devicetype=Wildcard X100P location=PCI Bus 02 Slot 02 basechan=1 totchans=1 irq=193 type=analog port=1,FXO *Asterisk CLI* *astrisks*CLI dahdi show status* Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard X100P Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) * output when i dialing to a local number* Connected to Asterisk 1.6.2.11 currently running on astrisks (pid = 2890) Verbosity is at least 3 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [s@from-internal:1] Macro(SIP/199-003a, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-003a' in macro 'hangupcall' == Spawn extension (from-internal, s, 1) exited non-zero on 'SIP/199-003a' -- Executing [h@from-internal:1] Macro(SIP/199-003a, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-003a' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/199-003a' On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind govoi...@gmail.com wrote: Some CLI logs will get you better help on the issue ! also paste the FXO configurations and how you configured it ! On Wed, Sep 28, 2011 at 2:11 PM, michael k mich...@inapp.com wrote: Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before connecting to the FXO card, i am getting a ringing. But i get a message like the number is out of order when i just connect the line to FXO card. Please some one help me to resolve his issue -- _ -- 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
Re: [asterisk-users] PSTN connectivity
Can you please figure out the configuration issue in my freepbx ? On Thu, Sep 29, 2011 at 11:35 AM, Sam Govind govoi...@gmail.com wrote: The Call at this point is not even looking for FXO/Dahdi/Zap.. See the CLI. there is some misconfiguration in FreePBX and your dialled number is not hitting any dial-able rule. See your FreePBX guide. On Thu, Sep 29, 2011 at 11:01 AM, michael k mich...@inapp.com wrote: Hi, Please see the sample. A ) Analog HardwareType Ports Action FXO Ports 1 Edithttp://192.168.1.134/admin/config.php?type=setupdisplay=dahdidahdi_form=analog_signallingports=fxo FXS Ports -- B) Analog FXO PortsPort 1: *Loop Start* Group: *0* Context: * from-analog * * C ) ZAP Trunk (DAHDI compatibility Mode)* Trunk Description: Outbound Caller ID:CID Options: Maximum Channels: Disable Trunk: Disable Monitor Trunk Failures: Enable Outgoing Dial Rules Dial Rules: 0471+NXX Dial Rules Wizards: Outbound Dial Prefix:Outgoing Settings Zap Identifier (trunk name): *D ) INBOUND route * Description: Extensions: 199 * E ) **OUTBOUND Route* Route Name: 9_outside Route CID: Override Extension CID Route Password: PIN Set: Emergency Dialing: Intra Company Route: Music On Hold? Dial Patterns 8|NXXNXX 8|NXX Dial patterns wizards*: * Trunk SequenceZAP/g0 0 * F ) In command Line I can see the following things * [root@astrisks ~]# *dahdi_cfg -vv* DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == Channel map: Channel 01: FXS Loopstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to none [root@astrisks ~]# *dahdi_scan* [1] active=yes alarms=OK description=Wildcard X100P Board 1 name=WCFXO/0 manufacturer=Digium devicetype=Wildcard X100P location=PCI Bus 02 Slot 02 basechan=1 totchans=1 irq=193 type=analog port=1,FXO *Asterisk CLI* *astrisks*CLI dahdi show status* Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard X100P Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) * output when i dialing to a local number* Connected to Asterisk 1.6.2.11 currently running on astrisks (pid = 2890) Verbosity is at least 3 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [s@from-internal:1] Macro(SIP/199-003a, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-003a' in macro 'hangupcall' == Spawn extension (from-internal, s, 1) exited non-zero on 'SIP/199-003a' -- Executing [h@from-internal:1] Macro(SIP/199-003a, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-003a' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/199-003a' On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind govoi...@gmail.com wrote: Some CLI logs will get you better help on the issue ! also paste the FXO configurations and how you configured it ! On Wed, Sep 28, 2011 at 2:11 PM, michael k mich...@inapp.com wrote: Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before connecting to the FXO card, i am getting a ringing. But i get a message like the number is out of order when i just connect the line to FXO card. Please some one help me to resolve his issue -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk?
Re: [asterisk-users] Increasing volume ?
Hi, please use it then it will be helpfull for your application exten = 66,1,Answer() exten = 66,n,Set(CHANNEL(txgain)=20) exten = 66,n,Set(CHANNEL(rxgain)=20) exten = 66,n,Hangup() On Thu, Aug 11, 2011 at 3:26 AM, Kevin P. Fleming kpflem...@digium.comwrote: On 08/03/2011 08:47 PM, Matt Riddell wrote: On 4/08/11 2:12 AM, Zeeshan Ali Shah wrote: Hi, I am running asterisk with konference . tried to increase the conference voice but not success i tried to add in diaplain SetGlobalVar(Set(VOLUME(TX)=**10)) SetGlobalVar(Set(VOLUME(RX)=**10)) Should be: SetGlobalVar(VOLUME(TX)=10) SetGlobalVar(VOLUME(RX)=10) Dialplan functions cannot be set globally. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- 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] PSTN connectivity
Actually its easier. I haven't worked on FreePBX lately so what I remember is here: You've created a Zap trunk. Focus on the Dial Rule - You can keep it empty as well. Then you've created an outbound route its dial-rule is important. But the funny thing which I didn't mention before is that you've ZAP defined in FreePBX but actually its DAHDI so I remember they've this cute parameter in amportal.conf which tells FreePBX to convert ZAP into DAHDI. On Thu, Sep 29, 2011 at 11:57 AM, michael k mich...@inapp.com wrote: Can you please figure out the configuration issue in my freepbx ? On Thu, Sep 29, 2011 at 11:35 AM, Sam Govind govoi...@gmail.com wrote: The Call at this point is not even looking for FXO/Dahdi/Zap.. See the CLI. there is some misconfiguration in FreePBX and your dialled number is not hitting any dial-able rule. See your FreePBX guide. On Thu, Sep 29, 2011 at 11:01 AM, michael k mich...@inapp.com wrote: Hi, Please see the sample. A ) Analog HardwareType Ports Action FXO Ports 1 Edithttp://192.168.1.134/admin/config.php?type=setupdisplay=dahdidahdi_form=analog_signallingports=fxo FXS Ports -- B) Analog FXO PortsPort 1: *Loop Start* Group: *0* Context: *from-analog * * C ) ZAP Trunk (DAHDI compatibility Mode)* Trunk Description: Outbound Caller ID:CID Options: Maximum Channels: Disable Trunk: Disable Monitor Trunk Failures: Enable Outgoing Dial Rules Dial Rules: 0471+NXX Dial Rules Wizards: Outbound Dial Prefix:Outgoing Settings Zap Identifier (trunk name): *D ) INBOUND route * Description: Extensions: 199 * E ) **OUTBOUND Route* Route Name: 9_outside Route CID: Override Extension CID Route Password: PIN Set: Emergency Dialing: Intra Company Route: Music On Hold? Dial Patterns 8|NXXNXX 8|NXX Dial patterns wizards*: * Trunk SequenceZAP/g0 0 * F ) In command Line I can see the following things * [root@astrisks ~]# *dahdi_cfg -vv* DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == Channel map: Channel 01: FXS Loopstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to none [root@astrisks ~]# *dahdi_scan* [1] active=yes alarms=OK description=Wildcard X100P Board 1 name=WCFXO/0 manufacturer=Digium devicetype=Wildcard X100P location=PCI Bus 02 Slot 02 basechan=1 totchans=1 irq=193 type=analog port=1,FXO *Asterisk CLI* *astrisks*CLI dahdi show status* Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard X100P Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) * output when i dialing to a local number* Connected to Asterisk 1.6.2.11 currently running on astrisks (pid = 2890) Verbosity is at least 3 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [s@from-internal:1] Macro(SIP/199-003a, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-003a' in macro 'hangupcall' == Spawn extension (from-internal, s, 1) exited non-zero on 'SIP/199-003a' -- Executing [h@from-internal:1] Macro(SIP/199-003a, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-003a' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/199-003a' On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind govoi...@gmail.com wrote: Some CLI logs will get you better help on the issue ! also paste the FXO configurations and how you configured it ! On Wed, Sep 28, 2011 at 2:11 PM, michael k mich...@inapp.com wrote: Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in
Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP Abuse)
This is a brilliant idea. How do I contribute my attackers to this
list?
Cheers
Andy
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Huddleston
Sent: 22 September 2011 16:11
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP
Abuse)
Sounds like a great idea.. Hopefully the page/account never gets hacked
and bad IP's published.. I could see a great hack of
127.0.0.1
192.168.0.0/16
10.0.0.0/8
getting up there somehow and next thing you know - BAM!
But I haven't RTFM - I'm guessing there is probably a white list that
supersedes the naughty list.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Thursday, September 22, 2011 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP
Abuse)
very cool!
On Thu, Sep 22, 2011 at 10:37 AM, J. Oquendo aster...@tormenting.net
wrote:
Apologies for cross posting but some of us aren't on the other list
(vice/versa) and thought both groups would benefit.
For those familiar with the VoIP Abuse Project, no need to explain the
gist of this. I got tired of parsing through the alerts (lists) I
receive via email daily. They're long and sometimes I don't have the
time to post them all. So for now, posting VoIP Abuse addresses straight
to Twitter.
So, anyone trying to compromise a pbx, is now autoposted on an hourly
basis to Twitter. Still working on pulling, have about 4 machines linked
up now, will mop em up during the week.
http://twitter.com/#!/voipabuse
Now, you can concoct a quick script off of it, e.g.:
links -dump http://twitter.com/voipabuse;|awk '/attacker/{print
iptables -A INPUT -s $2 -j DROP| sort -u}'
Will get a quickie soon from my Acme's, nCites, etc. when I have time.
For those NOT familiar with it, please Google it as I don't feel like
typing anymore ;) (sorry)
--
=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT, RWSP, GREM
It takes 20 years to build a reputation and five minutes to
ruin it. If you think about that, you'll do things
differently. - Warren Buffett
42B0 5A53 6505 6638 44BB 3943 2BF7 D83F 210A 95AF
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x2BF7D83F210A95AF
--
_
-- 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
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
--
_
-- 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] Limit outbond calls duration to 1 minute
ok thanks it's work fine
now i have one question please
it's work fine when i call extension 222 but i want to call any number from
my sip account 222 and the call hang up after 1 Min
for exemple i call my mobile phone 067XXX using my sip 222 (x-lite) and
the call hangup after 1 min
any help please
thanks and regards
2011/9/28 Tarek Sawah tareksa...@hotmail.com
one adjustment i would suggest is using (|) instead of (,)
exten = 222,n,Dial(SIP/${EXTEN}||KkTtL(6))
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
--
Date: Wed, 28 Sep 2011 18:32:28 +
From: salah.elharit...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
sorry but the issue still the same there is no hangup after 1Min
regards
2011/9/28 Danny Nicholas da...@debsinc.com
As I read this, the following should be correct:
exten = 222,n,Dial(SIP/${EXTEN},,KkTtL(6))
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
elharit
*Sent:* Wednesday, September 28, 2011 1:23 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Limit outbond calls duration to 1 minute**
**
** **
but there is no exemple for when i must put X in order to limit the call**
**
can you please give me an exemple
regards
2011/9/28 Tarek Sawah tareksa...@hotmail.com
have a look at the following:
*L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when 'y' ms are left,
repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional.
source
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
--
Date: Wed, 28 Sep 2011 17:59:27 +
From: salah.elharit...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Limit outbond calls duration to 1 minute
** **
hello list
i have configured a sip account in order to do an outbound calls and i want
to force a hang up after 1 min for 222 sip
in extensions.conf i have
exten = 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 222,n,AbsoluteTimeout(60)
exten = 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 222,n,Dial(SIP/${EXTEN},,KkTt)
exten = 222,n,Hangup();
could you please see this code and tell me waht is wrong
thanks and regards
** **
-- _ --
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
--
_
-- 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
** **
--
_
-- 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
-- _ --
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
--
_
-- 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
--
_
-- 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
Re: [asterisk-users] Limit outbond calls duration to 1 minute
Replace your phone number in place of ${EXTEN} and send it to your outgoing
provider.
with same dial argument.
On Thu, Sep 29, 2011 at 3:09 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
ok thanks it's work fine
now i have one question please
it's work fine when i call extension 222 but i want to call any number
from my sip account 222 and the call hang up after 1 Min
for exemple i call my mobile phone 067XXX using my sip 222 (x-lite) and
the call hangup after 1 min
any help please
thanks and regards
2011/9/28 Tarek Sawah tareksa...@hotmail.com
one adjustment i would suggest is using (|) instead of (,)
exten = 222,n,Dial(SIP/${EXTEN}||KkTtL(6))
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
--
Date: Wed, 28 Sep 2011 18:32:28 +
From: salah.elharit...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
sorry but the issue still the same there is no hangup after 1Min
regards
2011/9/28 Danny Nicholas da...@debsinc.com
As I read this, the following should be correct:
exten = 222,n,Dial(SIP/${EXTEN},,KkTtL(6))
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
elharit
*Sent:* Wednesday, September 28, 2011 1:23 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Limit outbond calls duration to 1 minute*
***
** **
but there is no exemple for when i must put X in order to limit the call*
***
can you please give me an exemple
regards
2011/9/28 Tarek Sawah tareksa...@hotmail.com
have a look at the following:
*L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when 'y' ms are
left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are
optional.
source
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
--
Date: Wed, 28 Sep 2011 17:59:27 +
From: salah.elharit...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Limit outbond calls duration to 1 minute
** **
hello list
i have configured a sip account in order to do an outbound calls and i
want to force a hang up after 1 min for 222 sip
in extensions.conf i have
exten = 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 222,n,AbsoluteTimeout(60)
exten = 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 222,n,Dial(SIP/${EXTEN},,KkTt)
exten = 222,n,Hangup();
could you please see this code and tell me waht is wrong
thanks and regards
** **
-- _
-- 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
--
_
-- 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
** **
--
_
-- 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
-- _
-- 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
--
_
-- 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] Limit outbond calls duration to 1 minute
(top-posting mess fixed the lazy man's way .) On Thursday 29 September 2011, salaheddine elharit wrote: ok thanks it's work fine now i have one question please it's work fine when i call extension 222 but i want to call any number from my sip account 222 and the call hang up after 1 Min for exemple i call my mobile phone 067XXX using my sip 222 (x-lite) and the call hangup after 1 min any help please What you have to do is create a new context in extensions.conf, and specify this in sip.conf as the default context from extension 222. Then, use the same KkTtL(6) options to your Dial() command(s) within this context. If there are some numbers that you want to be able to make unlimited-length calls to (other SIP phones that don't require going out via the PSTN, for example), just give them their own extension(s) without the KkTlL(6) . Remember, Asterisk always tries to match hardest first, i.e. fewest wild card characters first, irrespective of the actual order of lines in extensions.conf. -- AJS Answers come *after* questions. -- _ -- 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] Limit outbond calls duration to 1 minute
ok thanks for your response i will try that and i will update you as soon as i have any result best regards 2011/9/29 A J Stiles asterisk_l...@earthshod.co.uk (top-posting mess fixed the lazy man's way .) On Thursday 29 September 2011, salaheddine elharit wrote: ok thanks it's work fine now i have one question please it's work fine when i call extension 222 but i want to call any number from my sip account 222 and the call hang up after 1 Min for exemple i call my mobile phone 067XXX using my sip 222 (x-lite) and the call hangup after 1 min any help please What you have to do is create a new context in extensions.conf, and specify this in sip.conf as the default context from extension 222. Then, use the same KkTtL(6) options to your Dial() command(s) within this context. If there are some numbers that you want to be able to make unlimited-length calls to (other SIP phones that don't require going out via the PSTN, for example), just give them their own extension(s) without the KkTlL(6) . Remember, Asterisk always tries to match hardest first, i.e. fewest wild card characters first, irrespective of the actual order of lines in extensions.conf. -- AJS Answers come *after* questions. -- _ -- 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 -- _ -- 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] PSTN connectivity
Thanks for the update. but how do i resolve this issue ? can you help me please ? On Thu, Sep 29, 2011 at 1:00 PM, Sam Govind govoi...@gmail.com wrote: Actually its easier. I haven't worked on FreePBX lately so what I remember is here: You've created a Zap trunk. Focus on the Dial Rule - You can keep it empty as well. Then you've created an outbound route its dial-rule is important. But the funny thing which I didn't mention before is that you've ZAP defined in FreePBX but actually its DAHDI so I remember they've this cute parameter in amportal.conf which tells FreePBX to convert ZAP into DAHDI. On Thu, Sep 29, 2011 at 11:57 AM, michael k mich...@inapp.com wrote: Can you please figure out the configuration issue in my freepbx ? On Thu, Sep 29, 2011 at 11:35 AM, Sam Govind govoi...@gmail.com wrote: The Call at this point is not even looking for FXO/Dahdi/Zap.. See the CLI. there is some misconfiguration in FreePBX and your dialled number is not hitting any dial-able rule. See your FreePBX guide. On Thu, Sep 29, 2011 at 11:01 AM, michael k mich...@inapp.com wrote: Hi, Please see the sample. A ) Analog HardwareType Ports Action FXO Ports 1 Edithttp://192.168.1.134/admin/config.php?type=setupdisplay=dahdidahdi_form=analog_signallingports=fxo FXS Ports -- B) Analog FXO PortsPort 1: *Loop Start* Group: *0* Context: *from-analog * * C ) ZAP Trunk (DAHDI compatibility Mode)* Trunk Description: Outbound Caller ID:CID Options: Maximum Channels: Disable Trunk: Disable Monitor Trunk Failures: Enable Outgoing Dial Rules Dial Rules: 0471+NXX Dial Rules Wizards: Outbound Dial Prefix:Outgoing Settings Zap Identifier (trunk name): *D ) INBOUND route * Description: Extensions: 199 * E ) **OUTBOUND Route* Route Name: 9_outside Route CID: Override Extension CID Route Password: PIN Set: Emergency Dialing: Intra Company Route: Music On Hold? Dial Patterns 8|NXXNXX 8|NXX Dial patterns wizards*: * Trunk SequenceZAP/g0 0 * F ) In command Line I can see the following things * [root@astrisks ~]# *dahdi_cfg -vv* DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == Channel map: Channel 01: FXS Loopstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to none [root@astrisks ~]# *dahdi_scan* [1] active=yes alarms=OK description=Wildcard X100P Board 1 name=WCFXO/0 manufacturer=Digium devicetype=Wildcard X100P location=PCI Bus 02 Slot 02 basechan=1 totchans=1 irq=193 type=analog port=1,FXO *Asterisk CLI* *astrisks*CLI dahdi show status* Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard X100P Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) * output when i dialing to a local number* Connected to Asterisk 1.6.2.11 currently running on astrisks (pid = 2890) Verbosity is at least 3 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [s@from-internal:1] Macro(SIP/199-003a, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-003a' in macro 'hangupcall' == Spawn extension (from-internal, s, 1) exited non-zero on 'SIP/199-003a' -- Executing [h@from-internal:1] Macro(SIP/199-003a, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-003a' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/199-003a' On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind govoi...@gmail.com wrote: Some CLI logs will get you better help on the issue ! also paste the FXO configurations and how you configured it ! On Wed, Sep 28, 2011 at 2:11 PM, michael k mich...@inapp.com wrote: Hi All, I am trying to connect my asterisk box with freepbx to PSTN.
[asterisk-users] Problem with Queue Stats
Hi, We are trying to get working Queue Stats, but it seems we get stucked. Is anyone using this project for to track agent statistics ? http://www.asteriskguru.com/tools/queue_stats.php We are managed to install Queue Stats version 0.3 despite of the fact that its using Zend framework, which is not compatible with PHP version is latest stable Debian. Anyway, we are facing following problem now, on main page of project I am getting : The database in empty, but its not empty. Access to database in ./include/config.inc.php and in ./log/config.inc.php has been configured correctly. If anyone knows how to solve it, please give me a hand or if you know other Opensource application which do the same job just advise what I can use. Beside of this error i see bunch of errors in apache log file. [Thu Sep 29 14:18:24 2011] [error] [client 192.168.0.2] PHP Notice: Undefined index: ACCESS in /usr/share/queue_stats/public/error.php on line 26 [Thu Sep 29 14:18:24 2011] [error] [client 192.168.0.2] PHP Notice: Undefined variable: queue_select_in in /usr/share/queue_stats/public/error.php on line 29 [Thu Sep 29 14:18:24 2011] [error] [client 192.168.0.2] PHP Warning: pg_query(): Query failed: ERROR: invalid input syntax for integer: \nLINE 12: aq.access_sid = ''\n ^ in /usr/share/queue_stats/public/error.php on line 61 [Thu Sep 29 14:18:24 2011] [error] [client 192.168.0.2] PHP Warning: pg_fetch_object() expects parameter 1 to be resource, boolean given in /usr/share/queue_stats/public/error.php on line 62 [Thu Sep 29 14:18:24 2011] [error] [client 192.168.0.2] PHP Notice: Undefined index: ACCESS in /usr/share/queue_stats/public/error.php on line 152 [Thu Sep 29 14:18:24 2011] [error] [client 192.168.0.2] PHP Notice: Undefined variable: queue in /usr/share/queue_stats/public/error.php on line 157 [Thu Sep 29 14:18:24 2011] [error] [client 192.168.0.2] PHP Warning: Invalid argument supplied for foreach() in /usr/share/queue_stats/public/error.php on line 157 192.168.0.2 - - [29/Sep/2011:14:18:24 +0200] GET /public/error.php?warning=The%20database%20is%20empty!type=incoming HTTP/1.1 200 2124 - Mozilla/5.0 (X11; Linux x86_64; rv:6.0.2) Gecko/20100101 Firefox/6.0.2 192.168.0.2 - - [29/Sep/2011:14:18:28 +0200] POST /public/incoming_general.php HTTP/1.1 302 497 http://queue_stats.omicrons.pl/public/error.php?warning=The%20database%20is%20empty!type=incoming; Mozilla/5.0 (X11; Linux x86_64; rv:6.0.2) Gecko/20100101 Firefox/6.0.2 [Thu Sep 29 14:18:28 2011] [error] [client 192.168.0.2] PHP Notice: Undefined index: ACCESS in /usr/share/queue_stats/public/error.php on line 26, referer: http://queue_stats.omicrons.pl/public/error.php?warning=The%20database%20is%20empty!type=incoming [Thu Sep 29 14:18:28 2011] [error] [client 192.168.0.2] PHP Notice: Undefined variable: queue_select_in in /usr/share/queue_stats/public/error.php on line 29, referer: http://queue_stats.omicrons.pl/public/error.php?warning=The%20database%20is%20empty!type=incoming [Thu Sep 29 14:18:28 2011] [error] [client 192.168.0.2] PHP Warning: pg_query(): Query failed: ERROR: invalid input syntax for integer: \nLINE 12: aq.access_sid = ''\n ^ in /usr/share/queue_stats/public/error.php on line 61, referer: http://queue_stats.omicrons.pl/public/error.php?warning=The%20database%20is%20empty!type=incoming [Thu Sep 29 14:18:28 2011] [error] [client 192.168.0.2] PHP Warning: pg_fetch_object() expects parameter 1 to be resource, boolean given in /usr/share/queue_stats/public/error.php on line 62, referer: http://queue_stats.omicrons.pl/public/error.php?warning=The%20database%20is%20empty!type=incoming [Thu Sep 29 14:18:28 2011] [error] [client 192.168.0.2] PHP Notice: Undefined index: ACCESS in /usr/share/queue_stats/public/error.php on line 152, referer: http://queue_stats.omicrons.pl/public/error.php?warning=The%20database%20is%20empty!type=incoming [Thu Sep 29 14:18:28 2011] [error] [client 192.168.0.2] PHP Notice: Undefined variable: queue in /usr/share/queue_stats/public/error.php on line 157, referer: http://queue_stats.omicrons.pl/public/error.php?warning=The%20database%20is%20empty!type=incoming [Thu Sep 29 14:18:28 2011] [error] [client 192.168.0.2] PHP Warning: Invalid argument supplied for foreach() in /usr/share/queue_stats/public/error.php on line 157, referer: http://queue_stats.omicrons.pl/public/error.php?warning=The%20database%20is%20empty!type=incoming 192.168.0.2 - - [29/Sep/2011:14:18:28 +0200] GET /public/error.php?warning=The%20database%20is%20empty!type=incoming HTTP/1.1 200 2124 http://queue_stats.omicrons.pl/public/error.php?warning=The%20database%20is%20empty!type=incoming; Mozilla/5.0 (X11; Linux x86_64; rv:6.0.2) Gecko/20100101 Firefox/6.0.2 Thanks and regards, Robert -- _ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] PSTN connectivity
michael k wrote: Thanks for the update. but how do i resolve this issue ? can you help me please ? Can you PLEASE take this to the FreePBX support group? It seems obvious to most that therein lies the problem You are thinking you wish to dial out through the X100, but Asterisk is attempting to dial out on a non existent SIP connection Something isn't right in your dialplan, created by FreePBX Also, no echo canceller on the X100 card isn't wise, but you will not realize that until you are able to use it! John Novack On Thu, Sep 29, 2011 at 1:00 PM, Sam Govind govoi...@gmail.com mailto:govoi...@gmail.com wrote: Actually its easier. I haven't worked on FreePBX lately so what I remember is here: You've created a Zap trunk. Focus on the Dial Rule - You can keep it empty as well. Then you've created an outbound route its dial-rule is important. But the funny thing which I didn't mention before is that you've ZAP defined in FreePBX but actually its DAHDI so I remember they've this cute parameter in amportal.conf which tells FreePBX to convert ZAP into DAHDI. snip -- Dog is my Co-pilot -- _ -- 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] No Bull Service Providers
Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- 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] No Bull Service Providers
This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Bull Service Providers Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- 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 -- _ -- 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] No Bull Service Providers
There is a commercial list! Sorry about that Nick. On Thu, Sep 29, 2011 at 10:47 AM, Danny Nicholas da...@debsinc.com wrote: This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Bull Service Providers Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- 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 -- _ -- 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 -- _ -- 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] No Bull Service Providers
In my professional opinion, the phrases I don't want no Bull service and I want the cheapest service are total contradictions. Down the road something is not going to give. C. Savinovich On September 29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com wrote: This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Bull Service Providers Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- 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 -- _ -- 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-usersChristian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com-- _ -- 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] No Bull Service Providers
Very true... But there should be an equilibrium, the relaiable service, and aggressive pricing comes to meet? Guys please share your experiences. Cheers, Nick On Thu, Sep 29, 2011 at 10:58 AM, C. Savinovich c.savinov...@itntelecom.com wrote: In my professional opinion, the phrases I don't want no Bull service and I want the cheapest service are total contradictions. Down the road something is not going to give. C. Savinovich On September 29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com wrote: This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Bull Service Providers Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- 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 -- _ -- 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 Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- 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 -- _ -- 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] No Bull Service Providers
They aren't everywhere, but we have had good experience with Voicepulse and their rate is typically less than $0.015 per minute. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No Bull Service Providers Very true... But there should be an equilibrium, the relaiable service, and aggressive pricing comes to meet? Guys please share your experiences. Cheers, Nick On Thu, Sep 29, 2011 at 10:58 AM, C. Savinovich c.savinov...@itntelecom.com wrote: In my professional opinion, the phrases I don't want no Bull service and I want the cheapest service are total contradictions. Down the road something is not going to give. C. Savinovich On September 29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com wrote: This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Bull Service Providers Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- 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 -- _ -- 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 Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- 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 -- _ -- 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 -- _ -- 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] No Bull Service Providers
for some reason i don't think (unlimited incoming channels) fits with (dirt cheap DIDs) as you will be abusing their network .. they should start charging per minute .. or you should pay for extra channels several DID providers would offer you 20 channels per did at some rate of 9$ a month per did.. 5 Euros per month and you should pay Extra for Extra channels.. could be the same amount for the same amount of channels Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Thu, 29 Sep 2011 11:09:10 -0400 From: sym...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No Bull Service Providers Very true... But there should be an equilibrium, the relaiable service, and aggressive pricing comes to meet? Guys please share your experiences. Cheers, Nick On Thu, Sep 29, 2011 at 10:58 AM, C. Savinovich c.savinov...@itntelecom.com wrote: In my professional opinion, the phrases I don't want no Bull service and I want the cheapest service are total contradictions. Down the road something is not going to give. C. Savinovich On September 29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com wrote: This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Bull Service Providers Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- 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 -- _ -- 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 Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- 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 -- _ -- 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 -- _ -- 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] No Bull Service Providers
I should have mentioned we are interested in international long distance. That will be a big part of our business. Cheers, Nick. On Thu, Sep 29, 2011 at 11:12 AM, Danny Nicholas da...@debsinc.com wrote: They aren't everywhere, but we have had good experience with Voicepulse and their rate is typically less than $0.015 per minute. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No Bull Service Providers Very true... But there should be an equilibrium, the relaiable service, and aggressive pricing comes to meet? Guys please share your experiences. Cheers, Nick On Thu, Sep 29, 2011 at 10:58 AM, C. Savinovich c.savinov...@itntelecom.com wrote: In my professional opinion, the phrases I don't want no Bull service and I want the cheapest service are total contradictions. Down the road something is not going to give. C. Savinovich On September 29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com wrote: This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Bull Service Providers Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- 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 -- _ -- 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 Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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] PSTN connectivity
On Thu, Sep 29, 2011 at 7:51 AM, michael k mich...@inapp.com wrote: Thanks for the update. but how do i resolve this issue ? can you help me please ? You didn't provide a full CLI trace of the outgoing call, you only supplied the hangup portion of the call. Please try again. Also, what are the dialing rules like in your country? You only have outbound dial patterns setup to handle North American numbers (8+ NXXNXX or 8+ NXX). The Dial Pattern box in the Outbound Rules box is where you define what numbers you want to go out over this trunk. If you dial a number that doesn't match one of these patterns, FreePBX is going to look internally for a dial pattern to match against, and if it doesn't find one there, it will end the call. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- 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] No Bull Service Providers
What does (international long) mean exactly? are you a calling cards company? if so you should look for some company that will be charging you like 0.004 Cents per minute.. and you can find companies that will add more channels to your DID. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Thu, 29 Sep 2011 11:15:13 -0400 From: sym...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No Bull Service Providers I should have mentioned we are interested in international long distance. That will be a big part of our business. Cheers, Nick. On Thu, Sep 29, 2011 at 11:12 AM, Danny Nicholas da...@debsinc.com wrote: They aren't everywhere, but we have had good experience with Voicepulse and their rate is typically less than $0.015 per minute. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No Bull Service Providers Very true... But there should be an equilibrium, the relaiable service, and aggressive pricing comes to meet? Guys please share your experiences. Cheers, Nick On Thu, Sep 29, 2011 at 10:58 AM, C. Savinovich c.savinov...@itntelecom.com wrote: In my professional opinion, the phrases I don't want no Bull service and I want the cheapest service are total contradictions. Down the road something is not going to give. C. Savinovich On September 29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com wrote: This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Bull Service Providers Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- 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 -- _ -- 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 Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] No Bull Service Providers
On Thu, 29 Sep 2011, Nick Khamis wrote: I should have mentioned we are interested in international long distance. That will be a big part of our business. It sounds like you are intending to start a calling card company. Good luck - the competition is fierce, and you will be competing against companies that outright lie about the capacity of their cards, and use stolen minutes to fulfill them as often as they can. If you intend to do wholesale by reselling, you don't need to use the same company for inbound and outbound. In fact for outbound you will probably have many upstream providers, as your goal will be to find the cheapest reliable route in every case. You will need to code something for route selection (I did this in C/AGI). For inbound I use IP Comms, which has worked well. Unlimited inbound per DID, but NOT unlimited channels. For outbound I had arrangements with STi, Voipjet, and many others for smaller route sets. I also participated in the wholesale market at Arbinet, who just got bought out by someone... Cheers, j On Thu, Sep 29, 2011 at 11:12 AM, Danny Nicholas da...@debsinc.com wrote: They aren't everywhere, but we have had good experience with Voicepulse and their rate is typically less than $0.015 per minute. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No Bull Service Providers Very true... But there should be an equilibrium, the relaiable service, and aggressive pricing comes to meet? Guys please share your experiences. Cheers, Nick On Thu, Sep 29, 2011 at 10:58 AM, C. Savinovich c.savinov...@itntelecom.com wrote: In my professional opinion, the phrases I don't want no Bull service and I want the cheapest service are total contradictions. Down the road something is not going to give. C. Savinovich On September 29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com wrote: This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Bull Service Providers Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- 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 -- _ -- 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 Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- 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 -- _ -- 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 -- _ -- 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] No Bull Service Providers
Hello Tarek, For channels, usually they charge per additional channels. I guess being more explicit what it comes down to is: * Reliable service * Agressive Pricing * For DIDs - International Coverage - Per Aditional Channel Pricing * For SIP Termination - International Rates - Per additional trunk pricing We are looking to provide large scale long distance service to thrid world countries such as Sri Lanka, Philippines, India, Pakistan etc... So would require DID for those reagions with the channel support, and sip termintation to Canada and the US with trunk support. Nick. On Thu, Sep 29, 2011 at 11:13 AM, Tarek Sawah tareksa...@hotmail.com wrote: for some reason i don't think (unlimited incoming channels) fits with (dirt cheap DIDs) as you will be abusing their network .. they should start charging per minute .. or you should pay for extra channels several DID providers would offer you 20 channels per did at some rate of 9$ a month per did.. 5 Euros per month and you should pay Extra for Extra channels.. could be the same amount for the same amount of channels Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Thu, 29 Sep 2011 11:09:10 -0400 From: sym...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No Bull Service Providers Very true... But there should be an equilibrium, the relaiable service, and aggressive pricing comes to meet? Guys please share your experiences. Cheers, Nick On Thu, Sep 29, 2011 at 10:58 AM, C. Savinovich c.savinov...@itntelecom.com wrote: In my professional opinion, the phrases I don't want no Bull service and I want the cheapest service are total contradictions. Down the road something is not going to give. C. Savinovich On September 29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com wrote: This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Bull Service Providers Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- 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 -- _ -- 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 Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- 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 -- _ -- 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 -- _ -- 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] No Bull Service Providers
I have no knowledge of any commercial brand that operates in that region and would offer DIDs in those countries.. AND your channel requirements are a bit limited by technology in those regions... and VoIP termination Legislation in those countries whether they allow Calling Cards business, allow DID sales. those issues have more effect on your business. could have helped in US DIDs.. but in Asia i'm no aware of the presence of such providers. however TATACOMMUNICATIONS is the largest VoIP Operating entity in that region and you may find some luck contacting them? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Thu, 29 Sep 2011 11:24:43 -0400 From: sym...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No Bull Service Providers Hello Tarek, For channels, usually they charge per additional channels. I guess being more explicit what it comes down to is: * Reliable service * Agressive Pricing * For DIDs - International Coverage - Per Aditional Channel Pricing * For SIP Termination - International Rates - Per additional trunk pricing We are looking to provide large scale long distance service to thrid world countries such as Sri Lanka, Philippines, India, Pakistan etc... So would require DID for those reagions with the channel support, and sip termintation to Canada and the US with trunk support. Nick. On Thu, Sep 29, 2011 at 11:13 AM, Tarek Sawah tareksa...@hotmail.com wrote: for some reason i don't think (unlimited incoming channels) fits with (dirt cheap DIDs) as you will be abusing their network .. they should start charging per minute .. or you should pay for extra channels several DID providers would offer you 20 channels per did at some rate of 9$ a month per did.. 5 Euros per month and you should pay Extra for Extra channels.. could be the same amount for the same amount of channels Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Thu, 29 Sep 2011 11:09:10 -0400 From: sym...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No Bull Service Providers Very true... But there should be an equilibrium, the relaiable service, and aggressive pricing comes to meet? Guys please share your experiences. Cheers, Nick On Thu, Sep 29, 2011 at 10:58 AM, C. Savinovich c.savinov...@itntelecom.com wrote: In my professional opinion, the phrases I don't want no Bull service and I want the cheapest service are total contradictions. Down the road something is not going to give. C. Savinovich On September 29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com wrote: This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Bull Service Providers Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- 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 -- _ -- 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 Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every
Re: [asterisk-users] No Bull Service Providers
Hello Jeff, There will always be fierce competition, we are starting of with prepaid for an obvious source of quick revenue, we will also be rolling out a few more products in the next year.. It seems like they LIE about their LD rates. A company in Australia was charged with this not too long ago. Stolen minutes? Not that I would be interested in stealing! I just want to be educated in such an act. Of course we don't have to use the same in/outbound providers. I should have been clearer about that. You mentioned Least Cost Route/Rate (LCR), any reason why you did not use what is already out there? Provided by a2billing etc...? We can also implement something using AGI if needed Nick. On Thu, Sep 29, 2011 at 11:21 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Thu, 29 Sep 2011, Nick Khamis wrote: I should have mentioned we are interested in international long distance. That will be a big part of our business. It sounds like you are intending to start a calling card company. Good luck - the competition is fierce, and you will be competing against companies that outright lie about the capacity of their cards, and use stolen minutes to fulfill them as often as they can. If you intend to do wholesale by reselling, you don't need to use the same company for inbound and outbound. In fact for outbound you will probably have many upstream providers, as your goal will be to find the cheapest reliable route in every case. You will need to code something for route selection (I did this in C/AGI). For inbound I use IP Comms, which has worked well. Unlimited inbound per DID, but NOT unlimited channels. For outbound I had arrangements with STi, Voipjet, and many others for smaller route sets. I also participated in the wholesale market at Arbinet, who just got bought out by someone... Cheers, j On Thu, Sep 29, 2011 at 11:12 AM, Danny Nicholas da...@debsinc.com wrote: They aren't everywhere, but we have had good experience with Voicepulse and their rate is typically less than $0.015 per minute. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No Bull Service Providers Very true... But there should be an equilibrium, the relaiable service, and aggressive pricing comes to meet? Guys please share your experiences. Cheers, Nick On Thu, Sep 29, 2011 at 10:58 AM, C. Savinovich c.savinov...@itntelecom.com wrote: In my professional opinion, the phrases I don't want no Bull service and I want the cheapest service are total contradictions. Down the road something is not going to give. C. Savinovich On September 29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com wrote: This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Bull Service Providers Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- 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 -- _ -- 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 Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- 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:
Re: [asterisk-users] No Bull Service Providers
Hello Tarek, Thanks again! I will look into TATA. Let's hope their comunication is better than their cars ;). Nick. On Thu, Sep 29, 2011 at 11:38 AM, Nick Khamis sym...@gmail.com wrote: Hello Jeff, There will always be fierce competition, we are starting of with prepaid for an obvious source of quick revenue, we will also be rolling out a few more products in the next year.. It seems like they LIE about their LD rates. A company in Australia was charged with this not too long ago. Stolen minutes? Not that I would be interested in stealing! I just want to be educated in such an act. Of course we don't have to use the same in/outbound providers. I should have been clearer about that. You mentioned Least Cost Route/Rate (LCR), any reason why you did not use what is already out there? Provided by a2billing etc...? We can also implement something using AGI if needed Nick. On Thu, Sep 29, 2011 at 11:21 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Thu, 29 Sep 2011, Nick Khamis wrote: I should have mentioned we are interested in international long distance. That will be a big part of our business. It sounds like you are intending to start a calling card company. Good luck - the competition is fierce, and you will be competing against companies that outright lie about the capacity of their cards, and use stolen minutes to fulfill them as often as they can. If you intend to do wholesale by reselling, you don't need to use the same company for inbound and outbound. In fact for outbound you will probably have many upstream providers, as your goal will be to find the cheapest reliable route in every case. You will need to code something for route selection (I did this in C/AGI). For inbound I use IP Comms, which has worked well. Unlimited inbound per DID, but NOT unlimited channels. For outbound I had arrangements with STi, Voipjet, and many others for smaller route sets. I also participated in the wholesale market at Arbinet, who just got bought out by someone... Cheers, j On Thu, Sep 29, 2011 at 11:12 AM, Danny Nicholas da...@debsinc.com wrote: They aren't everywhere, but we have had good experience with Voicepulse and their rate is typically less than $0.015 per minute. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No Bull Service Providers Very true... But there should be an equilibrium, the relaiable service, and aggressive pricing comes to meet? Guys please share your experiences. Cheers, Nick On Thu, Sep 29, 2011 at 10:58 AM, C. Savinovich c.savinov...@itntelecom.com wrote: In my professional opinion, the phrases I don't want no Bull service and I want the cheapest service are total contradictions. Down the road something is not going to give. C. Savinovich On September 29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com wrote: This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Bull Service Providers Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- 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 -- _ -- 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 Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] No Bull Service Providers
On Thu, 29 Sep 2011, Nick Khamis wrote: Hello Jeff, There will always be fierce competition, we are starting of with prepaid for an obvious source of quick revenue, we will also be rolling out a few more products in the next year.. It seems like they LIE about their LD rates. A company in Australia was charged with this not too long ago. Stolen minutes? Not that I would be interested in stealing! I just want to be educated in such an act. You will often see discussion on this list about asterisk servers being compromised and the result being very expensive calls placed until the compromise is noticed and shutdown. Those calls are placed by nefarious wholesalers that take advantage of the free routes they manage to find as long as possible. Hard to compete against free! Other games the calling card companies play - they will release a card with unbelievable rates so that it quickly gains market share, then slowly back off the minutes offered by the card (without changing the rate sheets of course) until it is noticed by the consumers, who stop buying it. Then that card is discontinued and another is produced in the same manner. You will notice on the calling card shelves there are only a handful of companies producing lots of different cards. There are many more tricks they use to dupe the consumers and stifle competition. Hidden or non-disclosed connection rates, maintenance fees charged every few days to burn off credit on the cards, time restrictions on the lower rates, etc. We actually produced a card once we called TRUTH (which was honest about rates, had no hidden fees, etc) and it sold ok for a while, but when the card next to it on the shelf claims twice the minutes for the same $$$, eventually they win. In the end this business doesn't make money unless you are selling millions of minutes per month, and even then the margins are slim and you have to play the same games to compete. What we thought would be a fairly easy business to run became a maintenance nightmare, and a single instance of fraud could wipe out months worth of profits. A2billing didn't exist when we started, so we rolled our own. Seems pretty popular now - maybe it would work well for you. Good luck, j Of course we don't have to use the same in/outbound providers. I should have been clearer about that. You mentioned Least Cost Route/Rate (LCR), any reason why you did not use what is already out there? Provided by a2billing etc...? We can also implement something using AGI if needed Nick. On Thu, Sep 29, 2011 at 11:21 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Thu, 29 Sep 2011, Nick Khamis wrote: I should have mentioned we are interested in international long distance. That will be a big part of our business. It sounds like you are intending to start a calling card company. Good luck - the competition is fierce, and you will be competing against companies that outright lie about the capacity of their cards, and use stolen minutes to fulfill them as often as they can. If you intend to do wholesale by reselling, you don't need to use the same company for inbound and outbound. In fact for outbound you will probably have many upstream providers, as your goal will be to find the cheapest reliable route in every case. You will need to code something for route selection (I did this in C/AGI). For inbound I use IP Comms, which has worked well. Unlimited inbound per DID, but NOT unlimited channels. For outbound I had arrangements with STi, Voipjet, and many others for smaller route sets. I also participated in the wholesale market at Arbinet, who just got bought out by someone... Cheers, j On Thu, Sep 29, 2011 at 11:12 AM, Danny Nicholas da...@debsinc.com wrote: They aren't everywhere, but we have had good experience with Voicepulse and their rate is typically less than $0.015 per minute. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No Bull Service Providers Very true... But there should be an equilibrium, the relaiable service, and aggressive pricing comes to meet? Guys please share your experiences. Cheers, Nick On Thu, Sep 29, 2011 at 10:58 AM, C. Savinovich c.savinov...@itntelecom.com wrote: In my professional opinion, the phrases I don't want no Bull service and I want the cheapest service are total contradictions. Down the road something is not going to give. C. Savinovich On September 29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com wrote: This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing
Re: [asterisk-users] No Bull Service Providers
You will notice on the calling card shelves there are only a handful of companies producing lots of different cards. I have! That's what led me to CC for starters, then implementing a more novel startup product. But. Regardless of all the corruption, my goal is to offer something honest TRUTH, I like that ;), reliable and as consistent as possible. We cannot compete against free, but we can try our best. Again, CC is just an entry point, we can doing this like: speech to text - Natural Language Processing (NLP) - text to speech. Bringing computer science to VoIP. This is our long term.. I just need to keep the investor happy for now.. Nick On Thu, Sep 29, 2011 at 11:48 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Thu, 29 Sep 2011, Nick Khamis wrote: Hello Jeff, There will always be fierce competition, we are starting of with prepaid for an obvious source of quick revenue, we will also be rolling out a few more products in the next year.. It seems like they LIE about their LD rates. A company in Australia was charged with this not too long ago. Stolen minutes? Not that I would be interested in stealing! I just want to be educated in such an act. You will often see discussion on this list about asterisk servers being compromised and the result being very expensive calls placed until the compromise is noticed and shutdown. Those calls are placed by nefarious wholesalers that take advantage of the free routes they manage to find as long as possible. Hard to compete against free! Other games the calling card companies play - they will release a card with unbelievable rates so that it quickly gains market share, then slowly back off the minutes offered by the card (without changing the rate sheets of course) until it is noticed by the consumers, who stop buying it. Then that card is discontinued and another is produced in the same manner. You will notice on the calling card shelves there are only a handful of companies producing lots of different cards. There are many more tricks they use to dupe the consumers and stifle competition. Hidden or non-disclosed connection rates, maintenance fees charged every few days to burn off credit on the cards, time restrictions on the lower rates, etc. We actually produced a card once we called TRUTH (which was honest about rates, had no hidden fees, etc) and it sold ok for a while, but when the card next to it on the shelf claims twice the minutes for the same $$$, eventually they win. In the end this business doesn't make money unless you are selling millions of minutes per month, and even then the margins are slim and you have to play the same games to compete. What we thought would be a fairly easy business to run became a maintenance nightmare, and a single instance of fraud could wipe out months worth of profits. A2billing didn't exist when we started, so we rolled our own. Seems pretty popular now - maybe it would work well for you. Good luck, j Of course we don't have to use the same in/outbound providers. I should have been clearer about that. You mentioned Least Cost Route/Rate (LCR), any reason why you did not use what is already out there? Provided by a2billing etc...? We can also implement something using AGI if needed Nick. On Thu, Sep 29, 2011 at 11:21 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Thu, 29 Sep 2011, Nick Khamis wrote: I should have mentioned we are interested in international long distance. That will be a big part of our business. It sounds like you are intending to start a calling card company. Good luck - the competition is fierce, and you will be competing against companies that outright lie about the capacity of their cards, and use stolen minutes to fulfill them as often as they can. If you intend to do wholesale by reselling, you don't need to use the same company for inbound and outbound. In fact for outbound you will probably have many upstream providers, as your goal will be to find the cheapest reliable route in every case. You will need to code something for route selection (I did this in C/AGI). For inbound I use IP Comms, which has worked well. Unlimited inbound per DID, but NOT unlimited channels. For outbound I had arrangements with STi, Voipjet, and many others for smaller route sets. I also participated in the wholesale market at Arbinet, who just got bought out by someone... Cheers, j On Thu, Sep 29, 2011 at 11:12 AM, Danny Nicholas da...@debsinc.com wrote: They aren't everywhere, but we have had good experience with Voicepulse and their rate is typically less than $0.015 per minute. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No Bull Service Providers
Re: [asterisk-users] No Bull Service Providers
one thing i'm sure of? Honesty is a waste in this type of business.. all the features youa re talking about .. have been offered and tested with customers.. the bottom like .. when a customer buys a 2$ calling card . he expects to make a call and say his words and hangs up .. all those features won't be of use for him for a card that will allow him to talk as much minutes as he can! you abusing free routes or not.. is not his business actually. those features can be offered to PINLESS customers who can pay 100-300 $ per account! Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Thu, 29 Sep 2011 12:03:26 -0400 From: sym...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No Bull Service Providers You will notice on the calling card shelves there are only a handful of companies producing lots of different cards. I have! That's what led me to CC for starters, then implementing a more novel startup product. But. Regardless of all the corruption, my goal is to offer something honest TRUTH, I like that ;), reliable and as consistent as possible. We cannot compete against free, but we can try our best. Again, CC is just an entry point, we can doing this like: speech to text - Natural Language Processing (NLP) - text to speech. Bringing computer science to VoIP. This is our long term.. I just need to keep the investor happy for now.. Nick On Thu, Sep 29, 2011 at 11:48 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Thu, 29 Sep 2011, Nick Khamis wrote: Hello Jeff, There will always be fierce competition, we are starting of with prepaid for an obvious source of quick revenue, we will also be rolling out a few more products in the next year.. It seems like they LIE about their LD rates. A company in Australia was charged with this not too long ago. Stolen minutes? Not that I would be interested in stealing! I just want to be educated in such an act. You will often see discussion on this list about asterisk servers being compromised and the result being very expensive calls placed until the compromise is noticed and shutdown. Those calls are placed by nefarious wholesalers that take advantage of the free routes they manage to find as long as possible. Hard to compete against free! Other games the calling card companies play - they will release a card with unbelievable rates so that it quickly gains market share, then slowly back off the minutes offered by the card (without changing the rate sheets of course) until it is noticed by the consumers, who stop buying it. Then that card is discontinued and another is produced in the same manner. You will notice on the calling card shelves there are only a handful of companies producing lots of different cards. There are many more tricks they use to dupe the consumers and stifle competition. Hidden or non-disclosed connection rates, maintenance fees charged every few days to burn off credit on the cards, time restrictions on the lower rates, etc. We actually produced a card once we called TRUTH (which was honest about rates, had no hidden fees, etc) and it sold ok for a while, but when the card next to it on the shelf claims twice the minutes for the same $$$, eventually they win. In the end this business doesn't make money unless you are selling millions of minutes per month, and even then the margins are slim and you have to play the same games to compete. What we thought would be a fairly easy business to run became a maintenance nightmare, and a single instance of fraud could wipe out months worth of profits. A2billing didn't exist when we started, so we rolled our own. Seems pretty popular now - maybe it would work well for you. Good luck, j Of course we don't have to use the same in/outbound providers. I should have been clearer about that. You mentioned Least Cost Route/Rate (LCR), any reason why you did not use what is already out there? Provided by a2billing etc...? We can also implement something using AGI if needed Nick. On Thu, Sep 29, 2011 at 11:21 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Thu, 29 Sep 2011, Nick Khamis wrote: I should have mentioned we are interested in international long distance. That will be a big part of our business. It sounds like you are intending to start a calling card company. Good luck - the competition is fierce, and you will be competing against companies that outright lie about the capacity of their cards, and use stolen minutes to fulfill them as often as they can. If you intend to do wholesale by reselling, you don't need to use the same company for inbound and outbound. In fact for outbound you will probably have many upstream providers, as your goal will be to find the cheapest reliable route in
[asterisk-users] record calls of specific agnets
Hello Asterisk List! I have been asked to record calls from specific agents, and I am having difficulty finding if this is possible, and if so, how exactly to do it. Some pertinent info: We are using Asterisk 1.4.31 with T1/PRI/IAX/SIP calls coming inbound. We have about 60 queues, but only a few that will need to be recorded on for this. We use AgentCallbackLogin for agent login. We have in place (but not active currently) the ability to record calls per queue, but in this case, we only want a specific set of agents that are in a given queue to be recorded. The issue seems to be that in this case the recording information is setup with a Set(MONITOR_FILENAME=$filename) statement before the calls are placed in queue, at which point there is no what to know what agent will answer the call. If anyone could offer some advise on this, I would be very grateful! I will also post to the list anything I find. Have a very nice day. -- Thanks, Lyle J. McKarns --- Network Engineering Team n|m Nexus Management 4 Industrial Parkway Suite 101 Brunswick, Maine 04011 Tel (USA) : 1 207 319 1105 Tel (UK) : 0207 100 4968 Fax: 1 207 725 8552 Nexus Management, Inc.│ Registered Office: 4 Industrial Parkway, Suite 101, Brunswick, Maine. 04011│Company No. 19891257D, Registered in Maine│ A member of the Nexus Management Plc group of companies -- _ -- 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] record calls of specific agnets
I would either use a gotoif to determine which queues get recorded or put the recordable queues into a separate context (probably the simpler solution). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle McKarns Sent: Thursday, September 29, 2011 11:22 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] record calls of specific agnets Hello Asterisk List! I have been asked to record calls from specific agents, and I am having difficulty finding if this is possible, and if so, how exactly to do it. Some pertinent info: We are using Asterisk 1.4.31 with T1/PRI/IAX/SIP calls coming inbound. We have about 60 queues, but only a few that will need to be recorded on for this. We use AgentCallbackLogin for agent login. We have in place (but not active currently) the ability to record calls per queue, but in this case, we only want a specific set of agents that are in a given queue to be recorded. The issue seems to be that in this case the recording information is setup with a Set(MONITOR_FILENAME=$filename) statement before the calls are placed in queue, at which point there is no what to know what agent will answer the call. If anyone could offer some advise on this, I would be very grateful! I will also post to the list anything I find. Have a very nice day. -- Thanks, Lyle J. McKarns --- Network Engineering Team n|m Nexus Management 4 Industrial Parkway Suite 101 Brunswick, Maine 04011 Tel (USA) : 1 207 319 1105 Tel (UK) : 0207 100 4968 Fax: 1 207 725 8552 Nexus Management, Inc.│ Registered Office: 4 Industrial Parkway, Suite 101, Brunswick, Maine. 04011│Company No. 19891257D, Registered in Maine│ A member of the Nexus Management Plc group of companies -- _ -- 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] PRI Issues After Upgrade
I'm digging this back up since the problem persists. I've been attempting to figure out what's been going on and I'm at a stopping point again. Even though I swore I checked it, it turns out the two cards and the ethernet controller were all on the same IRQ. I moved the cards around so they are all isolated on their own IRQs, but the same problems persist. I'm getting all of the following: PRI Span: 1 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state 7(Multi-frame established) PRI Span: 1 !! Unknown IE 128 (cs0) -- Span 1: Channel 0/23 got hangup, cause 87 [Sep 29 11:55:51] WARNING[1331]: sig_pri.c:1054 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway! == Primary D-Channel on span 1 up PRI Span: 1 !! Unknown IE 128 (cs0) -- Span 1: Channel 0/23 got hangup, cause 14 [Sep 29 11:52:57] WARNING[15586]: app_dial.c:1452 wait_for_answer: Unable to forward frametype: 2 I've also replaced the cable from the PRI to the card, just in case… Any ideas? Stephen H. Gerstacker Sr. Database Developer Electronic Data Payment Systems Phone: 866.578.9740 ext. 114 Fax: 866.528.3854 www.edpaymentsystems.comhttp://www.edpaymentsystems.com On Sep 14, 2011, at 11:04 AM, Doug Lytle wrote: Stephen H. Gerstacker wrote: Came in this morning to more of the same: Then, if you have the ability, I'd drop 1.2 back into place and see if it's happy. But, my feeling is that you'll need to contact the provider. The other thing that comes to mind is that your PRI card is having issues. Doug -- _ -- 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 -- _ -- 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] record calls of specific agnets
On Thursday 29 September 2011, Lyle McKarns wrote: Hello Asterisk List! I have been asked to record calls from specific agents, and I am having difficulty finding if this is possible, and if so, how exactly to do it. Have a recorded context and an unrecorded context in your dialplan, identical save for the lines that start the recording and cleanup processes being absent from the latter. Then set contexts per extension in sip.conf. -- AJS Answers come *after* questions. -- _ -- 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/DAHDI with Dynamic T1s
Greetings- From time to time, I find myself working with (or customers working with) dynamic T1s. They are typically standard T1s that terminate to an Adtran device which utilizes the channels for data (64kbps X 24) until a call is pushed inbound/outbound on the circuit. One data channel is automatically peeled off the circuit (removing 64kbps from total data throughput capacity), and reallocated as a voice channel. Is it possible for Asterisk/DAHDI to handle a situation such as this? If I recall, DAHDI does have some data functions to it, but I'm not sure if it can handle the circuit as data (presented to kernel for iptables routing/nat), and/or if it can automagically reallocate channels for voice usage on the fly. Thoughts? --Tim -- _ -- 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] Asterisk/DAHDI with Dynamic T1s
On 09/29/2011 12:22 PM, Tim Nelson wrote: Greetings- From time to time, I find myself working with (or customers working with) dynamic T1s. They are typically standard T1s that terminate to an Adtran device which utilizes the channels for data (64kbps X 24) until a call is pushed inbound/outbound on the circuit. One data channel is automatically peeled off the circuit (removing 64kbps from total data throughput capacity), and reallocated as a voice channel. Is it possible for Asterisk/DAHDI to handle a situation such as this? If I recall, DAHDI does have some data functions to it, but I'm not sure if it can handle the circuit as data (presented to kernel for iptables routing/nat), and/or if it can automagically reallocate channels for voice usage on the fly. There are methods of peeling off channels dynamically to be used as data channels (PPP/RAS style), but not the other way around to my knowledge. Once an HDLC network link has been setup in the kernel's HDLC layer, I don't believe it can be shrunk or grown. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] Features not working
Hi all. I could have sworn this working at one time... But it doesn't look like any of the functions provided by features.so is working for me. (one-touch monitoring, attended/blind transfer, etc) I've (re)loaded features.so, as well as bridge_builtin_features.so. The config file looks sane. What else should I try? TIA, -- Take care and have fun, Mike Diehl. -- _ -- 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] [asterik-users] Installing PRI card
Hi, We have got a new PRI card at one of our Office locations and now I need to install the the device on a remote server. Is there any way to know if the device is loaded already. When I give cat /proc/zaptel/* it returns the following. # cat /proc/zaptel/* Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) B8ZS/ESF RED IRQ misses: 2 1 WCT1/0/1 Clear (In use) RED 2 WCT1/0/2 Clear (In use) RED 3 WCT1/0/3 Clear (In use) RED 4 WCT1/0/4 Clear (In use) RED 5 WCT1/0/5 Clear (In use) RED 6 WCT1/0/6 Clear (In use) RED 7 WCT1/0/7 Clear (In use) RED 8 WCT1/0/8 Clear (In use) RED 9 WCT1/0/9 Clear (In use) RED 10 WCT1/0/10 Clear (In use) RED 11 WCT1/0/11 Clear (In use) RED 12 WCT1/0/12 Clear (In use) RED 13 WCT1/0/13 Clear (In use) RED 14 WCT1/0/14 Clear (In use) RED 15 WCT1/0/15 Clear (In use) RED 16 WCT1/0/16 Clear RED 17 WCT1/0/17 Clear (In use) RED 18 WCT1/0/18 Clear (In use) RED 19 WCT1/0/19 Clear (In use) RED 20 WCT1/0/20 Clear (In use) RED 21 WCT1/0/21 Clear (In use) RED 22 WCT1/0/22 Clear (In use) RED 23 WCT1/0/23 Clear (In use) RED 24 WCT1/0/24 HDLCFCS (In use) RED But when I connect to the console, I am unable to give any ZAP related commands. Does this mean that my device is loaded and I just need to load the module. Or do I need to compile asterisk again?? Any help would be highly appreciated. My asterisk version is Asterisk 1.4.19.2 and I am on a Fedora release 9 server. Thanks, Najim -- _ -- 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] [asterik-users] Installing PRI card
On 9/29/2011 2:52 PM, NaJIm wrote: IRQ misses: 2 You are risking lots of audio problems if the card shares the IRQ with any other device. Try and go in the BIOS and disable the other device or change the IRQ it is using so that they do not conflict. What version of zaptel are you running? What zaptel commands have you tried? Have you added any lines to zaptel.conf and zapata.conf? Who is your Telco provider and what signalling are they using on the T1? It looks like there is a control channel on 24, but 16 isn't showing the same status as I would expect. Mike -- _ -- 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] [asterik-users] Installing PRI card
Try module load chan_zap.so in the CLI. You should see whatever errors are generated. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of NaJIm Sent: Thursday, September 29, 2011 5:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] [asterik-users] Installing PRI card Hi, We have got a new PRI card at one of our Office locations and now I need to install the the device on a remote server. Is there any way to know if the device is loaded already. When I give cat /proc/zaptel/* it returns the following. # cat /proc/zaptel/* Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) B8ZS/ESF RED IRQ misses: 2 1 WCT1/0/1 Clear (In use) RED 2 WCT1/0/2 Clear (In use) RED 3 WCT1/0/3 Clear (In use) RED 4 WCT1/0/4 Clear (In use) RED 5 WCT1/0/5 Clear (In use) RED 6 WCT1/0/6 Clear (In use) RED 7 WCT1/0/7 Clear (In use) RED 8 WCT1/0/8 Clear (In use) RED 9 WCT1/0/9 Clear (In use) RED 10 WCT1/0/10 Clear (In use) RED 11 WCT1/0/11 Clear (In use) RED 12 WCT1/0/12 Clear (In use) RED 13 WCT1/0/13 Clear (In use) RED 14 WCT1/0/14 Clear (In use) RED 15 WCT1/0/15 Clear (In use) RED 16 WCT1/0/16 Clear RED 17 WCT1/0/17 Clear (In use) RED 18 WCT1/0/18 Clear (In use) RED 19 WCT1/0/19 Clear (In use) RED 20 WCT1/0/20 Clear (In use) RED 21 WCT1/0/21 Clear (In use) RED 22 WCT1/0/22 Clear (In use) RED 23 WCT1/0/23 Clear (In use) RED 24 WCT1/0/24 HDLCFCS (In use) RED But when I connect to the console, I am unable to give any ZAP related commands. Does this mean that my device is loaded and I just need to load the module. Or do I need to compile asterisk again?? Any help would be highly appreciated. My asterisk version is Asterisk 1.4.19.2 and I am on a Fedora release 9 server. Thanks, Najim -- _ -- 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] [asterik-users] Installing PRI card
Mike, *What version of zaptel are you running? * My Zaptel version is - zaptel-1.4.12.1 *What zaptel commands have you tried? * None of the zaptel commands are working on my CLI. Its like on CLI, none of the commands starting with zap are working. (When I give zap+TAB key nothing shows up) *Have you added any lines to zaptel.conf and zapata.conf? * I am working on a backup server of an already running PBX. In zapata.conf, the only configurations that are there is as below. But I guess those configurations are that of an E1 card and I will have configure it from start. group=1 switchtype=euroisdn signalling=pri_cpe callerid=asreceived usecallerid=yes cidsignalling=dtmf cidstart=ring context=TEST_EXTERNAL channel=1-15 channel=17-31 *Who is your Telco provider and what signalling are they using on the T1? * I am not sure about the signalling they are using. And thanks for the tip on IRQ. As I said I am working on a remote server. I will ask some one over there to change the IRQ value. Regards, Najim On Fri, Sep 30, 2011 at 4:05 AM, Mike Beirne bei...@mgjbnet.com wrote: On 9/29/2011 2:52 PM, NaJIm wrote: IRQ misses: 2 You are risking lots of audio problems if the card shares the IRQ with any other device. Try and go in the BIOS and disable the other device or change the IRQ it is using so that they do not conflict. What version of zaptel are you running? What zaptel commands have you tried? Have you added any lines to zaptel.conf and zapata.conf? Who is your Telco provider and what signalling are they using on the T1? It looks like there is a control channel on 24, but 16 isn't showing the same status as I would expect. Mike -- _ -- 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 -- _ -- 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] [asterik-users] Installing PRI card
Hi Eric, This is the error messages I get I try to load the module. *CLI module load chan_zap.so [Sep 30 04:45:57] WARNING[5182]: pbx.c:2979 ast_register_application: Already have an application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, ISDN PRI signalling -- Registered channel 2, ISDN PRI signalling -- Registered channel 3, ISDN PRI signalling -- Registered channel 4, ISDN PRI signalling -- Registered channel 5, ISDN PRI signalling -- Registered channel 6, ISDN PRI signalling -- Registered channel 7, ISDN PRI signalling -- Registered channel 8, ISDN PRI signalling -- Registered channel 9, ISDN PRI signalling -- Registered channel 10, ISDN PRI signalling -- Registered channel 11, ISDN PRI signalling -- Registered channel 12, ISDN PRI signalling -- Registered channel 13, ISDN PRI signalling -- Registered channel 14, ISDN PRI signalling -- Registered channel 15, ISDN PRI signalling -- Registered channel 17, ISDN PRI signalling -- Registered channel 18, ISDN PRI signalling -- Registered channel 19, ISDN PRI signalling -- Registered channel 20, ISDN PRI signalling -- Registered channel 21, ISDN PRI signalling -- Registered channel 22, ISDN PRI signalling -- Registered channel 23, ISDN PRI signalling [Sep 30 04:45:57] WARNING[5182]: chan_zap.c:905 zt_open: Unable to specify channel 24: Device or resource busy [Sep 30 04:45:57] ERROR[5182]: chan_zap.c:7219 mkintf: Unable to open channel 24: Device or resource busy here = 0, tmp-channel = 24, channel = 24 [Sep 30 04:45:57] ERROR[5182]: chan_zap.c:10582 build_channels: Unable to register channel '17-31' Regards, Najim On Fri, Sep 30, 2011 at 4:24 AM, Eric Wieling ewiel...@nyigc.com wrote: Try module load chan_zap.so in the CLI. You should see whatever errors are generated. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of NaJIm Sent: Thursday, September 29, 2011 5:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] [asterik-users] Installing PRI card Hi, We have got a new PRI card at one of our Office locations and now I need to install the the device on a remote server. Is there any way to know if the device is loaded already. When I give cat /proc/zaptel/* it returns the following. # cat /proc/zaptel/* Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) B8ZS/ESF RED IRQ misses: 2 1 WCT1/0/1 Clear (In use) RED 2 WCT1/0/2 Clear (In use) RED 3 WCT1/0/3 Clear (In use) RED 4 WCT1/0/4 Clear (In use) RED 5 WCT1/0/5 Clear (In use) RED 6 WCT1/0/6 Clear (In use) RED 7 WCT1/0/7 Clear (In use) RED 8 WCT1/0/8 Clear (In use) RED 9 WCT1/0/9 Clear (In use) RED 10 WCT1/0/10 Clear (In use) RED 11 WCT1/0/11 Clear (In use) RED 12 WCT1/0/12 Clear (In use) RED 13 WCT1/0/13 Clear (In use) RED 14 WCT1/0/14 Clear (In use) RED 15 WCT1/0/15 Clear (In use) RED 16 WCT1/0/16 Clear RED 17 WCT1/0/17 Clear (In use) RED 18 WCT1/0/18 Clear (In use) RED 19 WCT1/0/19 Clear (In use) RED 20 WCT1/0/20 Clear (In use) RED 21 WCT1/0/21 Clear (In use) RED 22 WCT1/0/22 Clear (In use) RED 23 WCT1/0/23 Clear (In use) RED 24 WCT1/0/24 HDLCFCS (In use) RED But when I connect to the console, I am unable to give any ZAP related commands. Does this mean that my device is loaded and I just need to load the module. Or do I need to compile asterisk again?? Any help would be highly appreciated. My asterisk version is Asterisk 1.4.19.2 and I am on a Fedora release 9 server. Thanks, Najim -- _ -- 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] [asterik-users] Installing PRI card
Am I getting these error messages due to wrong configurations in my zapata.conf. ?? I have got the following configurations in my zapata-channels.conf. ; Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) B8ZS/ESF RED group=0,11 context=from-pstn switchtype = national signalling = pri_cpe channel = 1-23 group= context=default Najim On Fri, Sep 30, 2011 at 4:51 AM, NaJIm getna...@gmail.com wrote: Hi Eric, This is the error messages I get I try to load the module. *CLI module load chan_zap.so [Sep 30 04:45:57] WARNING[5182]: pbx.c:2979 ast_register_application: Already have an application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, ISDN PRI signalling -- Registered channel 2, ISDN PRI signalling -- Registered channel 3, ISDN PRI signalling -- Registered channel 4, ISDN PRI signalling -- Registered channel 5, ISDN PRI signalling -- Registered channel 6, ISDN PRI signalling -- Registered channel 7, ISDN PRI signalling -- Registered channel 8, ISDN PRI signalling -- Registered channel 9, ISDN PRI signalling -- Registered channel 10, ISDN PRI signalling -- Registered channel 11, ISDN PRI signalling -- Registered channel 12, ISDN PRI signalling -- Registered channel 13, ISDN PRI signalling -- Registered channel 14, ISDN PRI signalling -- Registered channel 15, ISDN PRI signalling -- Registered channel 17, ISDN PRI signalling -- Registered channel 18, ISDN PRI signalling -- Registered channel 19, ISDN PRI signalling -- Registered channel 20, ISDN PRI signalling -- Registered channel 21, ISDN PRI signalling -- Registered channel 22, ISDN PRI signalling -- Registered channel 23, ISDN PRI signalling [Sep 30 04:45:57] WARNING[5182]: chan_zap.c:905 zt_open: Unable to specify channel 24: Device or resource busy [Sep 30 04:45:57] ERROR[5182]: chan_zap.c:7219 mkintf: Unable to open channel 24: Device or resource busy here = 0, tmp-channel = 24, channel = 24 [Sep 30 04:45:57] ERROR[5182]: chan_zap.c:10582 build_channels: Unable to register channel '17-31' Regards, Najim On Fri, Sep 30, 2011 at 4:24 AM, Eric Wieling ewiel...@nyigc.com wrote: Try module load chan_zap.so in the CLI. You should see whatever errors are generated. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of NaJIm Sent: Thursday, September 29, 2011 5:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] [asterik-users] Installing PRI card Hi, We have got a new PRI card at one of our Office locations and now I need to install the the device on a remote server. Is there any way to know if the device is loaded already. When I give cat /proc/zaptel/* it returns the following. # cat /proc/zaptel/* Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) B8ZS/ESF RED IRQ misses: 2 1 WCT1/0/1 Clear (In use) RED 2 WCT1/0/2 Clear (In use) RED 3 WCT1/0/3 Clear (In use) RED 4 WCT1/0/4 Clear (In use) RED 5 WCT1/0/5 Clear (In use) RED 6 WCT1/0/6 Clear (In use) RED 7 WCT1/0/7 Clear (In use) RED 8 WCT1/0/8 Clear (In use) RED 9 WCT1/0/9 Clear (In use) RED 10 WCT1/0/10 Clear (In use) RED 11 WCT1/0/11 Clear (In use) RED 12 WCT1/0/12 Clear (In use) RED 13 WCT1/0/13 Clear (In use) RED 14 WCT1/0/14 Clear (In use) RED 15 WCT1/0/15 Clear (In use) RED 16 WCT1/0/16 Clear RED 17 WCT1/0/17 Clear (In use) RED 18 WCT1/0/18 Clear (In use) RED 19 WCT1/0/19 Clear (In use) RED 20 WCT1/0/20 Clear (In use) RED 21 WCT1/0/21 Clear (In use) RED 22 WCT1/0/22 Clear (In use) RED 23 WCT1/0/23 Clear (In use) RED 24 WCT1/0/24 HDLCFCS (In use) RED But when I connect to the console, I am unable to give any ZAP related commands. Does this mean that my device is loaded and I just need to load the module. Or do I need to compile asterisk again?? Any help would be highly appreciated. My asterisk version is Asterisk 1.4.19.2 and I am on a Fedora release 9 server.
Re: [asterisk-users] [asterik-users] Installing PRI card
Hello NaJIm, Your zaptel.conf and zapata.conf files must match as to what channels and signaling are in use. See the examples at voip-info: http://www.voip-info.org/wiki/view/Asterisk+config+zaptel.conf On 9/29/2011 4:21 PM, NaJIm wrote: Hi Eric, This is the error messages I get I try to load the module. *CLI module load chan_zap.so [Sep 30 04:45:57] WARNING[5182]: pbx.c:2979 ast_register_application: Already have an application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, ISDN PRI signalling -- Registered channel 2, ISDN PRI signalling -- Registered channel 3, ISDN PRI signalling -- Registered channel 4, ISDN PRI signalling -- Registered channel 5, ISDN PRI signalling -- Registered channel 6, ISDN PRI signalling -- Registered channel 7, ISDN PRI signalling -- Registered channel 8, ISDN PRI signalling -- Registered channel 9, ISDN PRI signalling -- Registered channel 10, ISDN PRI signalling -- Registered channel 11, ISDN PRI signalling -- Registered channel 12, ISDN PRI signalling -- Registered channel 13, ISDN PRI signalling -- Registered channel 14, ISDN PRI signalling -- Registered channel 15, ISDN PRI signalling -- Registered channel 17, ISDN PRI signalling -- Registered channel 18, ISDN PRI signalling -- Registered channel 19, ISDN PRI signalling -- Registered channel 20, ISDN PRI signalling -- Registered channel 21, ISDN PRI signalling -- Registered channel 22, ISDN PRI signalling -- Registered channel 23, ISDN PRI signalling [Sep 30 04:45:57] WARNING[5182]: chan_zap.c:905 zt_open: Unable to specify channel 24: Device or resource busy [Sep 30 04:45:57] ERROR[5182]: chan_zap.c:7219 mkintf: Unable to open channel 24: Device or resource busy here = 0, tmp-channel = 24, channel = 24 [Sep 30 04:45:57] ERROR[5182]: chan_zap.c:10582 build_channels: Unable to register channel '17-31' Regards, Najim On Fri, Sep 30, 2011 at 4:24 AM, Eric Wieling ewiel...@nyigc.com wrote: Try module load chan_zap.so in the CLI. You should see whatever errors are generated. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of NaJIm Sent: Thursday, September 29, 2011 5:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] [asterik-users] Installing PRI card Hi, We have got a new PRI card at one of our Office locations and now I need to install the the device on a remote server. Is there any way to know if the device is loaded already. When I give cat /proc/zaptel/* it returns the following. # cat /proc/zaptel/* Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) B8ZS/ESF RED IRQ misses: 2 1 WCT1/0/1 Clear (In use) RED 2 WCT1/0/2 Clear (In use) RED 3 WCT1/0/3 Clear (In use) RED 4 WCT1/0/4 Clear (In use) RED 5 WCT1/0/5 Clear (In use) RED 6 WCT1/0/6 Clear (In use) RED 7 WCT1/0/7 Clear (In use) RED 8 WCT1/0/8 Clear (In use) RED 9 WCT1/0/9 Clear (In use) RED 10 WCT1/0/10 Clear (In use) RED 11 WCT1/0/11 Clear (In use) RED 12 WCT1/0/12 Clear (In use) RED 13 WCT1/0/13 Clear (In use) RED 14 WCT1/0/14 Clear (In use) RED 15 WCT1/0/15 Clear (In use) RED 16 WCT1/0/16 Clear RED 17 WCT1/0/17 Clear (In use) RED 18 WCT1/0/18 Clear (In use) RED 19 WCT1/0/19 Clear (In use) RED 20 WCT1/0/20 Clear (In use) RED 21 WCT1/0/21 Clear (In use) RED 22 WCT1/0/22 Clear (In use) RED 23 WCT1/0/23 Clear (In use) RED 24 WCT1/0/24 HDLCFCS (In use) RED But when I connect to the console, I am unable to give any ZAP related commands. Does this mean that my device is loaded and I just need to load the module. Or do I need to compile asterisk again?? Any help would be highly appreciated. My asterisk version is Asterisk 1.4.19.2 and I am on a Fedora release 9 server. Thanks, Najim -- _ -- Bandwidth and Colocation Provided by
[asterisk-users] OUTBOUND and INBOUND routes
Hello All, I have a pstn line can have the local, STD and ISD capabilities. My local number is 91471-2527XXX and the region is India. I would like to use the number for all possible calls ( local, STD and ISD call facilities to Land line and mobile phones) through an FXO card configured in asterisk freepbx. Can anybody help me to create an outbound route and inbound route required in freepbx for the above requirement ? Thanks, Michael.k -- _ -- 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] Asynchronous AGI Problems (Asterisk 1.8.7.0), ubuntu-server
On Sun, Sep 25, 2011 at 8:26 AM, Mehmet Avcioglu meh...@activecom.net wrote: Actually it doesn't say AGI(async:script) it says AGI(async:agi) and than continues further to setting up an AMI user so the script is executed through the manager interface?? Than it says AGI(agi:async).?? Well most importantly it says Cons of async AGI: It is the most complex method of using AGI to implement. ..:) I have been interested in Async AGI as well and after reading your post looked into the link you provided, seems different than what we immediately think, a background process. Perhaps just start the script normally AGI(script.sh) and than inside it run your background process background-script.sh /dev/null 21 /dev/null or fork a new process, detach, run in background, etc... Hopefully somebody else can point us towards the right direction in setting up a real asterisk asynchronous AGI application. Despite being some shameless self-promotion, I want to point out this post I wrote several years ago explaining the basics: http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ Moises Silva Senior Software Engineer, Software Development Manager Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- 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] PSTN connectivity
Hey Warren I thought that these are the complete CLI logs for one call. It started like == Using SIP RTP CoS mark 5 and from-internal priority-1 ..So that seemed legit to me. Yeah I too suspect that dialing rules are not being matched and thats why Gotoif's are failing. On Thu, Sep 29, 2011 at 8:15 PM, Warren Selby wcse...@selbytech.com wrote: On Thu, Sep 29, 2011 at 7:51 AM, michael k mich...@inapp.com wrote: Thanks for the update. but how do i resolve this issue ? can you help me please ? You didn't provide a full CLI trace of the outgoing call, you only supplied the hangup portion of the call. Please try again. Also, what are the dialing rules like in your country? You only have outbound dial patterns setup to handle North American numbers (8+ NXXNXX or 8+ NXX). The Dial Pattern box in the Outbound Rules box is where you define what numbers you want to go out over this trunk. If you dial a number that doesn't match one of these patterns, FreePBX is going to look internally for a dial pattern to match against, and if it doesn't find one there, it will end the call. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- 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 -- _ -- 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] record calls of specific agnets
I guess that was this variable like SPYGROUP which needs to be set for specific extensions and then ask Chanspy to spy on that group. !! On Thu, Sep 29, 2011 at 9:37 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Thursday 29 September 2011, Lyle McKarns wrote: Hello Asterisk List! I have been asked to record calls from specific agents, and I am having difficulty finding if this is possible, and if so, how exactly to do it. Have a recorded context and an unrecorded context in your dialplan, identical save for the lines that start the recording and cleanup processes being absent from the latter. Then set contexts per extension in sip.conf. -- AJS Answers come *after* questions. -- _ -- 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 -- _ -- 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] Features not working
Hey, Whats the output of command features show ? on CLI ? On Fri, Sep 30, 2011 at 1:51 AM, Mike Diehl mdi...@diehlnet.com wrote: Hi all. I could have sworn this working at one time... But it doesn't look like any of the functions provided by features.so is working for me. (one-touch monitoring, attended/blind transfer, etc) I've (re)loaded features.so, as well as bridge_builtin_features.so. The config file looks sane. What else should I try? TIA, -- Take care and have fun, Mike Diehl. -- _ -- 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 -- _ -- 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] Features not working
The output of that command is sane. I restarted Asterisk and things seem OK, now. Not sure what happened, but I don't have time to ponder. Thank you for your time, though. Mike. On Thursday 29 September 2011 11:17:24 pm Sam Govind wrote: Hey, Whats the output of command features show ? on CLI ? On Fri, Sep 30, 2011 at 1:51 AM, Mike Diehl mdi...@diehlnet.com wrote: Hi all. I could have sworn this working at one time... But it doesn't look like any of the functions provided by features.so is working for me. (one-touch monitoring, attended/blind transfer, etc) I've (re)loaded features.so, as well as bridge_builtin_features.so. The config file looks sane. What else should I try? TIA, -- Take care and have fun, Mike Diehl. -- _ -- 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 -- Take care and have fun, Mike Diehl. -- _ -- 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
