Re: [asterisk-users] Polycom VLAN
This is the whole point behind using VLAN on the phone. Tagged VLAN for your phone with QoS configured accordingly on your switch and untagged VLAN for your PC, both on the same wire. This way you can always guarantee enough bandwidth for your VoIP packets. Thanks, Wojtek On 2-Jan-08, at 1:04 PM, Alex Balashov wrote: > > On Wed, 2 Jan 2008, Jeremy Mann wrote: > >> Just curious, if I have my Polycom IP 550 phone VLAN tag 30, will the >> packets I send from my PC(on the PC port of the phone) have the same >> VLAN tag? THe PC is sending untagged packets. > > According to this -- > > http://www.polycom.com/common/documents/whitepapers/ > vlans_and_polycom_soundpoint_ip_desktop_ip_telephones.pdf > > "If a PC is connected to the phone, all packets generated by the > PC will >be passed through unmodified, regardless of the presence of an > 802.1q/p >tag or its contents. Since PCs do not typically tag frames, this > means >they will be on the native VLAN. " > > Cheers, > > -- > Alex Balashov > Evariste Systems > Web: http://www.evaristesys.com/ > Tel: +1-678-954-0670 > Direct : +1-678-954-0671 > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone using VoIP WiFi phones?
The only advantage is when you travel. Last year I took my wifi sip phone to Astricon in Madrid and everything worked as expected. I am just packing it and heading for Paris... Wojtek -Original Message- From: Mojo with Horan & Company, LLC [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 20, 2006 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone using VoIP WiFi phones? I have been pretty happy with my cisco 7920, but it has been by the wayside for six months or more now due to the wimpy battery life. I recommend a standard cordless phone (yes, even 2.4ghz) and ATA to beat the wifi voip phones I've tried :( Warren wrote: > If anyone out there using VoIP WiFi phones? If so, which ones and what > do you think about it? > > Thanks, > W > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Mojo <[EMAIL PROTECTED]> Office Manger, Horan & Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to connect to Asterisk? (simple[?] question)
Telnet uses TCP, Asterisk's SIP is UDP Wojtek - Original Message - From: "John Klimek" <[EMAIL PROTECTED]> To: Sent: Monday, June 12, 2006 10:42 PM Subject: [Asterisk-Users] Unable to connect to Asterisk? (simple[?] question) I'm trying to setup Asterisk on my Linksys WRT54G router and it appears to startup successfully (no errors) and it says it is listening on 0.0.0.0 port 5060, but I am unable to connect to it. I've tried "telnet localhost 5060" but it just says connection refused. I've also tried connecting from another machine on my network (eg. telnet 192.168.0.1 5060) but it also says connection refused. Finally, I've tried changing the bound address in sip.conf to "127.0.0.1" and "192.168.0.1" but I am still unable to connect using all the methods mentioned above. What else can be the problem? Can I have some sort of iptables problem? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + PRI Card -> Nortel BCM
Asterisk with Digium's single span PRI works just fine with BCM. Contact me off the list if you need details. Thanks, Wojtek - Original Message - From: "Mr. Jones" <[EMAIL PROTECTED]> To: Sent: Saturday, June 03, 2006 1:08 PM Subject: [Asterisk-Users] Asterisk + PRI Card -> Nortel BCM Has anyone fed a Nortel BCM from Asterisk? I'm interested in switching our company over, but don't want to replace all the handsets in one fell swoop. I imagine some of the PRI cards can "emulate" a switch? I'd still like to pass CallerID into the Nortel, etc but all the external traffic would be VOIP, not TDM. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom - directory dial
In your dialplan , inside context which Polycom is using check if the dialed number begins with 9 (or check length of dialing string) and prepend it with 9 if necessary. Thanks, Wojtek From: Bill Gibbs [mailto:[EMAIL PROTECTED] Sent: Saturday, March 11, 2006 6:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom - directory dial This is not an Asterisk specific question but doesn’t anyone know if you can automatically prepend a 9 on the call lists so clients can return dial without having to repunch in the number? If you go to directories now it just shows the number without a 9 (obviously). Maybe on the Asterisk side?? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT - Cisco IP Phone and PC in different VLANs(with802.1x)
Switch is only tagging the vlan packets. Once the PC loads the vlan
aware driver ("client") it will be able to read tagged packet for the
vlan which PC has been configured to use. Nothing to be done on the
switch.
W
-Original Message-
From: Joao Pereira [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 02, 2006 1:15 PM
To: Wojciech Tryc; asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent
VLANs(with802.1x)
Ok, but the PC has an 802.1x client that configures the VLAN when he
authenticates.
Is this going to pass through the phone?
And will the switch accept it?
Thanks
Joao Pereira
Wojciech Tryc wrote:
> Your pc has to able to support tagged vlans. The switch on the phone
> will pass through both tagged and untagged vlans.
> W
> - Original Message - From: "Joao Pereira"
<[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Thursday, March 02, 2006 11:51 AM
> Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent
> VLANs(with 802.1x)
>
>
>> And about the 802.1x ?
>> The phones can work as passthrough and force the PC to use 802.1x ?
>> What configuration do we put in the switches? Do we put the switch as
>> "access" (with 802.1x) or "trunk" (without 802.1x) ?
>>
>> Thanks
>> Joao Pereira
>>
>>
>>
>> Greg Oliver wrote:
>>
>>> It actually depends on the switch model. Some put the port into
>>> trunking mode automatically with the sw voi command, and some do
not.
>>>
>>> Hopefully one day Cisco will finally make their own products and
become
>>> uniform instead of buying several companies and glue'ing them all
>>> together to get an ethernet switch that works. At least they got
the
>>> routers right :)
>>>
>>> On Thu, 2006-03-02 at 08:13 -0800, Gary Richardson wrote:
>>>
>>>> You don't need switchport mode trunk when using switchport voice
>>>> vlan..
>>>> On 3/1/06, Nicholas Kathmann
>>>> <[EMAIL PROTECTED]> wrote:
>>>>Joao Pereira wrote:
>>>>> Hello to all > I would like to know If some of you have
>>>> already configured
>>>>an Cisco
>>>>> IP Phone (7940 or 7960) to work in a different VLAN than
the
>>>>PC that
>>>>> is connected through the phone switch?
>>>>> I know that this can be done with the Skinny firmware, but
I
>>>>dont if > it works with the SIP firmware.
>>>>>
>>>>> The Cisco technical staff told me that these phones dont
>>>>support
>>>>> 802.1x but can work as pass-through. This way I can still
>>>>use the PCs
>>>>> with 802.1x and the phones in the same Ethernet plug. >
>>>>> Did someone made it with the Cisco IP phones? What
>>>>configuration do I
>>>>> need in the phones and in the switch?
>>>>> Thanks
>>>>> Joao Pereira
>>>>>
>>>>If configuring with Cisco switches, I'm pretty sure they
pull
>>>>the information for which VLAN to operate in from the
>>>> switch. You
>>>>have to
>>>>configure the switchports on the Cisco switch like so:
>>>>interface fastethernet 0/1
>>>> switchport trunk native vlan switchport
>>>> mode trunk
>>>> switchport voice vlan
>>>> spanning-tree portfast trunk
>>>>etc.
>>>>Thanks,
>>>>Nicholas Kathmann, CISSP
>>>>Kathmann Consulting, LLC
>>>>___ --Bandwidth
>>>> and Colocation provided by Easynews.com --
>>>>Asterisk-Users mailing list
>>>>To UNSUBSCRIBE or update options visit:
>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>> ___
>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>
>>>> Asterisk-Users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>> ___
>>> --Bandwidth and Colocation provided by Easynews.com --
>>>
>>> Asterisk-Users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>> ___
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> Asterisk-Users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs(with 802.1x)
Your pc has to able to support tagged vlans. The switch on the phone will pass through both tagged and untagged vlans. W - Original Message - From: "Joao Pereira" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, March 02, 2006 11:51 AM Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs(with 802.1x) And about the 802.1x ? The phones can work as passthrough and force the PC to use 802.1x ? What configuration do we put in the switches? Do we put the switch as "access" (with 802.1x) or "trunk" (without 802.1x) ? Thanks Joao Pereira Greg Oliver wrote: It actually depends on the switch model. Some put the port into trunking mode automatically with the sw voi command, and some do not. Hopefully one day Cisco will finally make their own products and become uniform instead of buying several companies and glue'ing them all together to get an ethernet switch that works. At least they got the routers right :) On Thu, 2006-03-02 at 08:13 -0800, Gary Richardson wrote: You don't need switchport mode trunk when using switchport voice vlan.. On 3/1/06, Nicholas Kathmann <[EMAIL PROTECTED]> wrote: Joao Pereira wrote: > Hello to all > I would like to know If some of you have already configured an Cisco > IP Phone (7940 or 7960) to work in a different VLAN than the PC that > is connected through the phone switch? > I know that this can be done with the Skinny firmware, but I dont if > it works with the SIP firmware. > > The Cisco technical staff told me that these phones dont support > 802.1x but can work as pass-through. This way I can still use the PCs > with 802.1x and the phones in the same Ethernet plug. > > Did someone made it with the Cisco IP phones? What configuration do I > need in the phones and in the switch? > Thanks > Joao Pereira > If configuring with Cisco switches, I'm pretty sure they pull the information for which VLAN to operate in from the switch. You have to configure the switchports on the Cisco switch like so: interface fastethernet 0/1 switchport trunk native vlan switchport mode trunk switchport voice vlan spanning-tree portfast trunk etc. Thanks, Nicholas Kathmann, CISSP Kathmann Consulting, LLC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] auto provision of IP501 polycom
This is dhcpd.conf which works great with
Polycom 501:
option domain-name "blah.com";
option domain-name-servers 192.168.80.3;
default-lease-time 7200;
max-lease-time 14400;
authoritative;
# ad-hoc DNS update scheme - set to
"none" to disable dynamic DNS updates.
ddns-update-style none;
log-facility local7;
# A slightly different configuration for
an internal subnet.
subnet 192.168.80.0 netmask 255.255.255.0
{
range 192.168.80.20
192.168.80.199;
option domain-name-servers
192.168.80.3;
option domain-name "blah.com";
option routers 192.168.80.1;
option tftp-server-name
"192.168.80.3";
option broadcast-address
192.168.80.255;
option ntp-servers 192.168.80.3;
option time-offset -18000;
default-lease-time 7200;
max-lease-time 144
Wojtek
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
Sent: Thursday, February 23, 2006
10:17 AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] auto
provision of IP501 polycom
This has worked for several months for us.
It's /etc/dhcpd.conf
ddns-update-style ad-hoc;
authoritative;
option option-66 code 66 = string;
subnet 172.32.16.0 netmask 255.255.255.0 {
#range 192.168.10.101
192.168.10.120;
default-lease-time 600;
max-lease-time 7200;
option
option-66
"boot.voip.com";
option
domain-name-servers 172.32.16.16, 172.32.16.16;
option
time-servers
clock.voip.com;
option
domain-name
"uac.bil.voip.com";
option
time-offset
-25200;
host uap301-1 {
hardware ethernet 00:04:f2:02:1b:b8;
fixed-address 172.32.16.128;
}
host uap301-2 {
hardware ethernet 00:04:f2:02:96:23;
fixed-address 172.32.16.129;
}
host uap601-1 {
hardware ethernet 00:04:f2:02:3d:42;
fixed-address 172.32.16.130;
}
host uap601-2 {
hardware ethernet 00:04:f2:02:86:47;
fixed-address 172.32.16.131;
}
host uap501-1 {
hardware ethernet 00:04:f2:02:f4:0e;
fixed-address 172.31.16.132;
}
host uap501-2 {
hardware ethernet 00:04:f2:02:2d:00;
fixed-address 172.32.16.133;
}
host uap501-3 {
hardware ethernet 00:04:f2:03:6a:c7;
fixed-address 172.32.16.134;
}
}
-Original Message-
From: Wojciech Tryc
[mailto:[EMAIL PROTECTED]
Sent: Thursday, February 23, 2006
6:15 AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] auto
provision of IP501 polycom
Damon,
I have no problem provisioning 501s
through tftp. The tftp address is distributed via dhcp.
Thx,
Wojtek
From: Damon Estep
[mailto:[EMAIL PROTECTED]
Sent: Thursday, February 23, 2006
8:09 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] auto
provision of IP501 polycom
Has anyone been able to get the IP501 to discover the FTP
server IP address (via dhcp or dns) and download 100% of the config from a
provisioning server?
We are still having to touch each unit to enter the ftp
server address and password, as well as set many of the options that will not
take from the config file.
Have a sample config file you are willing to share?
What is required in the way of dhcp options or dns entries
to get the polycom to discover the ftp boot server?
What about changing default passwords via ftp?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] auto provision of IP501 polycom
Damon, I have no problem provisioning 501s through tftp. The tftp address is distributed via dhcp. Thx, Wojtek From: Damon Estep [mailto:[EMAIL PROTECTED] Sent: Thursday, February 23, 2006 8:09 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] auto provision of IP501 polycom Has anyone been able to get the IP501 to discover the FTP server IP address (via dhcp or dns) and download 100% of the config from a provisioning server? We are still having to touch each unit to enter the ftp server address and password, as well as set many of the options that will not take from the config file. Have a sample config file you are willing to share? What is required in the way of dhcp options or dns entries to get the polycom to discover the ftp boot server? What about changing default passwords via ftp? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A unique 'click to call' project - Could usesomeadvice
You could do something like :
[router-local]
exten => _613XXX,1,Goto(trunklocal,
${EXTEN:${TRUNKMSD3}},1)
exten => _613XXX,2,Congestion
[router-ld]
exten => _1NX,1,Goto(trunkld,91${EXTEN},1)
exten => _1NX,2,Congestion
[trunklocal]
exten =>
XXX,1,Dial(Zap/g1/${EXTEN}|20)
exten => XXX,2,Congestion
[router-agents]
include => router-local
include => router-ld
include => trunklocal
[agents]
exten => s,1,Dial(Local/[EMAIL PROTECTED])
In your call file specify “agents”
as your context to call agents through PSTN
Thanks,
W
From: Aloi,
Christopher [mailto:[EMAIL PROTECTED]
Sent: Friday, February 17, 2006
11:56 AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] A
unique 'click to call' project - Could usesomeadvice
Colin,
Thanks for your assistance.
Reading over your advice I seem to still
be a bit confused.
My agents are not on the Asterisk server;
it appears in your advice that my the call will travel this path:
WWW interface --> agent enters their
DID, platform to use, and termination DID --> AST calls agent --> Agent
calls termination DID
If my agents are not on the Asterisk
server (believe me, I wish there were) :) how will this work?
I need a way to pass both the desired
termination DID and the origination DID.
Maybe I missed something
Thanks,
-- -- --
Christopher T. Aloi
USA Datanet - Technical Support Engineer
318 South Clinton Street
Syracuse, NY 13202
C: (315) 569 4033
O: (315) 579 7074
E: [EMAIL PROTECTED]
-- -- --
From: Colin
Anderson [mailto:[EMAIL PROTECTED]
Sent: Friday, February 17, 2006
10:42 AM
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] A
unique 'click to call' project - Could usesome advice
You create a context in your dialplan that
accepts the DID to call as a variable using the SetVar: syntax in your .call
file. You then set up the context to call your agent, and when they pick up,
the context takes the variable you set in your .call file as the
dialstring argument for a subsequent Dial(). Once the DID picks up, the calls
are bridged together. Whatever web scripting language you use writes the .call
file, and you use POSTed arguments or querystrings:
http://foo.com/call?context=MyContext&Agent=SIP/&DID=1551212
You can see this in action at www.landmarkhomes.ca - click on any of
the pretty buttons that say "Call us now"
However, I have noticed that * 1.2.x will
not wait for the caller to pick up before executing the rest of the directives
in the context - it keeps executing regardless of the calling party's
pickup status. Using * 1.0.x the context will wait for the caller to pick up
before placing the call to the callee (i.e. executing the rest of the
directives in the context)
.call file (shortened to relevant)
Channel: SIP/
(if you are using SIP phones)
SetVar: DID=XXX
Context: MyContext
[MyContext]
exten => s,1,Dial(ZAP/g0/${DID})
hth
-Original Message-
From: Aloi, Christopher
[mailto:[EMAIL PROTECTED]
Sent: Friday, February 17, 2006
8:07 AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] A unique
'click to call' project - Could use some advice
Hello List,
I work for an IP communication provider in
upstate NY as the engineer assisting our technical support team.
We provide a number of different Telco
systems to residential subscribers; and in an effort to more effectively
trouble shoot termination problems I came up with the idea of creating a click
to call system that will allow our agents to effortlessly place test calls.
On a daily basis we place numerous
(50-100) 'test' calls to various locations in the US; these 'test' calls are routed
using one of three different phone systems:
1) The PSTN
2) Broadband phone platform one
3) Broadband phone platform two
I have an Asterisk server configured that
can terminate out three platforms listed above.
Our support agents are behind a
Televantage ACD using D-TermSeries E NEC phones.
Each agent has a DID and are permitted to
receive inbound calls on that DID.
Here is my goal:
Create a web application that will allow
the agent to enter the following information into a form:
1) The agents DID
2) The platform the agent wishes to
terminate a test call through (either 1,2,3 above)
3) The number the agent wishes to
terminate to
My thought is this form will generate a
.call file in /var/spool/asterisk/outgoing that will then ring the agents
station, pause, and terminate to the selected DID using the selected
platform. I also thought about interacting directly with the AGI.
I can successfully generate the .call
files, and ring a station on the Asterisk se
RE: [Asterisk-Users] A unique 'click to call' project - Could use someadvice
Why don’t you use Local and router functionality to find a route to PSTN based agents? W From: Aloi, Christopher [mailto:[EMAIL PROTECTED] Sent: Friday, February 17, 2006 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] A unique 'click to call' project - Could use someadvice Hello List, I work for an IP communication provider in upstate NY as the engineer assisting our technical support team. We provide a number of different Telco systems to residential subscribers; and in an effort to more effectively trouble shoot termination problems I came up with the idea of creating a click to call system that will allow our agents to effortlessly place test calls. On a daily basis we place numerous (50-100) 'test' calls to various locations in the US; these 'test' calls are routed using one of three different phone systems: 1) The PSTN 2) Broadband phone platform one 3) Broadband phone platform two I have an Asterisk server configured that can terminate out three platforms listed above. Our support agents are behind a Televantage ACD using D-TermSeries E NEC phones. Each agent has a DID and are permitted to receive inbound calls on that DID. Here is my goal: Create a web application that will allow the agent to enter the following information into a form: 1) The agents DID 2) The platform the agent wishes to terminate a test call through (either 1,2,3 above) 3) The number the agent wishes to terminate to My thought is this form will generate a .call file in /var/spool/asterisk/outgoing that will then ring the agents station, pause, and terminate to the selected DID using the selected platform. I also thought about interacting directly with the AGI. I can successfully generate the .call files, and ring a station on the Asterisk server - the problem is the agents are not on the Asterisk server. Is there a way to use Asterisk to initiate these test calls? Is it possible to create a forwarding context to handle this? Any thoughts? Thanks for the help! Cheers, -- -- -- Christopher T. Aloi USA Datanet - Technical Support Engineer 318 South Clinton Street Syracuse, NY 13202 C: (315) 569 4033 O: (315) 579 7074 E: [EMAIL PROTECTED] -- -- -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PIKA Technologies Inc. Announces Support for Open Source Asterisk PBX
Ottawa, Canada – February 15, 2006 - PIKA Technologies Inc. today announced that they have integrated PIKA’s high-density analog computer plug-in boards with the open source Asterisk PBX, with the introduction of PIKA Connect for Asterisk. PIKA Connect for Asterisk is a software layer, available free of charge and distributed under the GNU Public License (GPL), which allows interoperability between PIKA high-density analog boards (Daytona MM) and Asterisk PBX software. “The Asterisk development community can now benefit from advanced features for fax and echo cancellation in high density analog applications, made possible by PIKA’s DSP processing power on the board,” stated Wojciech Tryc, Enterprise VoIP architect at PIKA Technologies. “Because of the native bridging for TDM calls, latency is drastically reduced—nearly eliminated— in this implementation. The solution is very reliable, as we have witnessed not only in the lab, but in live customer environments.” Stroudwater Contact Point, LLC, based in Portland Maine, provides software development services, applications and infrastructure for contact centers. "We have chosen PIKA's Daytona MM analog hardware for analog support in their upcoming Asterisk-based Dirigo iQueue™ PBX/ACD product because of the scalability and density of 24 ports for either loop start or POTS. On board switching of calls and on board echo cancellation make the product more efficient and ease demands on the server's CPU. Further, PIKA support throughout our development has been outstanding," said Bill Hunt, President and CTO at Stroudwater Contact Point. "Our next step is to integrate the PIKA on board fax solution." Unlimitel Inc. offers VoIP services to business customers across Canada using the VoIP/PSTN network and was looking for a way to deliver reliable fax. Stephan Monette, President of Unlimitel stated: “Using Asterisk and the PIKA [high density analog] Daytona board was quick and easy. We were able to demonstrate the stability of the fax service in our lab within 24 hours!” Kanatek Technologies Inc. is an Ottawa, Ontario-based systems integrator delivering IT consulting and support services to a variety of companies in North America. Paul Labelle, Vice President of Operations at Kanatek said, “We were impressed with the flexibility and customization that the PIKA Connect for Asterisk solution could provide. We were able to integrate it with our current PBX system, and extend our communications network to nearly 100 users at multiple locations including the corporate office, branch office and remote locations.” Asterisk developers can be up and running quickly with PIKA Connect for Asterisk and PIKA hardware. “For those familiar with using the Asterisk platform, no additional training is required. They can take advantage of the PIKA solution with minimal effort or investment,” said PIKA Technologies’ Wojciech Tryc. For more information on PIKA Connect for Asterisk, go to http://www.pikatechnologies.com/products/asterisk.htm About PIKA PIKA Technologies designs and manufactures computer plug in voice cards and software that connect a computer system to both TDM- and IP-based networks to provide advanced voice services. For almost two decades PIKA Technologies has been serving companies around the world that require voice cards to design sophisticated phone services for recording systems, voice services applications, and PC-PBX systems. The company has built a reputation for delivering innovative products and exceptional technical support by working closely with its customers. Headquartered in Ottawa, ON, Canada, the company has ranked in The Branham300, an authoritative ranking of successful Canadian high tech firms, for three consecutive years. Visit www.pikatechnologies.com or call +1-613-591-1555 for more information. © PIKA Technologies Inc., 2005. PIKA is a registered trademark of PIKA Technologies Inc. For more information, please contact: Miriam Rautiainen Head of Marketing | PIKA Technologies Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail Problem
You are looking for vn-goodbye, most likely under
sounds/vm
W
- Original Message -
From:
Sam Lee
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February 09, 2006 9:21
PM
Subject: RE: [Asterisk-Users] Voicemail
Problem
Strange thing that , its there !
[EMAIL PROTECTED]:/home/sam# ls
/var/lib/asterisk/sounds/goodbye.gsm/var/lib/asterisk/sounds/goodbye.gsm
[EMAIL PROTECTED]:/home/sam#
That's why i found it very strange. Thanks for replying.
Are there any other ideas ?
Regards,Sam
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wojciech
TrycSent: Friday, February 10, 2006 9:59 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Voicemail Problem
You don't have 'vm-goodbye' voice file. Check
under /var/lib/asterisk/sounds
Wojtek
- Original Message -
From:
Sam Lee
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February 09, 2006 8:38
PM
Subject: RE: [Asterisk-Users] Voicemail
Problem
Hey guys,
Any hint at all ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam
LeeSent: Thursday, February 09, 2006 3:30 PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Voicemail Problem
I have just
setup my OPENSER to work with the asterisk 1.2.2.
I've set
extension 400 in extension.conf to point to the VoicemailMain()
application
The entire
program works fine, but there seems to be some problem whenever the call is
hangup, either by pushing # to exit the VoicemailMain() apps or by hanging
the phone. If the # button is push, should Asterisk send something back to
tell OPENSER to hang up the party ?
Here's the log
of verbose level 3
Asterisk*CLI>
-- Playing 'vm-youhave'
(language 'en') -- Playing 'vm-no' (language
'en') -- Playing 'vm-messages' (language
'en') -- Playing 'vm-opts' (language
'en') -- Playing 'vm-goodbye' (language
'en') -- Executing
Playback("SIP/210.23.1.139-081ee3d8", "Goodbye") in new stackFeb 9 15:05:06 WARNING[23242]: file.c:509
ast_openstream_full: File Goodbye does not exist in any formatFeb
9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile: Unable to open Goodbye
(format alaw): No such file or directoryFeb 9 15:05:06
WARNING[23242]: app_playback.c:132 playback_exec: ast_streamfile failed on
SIP/203.125.68.66-081ee3d8for Goodbye -- Executing
Hangup("SIP/203.125.68.66-081ee3d8", "") in new stack == Spawn
extension (default, 400, 3) exited non-zero on
'SIP/203.125.68.66-081ee3d8'
Asterisk*CLI>
Any idea what is
this all about ?
Regards,Sam
___--Bandwidth and
Colocation provided by Easynews.com --Asterisk-Users mailing
listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___--Bandwidth and
Colocation provided by Easynews.com --Asterisk-Users mailing
listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail Problem
You don't have 'vm-goodbye' voice file. Check under
/var/lib/asterisk/sounds
Wojtek
- Original Message -
From:
Sam Lee
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February 09, 2006 8:38
PM
Subject: RE: [Asterisk-Users] Voicemail
Problem
Hey guys,
Any hint at all ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam
LeeSent: Thursday, February 09, 2006 3:30 PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Voicemail Problem
I have just setup
my OPENSER to work with the asterisk 1.2.2.
I've set extension
400 in extension.conf to point to the VoicemailMain()
application
The entire program
works fine, but there seems to be some problem whenever the call is hangup,
either by pushing # to exit the VoicemailMain() apps or by hanging the phone.
If the # button is push, should Asterisk send something back to tell OPENSER
to hang up the party ?
Here's the log of
verbose level 3
Asterisk*CLI>
-- Playing 'vm-youhave' (language 'en') -- Playing
'vm-no' (language 'en') -- Playing 'vm-messages'
(language 'en') -- Playing 'vm-opts' (language
'en') -- Playing 'vm-goodbye' (language
'en') -- Executing Playback("SIP/210.23.1.139-081ee3d8",
"Goodbye") in new stackFeb 9 15:05:06
WARNING[23242]: file.c:509 ast_openstream_full: File Goodbye does not exist in
any formatFeb 9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile:
Unable to open Goodbye (format alaw): No such file or
directoryFeb 9 15:05:06 WARNING[23242]: app_playback.c:132
playback_exec: ast_streamfile failed on SIP/203.125.68.66-081ee3d8for
Goodbye -- Executing
Hangup("SIP/203.125.68.66-081ee3d8", "") in new stack == Spawn
extension (default, 400, 3) exited non-zero on
'SIP/203.125.68.66-081ee3d8'
Asterisk*CLI>
Any idea what is
this all about ?
Regards,Sam
___--Bandwidth and
Colocation provided by Easynews.com --Asterisk-Users mailing
listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Work in Ukraine
Hi All, My friend is looking to for a fulltimer to work on various VoIP gateways (mostly Cisco) and few Asterisk servers. Some development skills and knowledge of Asterisk's API would be an asset. At least familiarity with Asterisk's AGI and Perl/C/PHP would help. He is located in Kiev, so obviously Kiev residents are preffered. Remote work to be discussed. Again, this is not a contract, this is Full-Time job. Please send your resumes to [EMAIL PROTECTED] Regards, Wojtek ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adding Cepstral to Asterisk
I am not following... Why would you need to integrate Cepstral directly into Asterisk? Just to be able to call it as Asterisk app from your dialplan? I am running Cepstral and calling it through the System call. Thanks, Wojtek - Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, October 03, 2005 11:27 AM Subject: Re: [Asterisk-Users] Adding Cepstral to Asterisk Then did you do a make clean / make / make install? Then do "show applications" at the CLI prompt after you have restarted asterisk. "service asterisk stop" "service asterisk start" ... I downloaded Cepstral to my Asterisk Box. I did the install and let it install to /opt/swift. I brought down a new CVS-HEAD as of today 10/1. I added APPS+=app_cepstral.so into the Makefile in /usr/src/asterisk/apps/Makefile Like: # Obsolete things... # #APPS+=app_sql_postgres.so #APPS+=app_sql_odbc.so APPS+=app_cepstral.so # I did this piece but wasn't sure exactly what part of the Makefile I was to add it in so I added it in here: Towards the top of the file where it talks obsolete programs are commented out. And then after the section that compiles voicemail add: app_cepstral.so: app_cepstral.c $(CC) -D_GNU_SOURCE -shared -Xlinker -x -o $@ $< -lz -lm -lswift -lceplex_us -lceplang_en -lz -ldl -L/opt/swift/lib -I/opt/swift/include Make sure the $(CC) line starts with a tab, not spaces. I didn't see a lot about voicemail: app_sql_odbc.so: app_sql_odbc.o $(CC) $(SOLINK) -o $@ $< -lodbc app_cepstral.so: app_cepstral.c $(CC) -D_GNU_SOURCE -shared -Xlinker -x -o $@ $< -lz -lm -lswift -lceplex_us -lceplang_en -lz -ldl -L/opt/swift/lib -I/opt/swift/include look: look.c $(CC) -pipe -O6 -g look.c -o look -lncurses I checked the /etc/ld.so.conf file for a line like: opt/swift/lib in the file. It wasn't there so I added it: include ld.so.conf.d/*.conf /opt/swift/lib I ran ldconfig when I was done. I can't see that Cepstral was added into Asterisk and I was wondering what I have done wrong that it doesn't work. Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enabling stun on asterisk?
Asterisk doesn't have a STUN client. Most likely you wany to configure your clients (phones) which are behind non-symmetrical NATs to use STUN while connecting to Asterisk acting as your proxy. Thanks, Wojtek - Original Message - From: hank To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, June 28, 2005 2:34 PM Subject: [Asterisk-Users] enabling stun on asterisk? hello I am going to be setting up a stun server on windows how do I enable it to work withasterisk? thanks hank ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Viva Madrid!
Agreed, I will post some pics early next week:) Wojtek - Original Message - From: "Nicolás Gudiño" <[EMAIL PROTECTED]> To: Sent: Thursday, June 16, 2005 8:27 PM Subject: [Asterisk-Users] Viva Madrid! enough said -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No Hope with Fax and X100P?
Chris, I am using the latest SpanDSP (but also tried with the old one) with X100P without any major problems. Send me your zapata config off line and I will try to help you. Also, what version of SpanDSP/Asterisk do you have? W - Original Message - From: "Chris" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, April 15, 2005 11:27 AM Subject: [Asterisk-Users] No Hope with Fax and X100P? I've read a lot on this board and in the WIKI. Is there no hope of a X100P with SpanDSP accepting incoming fax? Everytime I try it fails to train. Is there something I have missed that could fix the problem? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] failover outbound dialplan
You can use ChanIsAvail to confirm that specific
trunk is available before routing your call.
Wojtek
- Original Message -
From:
Jason Brown
To: asterisk-users@lists.digium.com
Sent: Thursday, April 07, 2005 9:59
PM
Subject: [Asterisk-Users] failover
outbound dialplan
Does anyone have a working
failover outbound calls that I could sponge a hint from? i.e.
Exten =>
_1NXXNXX,1,Dial(IAX/MyFirstCrappyIAXPeer/${EXTEN},60)
Exten =>
_1NXXNXX,2,Dial(IAX/IfMyFirstCrappyIAXPeerDontAnswerIn5SecDialMySecondCrappyIAXPeer
Exten =>
_1NXXNXX,3,Dial(IAX/IfTheyBothDontAnswerTryTheNextCrappyIAXPeer)
Exten =>
_1NXXNXX,4,Dial(ZAP/g1)(GiveUpTheyAllSuckSoUseThePRI)
___Asterisk-Users
mailing
listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo
UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime mysql problem?
can you send me a dump from SQL for this account? I have it working both ways, W - Original Message - From: "Matt Schulte" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, April 04, 2005 9:34 AM Subject: RE: [Asterisk-Users] Realtime mysql problem? Well, I made several posts. Basically realtime works fine on the system you register to, if you try to contact that peer from another Ast server (running realtime), it does a SELECT query and all finds the peer and continues to say "Unable to contact peer" as if the user doesn't exist. I even went as far as packet sniffing and noticed it doesn't ever go out on port 4569 or anything. Again, I've made several posts about this before for full details. :-) Thanks, Matt -Original Message- From: Wojciech Tryc [mailto:[EMAIL PROTECTED] Sent: Monday, April 04, 2005 8:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime mysql problem? What is your problem with IAX in realtime? I have it working (finally). Wojtek - Original Message - From: "Matt Schulte" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, April 04, 2005 9:01 AM Subject: RE: [Asterisk-Users] Realtime mysql problem? Sorry for the delay, do you have any clue when realtime will get added to stable? I never did get this working but before I go too much further I'd like to run production on a stable version.. I'll try out SIP today and let you know, the reason I'm using IAX is because everything SIP we do is through SER. Not to mention since realtime doesn't support qualify= and NAT mode must be manually set, it's kind of pointless to use Asterisk for SIP. :-) Just for curiosity sake, have you tried any SIP RealTime stuff? Perhaps this is an IAX problem? I remember > helping a guy a few weeks ago get his SIP RealTime working. This is the first IAX I've dealt with. And I have > no IAX stuff to test with. -Original Message- From: Matthew Boehm [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 30, 2005 9:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime mysql problem? Matt Schulte wrote: How do you toggle the realtime "cache"? Check in the configs/iax.conf.sample file of a recent CVS download and it should be in there. If there were too many fields in the table, could you foresee this being a problem? No, because I have lots of extra company specific fields in my sip_users table that asterisk doesn't use at all and I've had no problems. ie iax users have peercontext and auth. Just for curiosity sake, have you tried any SIP RealTime stuff? Perhaps this is an IAX problem? I remember helping a guy a few weeks ago get his SIP RealTime working. This is the first IAX I've dealt with. And I have no IAX stuff to test with. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime mysql problem?
What is your problem with IAX in realtime? I have it working (finally). Wojtek - Original Message - From: "Matt Schulte" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, April 04, 2005 9:01 AM Subject: RE: [Asterisk-Users] Realtime mysql problem? Sorry for the delay, do you have any clue when realtime will get added to stable? I never did get this working but before I go too much further I'd like to run production on a stable version.. I'll try out SIP today and let you know, the reason I'm using IAX is because everything SIP we do is through SER. Not to mention since realtime doesn't support qualify= and NAT mode must be manually set, it's kind of pointless to use Asterisk for SIP. :-) Just for curiosity sake, have you tried any SIP RealTime stuff? Perhaps this is an IAX problem? I remember > helping a guy a few weeks ago get his SIP RealTime working. This is the first IAX I've dealt with. And I have > no IAX stuff to test with. -Original Message- From: Matthew Boehm [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 30, 2005 9:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime mysql problem? Matt Schulte wrote: How do you toggle the realtime "cache"? Check in the configs/iax.conf.sample file of a recent CVS download and it should be in there. If there were too many fields in the table, could you foresee this being a problem? No, because I have lots of extra company specific fields in my sip_users table that asterisk doesn't use at all and I've had no problems. ie iax users have peercontext and auth. Just for curiosity sake, have you tried any SIP RealTime stuff? Perhaps this is an IAX problem? I remember helping a guy a few weeks ago get his SIP RealTime working. This is the first IAX I've dealt with. And I have no IAX stuff to test with. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Timecard application
the CID is too easy to spoof... W - Original Message - From: "Chuck Bunn" <[EMAIL PROTECTED]> To: Sent: Thursday, March 31, 2005 3:50 PM Subject: [Asterisk-Users] Timecard application Hi, Does anyone know of a time card application that could be used with Asterisk. I want to be able to dial in to an extension and then type in my employee id. The caller ID would record the number called from and the time called. This would be helpful for time logging of traveling employees such as nurses visiting a home. The caller ID would verify that they are where they are supposed and at what time the arrived and then left. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] username/password for PolyCom IP500 web interface?
Polycom and 456 - Original Message - From: "Garrett Nelson" <[EMAIL PROTECTED]> To: Sent: Wednesday, March 30, 2005 10:24 AM Subject: [Asterisk-Users] username/password for PolyCom IP500 web interface? Ok, I am still working on getting this PolyCom phone working with Asterisk. I have been looking all over, but I have not been able to find the username and password for the web interface on this phone. I found some site that said it was Polycom and spip, but that does not work. Anyone else have any ideas what it might be? Both PolyCom and the place I bought the phone from are useless for support. -Garrett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX realtime dynamic
Good Afternoon, I am just playing with realtime on one of my boxes (running obviously HEAD). The voicemail portion works just fine, howevere I am having difficulties getting iax portion to work. Sip and extensions left for later for now. Could anyone send me sample database dump of his/her config? Also, what about the iax.conf should i leave the [general] section? or remove the file completly. Basically, at this point it cannont create any iax channels unless the user name and password exists in extensions.conf. Thanks, Wojtek ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] make a call based on SMS request
At this point (proof of concept) I am using a GSM phone connected over the serial cable. The only problem is that all incoming/outgoing SMS messages are using the same number assigned to the SIM card on this phone. W - Original Message - From: "Iqbal" <[EMAIL PROTECTED]> To: Sent: Saturday, March 26, 2005 5:00 PM Subject: Re: [Asterisk-Users] make a call based on SMS request How are you connecting to the SMS gateway, are you calling an external script to send a http request or something, or is there a way of using SMPP Iqbal On 3/26/2005, "Wojciech Tryc" <[EMAIL PROTECTED]> wrote: I have such setup in testing. SER as SMS gateway and callback through Asterisk. W - Original Message - From: "Cristian T" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, March 26, 2005 12:53 AM Subject: [Asterisk-Users] make a call based on SMS request Hola I have a costumer whit this idea: I am looking for a solution that will make a call based on SMS request. Can you solve this problem with Asterisk? Let me know if you have the solution and what exactly it does. This is posible??? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] make a call based on SMS request
I have such setup in testing. SER as SMS gateway and callback through Asterisk. W - Original Message - From: "Cristian T" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, March 26, 2005 12:53 AM Subject: [Asterisk-Users] make a call based on SMS request Hola I have a costumer whit this idea: I am looking for a solution that will make a call based on SMS request. Can you solve this problem with Asterisk? Let me know if you have the solution and what exactly it does. This is posible??? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chanisavail and IAX2
Yes, there is more, but I don't remember of hand. I end up "downgradin" to the 1.0.7 W - Original Message - From: "Anton Krall" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Wednesday, March 23, 2005 12:29 PM Subject: RE: [Asterisk-Users] Chanisavail and IAX2 Damn! First I see something doesn't work with cvs-head but does in stable :) Any timeframe on when it will work again on cvs-head? Any other stuff like this one that doesn't work on head? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wojciech Tryc Sent: Miércoles, 23 de Marzo de 2005 11:16 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Chanisavail and IAX2 it doesn't work with current CVS, it works with 1.0.7 - Original Message - From: "Anton Krall" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Wednesday, March 23, 2005 9:59 AM Subject: [Asterisk-Users] Chanisavail and IAX2 Guys. Anybody doing ChanisAvail on IAX2 channels? Im trying to do this: exten => s,7,ChanIsAvail(IAX2/anton:[EMAIL PROTECTED]) But I get that the chan is unavailable eventhough I can make calls to that channel. Is there any chatch? The channels is defined as peer and Ialso tried doing a register on iax.conf for that channel. Everything is registering ok and I CAN make the call. Any tips? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chanisavail and IAX2
it doesn't work with current CVS, it works with 1.0.7 - Original Message - From: "Anton Krall" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Wednesday, March 23, 2005 9:59 AM Subject: [Asterisk-Users] Chanisavail and IAX2 Guys. Anybody doing ChanisAvail on IAX2 channels? Im trying to do this: exten => s,7,ChanIsAvail(IAX2/anton:[EMAIL PROTECTED]) But I get that the chan is unavailable eventhough I can make calls to that channel. Is there any chatch? The channels is defined as peer and Ialso tried doing a register on iax.conf for that channel. Everything is registering ok and I CAN make the call. Any tips? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ChanIsAvail for IAX2 broken in CVS current?
Hi, I am still looking for confirmation that ChanISAvail in CVS current doesn't work properly anymore. My config hasn't changed (it worked for months)... Right now, every time ChanIsAvail jumps to n+101 regardless if tested channel is available or not. Is it broken? Maybe the syntax has changed? BTW: it works like before in the 1.0.x. Regards, Wojtek ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ChanIsAvail not working anymore
Good Evening,
It seems that ChanIsAvail stopped working with the latest CVS, at least for
IAX2 channels
My dial plan hasn't changed, but the ChanIsAvail always goes n+101, same
dialplan works just fine with 1.0.7
Could anyone confirm that?
Regards,
Wojtek
snip...
exten => _XXX,1,Chanisavail(IAX2/pikatech)
exten => _XXX,2,Macro(enum-call-local,local,${EXTEN})
exten => _XXX,102,GoTo(local,${EXTEN},1)
...snip
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIPJet and g.711
Just in my dial plan. I am not using any real Lease cost routing package, as a matter of fact I am developing one but it's not ready yet. W - Original Message - From: "Robert Augustyn" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Sunday, March 13, 2005 6:21 PM Subject: RE: [Asterisk-Users] VoIPJet and g.711 Thanks, Are you doing it by setting the lowest cost? Is there anything in Asterisk which does it? Thanks, robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wojciech Tryc Sent: Sunday, March 13, 2005 12:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VoIPJet and g.711 Robert, Nufone, but it all depends on the destination. For some is gafachi, for some is VoicePulse etc.. W - Original Message - From: "Robert Augustyn" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" ; "'Justin Richards'" <[EMAIL PROTECTED]> Sent: Sunday, March 13, 2005 12:09 PM Subject: RE: [Asterisk-Users] VoIPJet and g.711 > Wojtek, > What are you using for your primary route? > robert > >> -Original Message- >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On Behalf Of >> Wojciech Tryc >> Sent: Sunday, March 13, 2005 9:31 AM >> To: Justin Richards; Asterisk Users Mailing List - >> Non-Commercial Discussion >> Subject: Re: [Asterisk-Users] VoIPJet and g.711 >> >> I can see errors on the console, g.729 and ilbc works no problem. >> I endup moving VoIPjet to the secondary route. >> Wojtek >> - Original Message - >> From: "Justin Richards" <[EMAIL PROTECTED]> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion" >> >> Sent: Saturday, March 12, 2005 11:00 PM >> Subject: Re: [Asterisk-Users] VoIPJet and g.711 >> >> >> >I am having problem with voipjet and g.711 (ulaw) as well. I tried >> > ilbc with no luck. basically my outbound call connects, i can hear >> > them talk, but they can't hear me. >> > >> > i am not getting errors in console with either ulaw or ilbc, just no >> > audio to the called party. >> > >> > it worked great yesterday, and I haven't changed anything.. my >> > connection to voicepulse (same settings ad voipjet) works great. >> > >> > On Fri, 11 Mar 2005 12:33:14 -0500, Wojciech Tryc >> <[EMAIL PROTECTED]> wrote: >> >> I am experiencing problems connecting to VoIPjet with >> g.711. It works >> >> with >> >> g.729 and ilbc. It used to work... >> >> Anyone? >> >> Regards, >> >> Wojtek >> >> >> >> ___ >> >> Asterisk-Users mailing list >> >> Asterisk-Users@lists.digium.com >> >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> To UNSUBSCRIBE or update options visit: >> >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> > ___ >> > Asterisk-Users mailing list >> > Asterisk-Users@lists.digium.com >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> > To UNSUBSCRIBE or update options visit: >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> >> ___ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIPJet and g.711
Robert, Nufone, but it all depends on the destination. For some is gafachi, for some is VoicePulse etc.. W - Original Message - From: "Robert Augustyn" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" ; "'Justin Richards'" <[EMAIL PROTECTED]> Sent: Sunday, March 13, 2005 12:09 PM Subject: RE: [Asterisk-Users] VoIPJet and g.711 Wojtek, What are you using for your primary route? robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wojciech Tryc Sent: Sunday, March 13, 2005 9:31 AM To: Justin Richards; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VoIPJet and g.711 I can see errors on the console, g.729 and ilbc works no problem. I endup moving VoIPjet to the secondary route. Wojtek - Original Message - From: "Justin Richards" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, March 12, 2005 11:00 PM Subject: Re: [Asterisk-Users] VoIPJet and g.711 >I am having problem with voipjet and g.711 (ulaw) as well. I tried > ilbc with no luck. basically my outbound call connects, i can hear > them talk, but they can't hear me. > > i am not getting errors in console with either ulaw or ilbc, just no > audio to the called party. > > it worked great yesterday, and I haven't changed anything.. my > connection to voicepulse (same settings ad voipjet) works great. > > On Fri, 11 Mar 2005 12:33:14 -0500, Wojciech Tryc <[EMAIL PROTECTED]> wrote: >> I am experiencing problems connecting to VoIPjet with g.711. It works >> with >> g.729 and ilbc. It used to work... >> Anyone? >> Regards, >> Wojtek >> >> ___ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIPJet and g.711
I can see errors on the console, g.729 and ilbc works no problem. I endup moving VoIPjet to the secondary route. Wojtek - Original Message - From: "Justin Richards" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, March 12, 2005 11:00 PM Subject: Re: [Asterisk-Users] VoIPJet and g.711 I am having problem with voipjet and g.711 (ulaw) as well. I tried ilbc with no luck. basically my outbound call connects, i can hear them talk, but they can't hear me. i am not getting errors in console with either ulaw or ilbc, just no audio to the called party. it worked great yesterday, and I haven't changed anything.. my connection to voicepulse (same settings ad voipjet) works great. On Fri, 11 Mar 2005 12:33:14 -0500, Wojciech Tryc <[EMAIL PROTECTED]> wrote: I am experiencing problems connecting to VoIPjet with g.711. It works with g.729 and ilbc. It used to work... Anyone? Regards, Wojtek ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIPJet and g.711
I am experiencing problems connecting to VoIPjet with g.711. It works with g.729 and ilbc. It used to work... Anyone? Regards, Wojtek ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callback on busy
Yes, it's relatively easy Contact me privately, if you need a hand W - Original Message - From: "Paradise Dove" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, March 01, 2005 1:47 PM Subject: [Asterisk-Users] callback on busy hi, is there anyway to implement "callback on busy" and "callback on no answer" on asterisk? has anybody done this before? thanks, Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream and VLANs
Do you know if there is a way around it? I can not use untagged VLANs as the phone and the PC are on physically seperate networks...Well, I know , I can just run 2 ethernet cables W - Original Message - From: "Bartosz Jozwiak" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, March 01, 2005 7:08 AM Subject: Re: [Asterisk-Users] Re: Grandstream and VLANs Just a couple of guesses: Have you configured the switch to supply a VLAN trunk to the phone? Yes Since the phone lets you configure actual tagging, that's what it needs; if you've just enabled VLANs on the switch, and placed the port the phone is on in a specific VLAN, the phone should not have tagging enabled. Exactly, both phone and the switch are configured to use tagging and to use the same VLAN # If you've got the switch doing trunk mode on that port, and the phone set up to use the right VLAN within the trunk, are you perhaps using a Cisco switch, and have accidentally set it up to use ISL encapsulation (Cisco's proprietary method) on the trunk? The phone does 802.1Q, so Cisco switches need "switchport trunk encapsulation dot1q" on the trunking interface. No, I am using the HP Pro Curve, Haven't tried Cisco yes but I woul assume that my switches are operating just fine Thanks, Wojtek I have the same problem with VLAN and Grandstream BG phone. I am using HP switches. There is a bug in the firmware of the phone, Grandstream tech. knows about it already. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream and VLANs
Just a couple of guesses: Have you configured the switch to supply a VLAN trunk to the phone? Yes Since the phone lets you configure actual tagging, that's what it needs; if you've just enabled VLANs on the switch, and placed the port the phone is on in a specific VLAN, the phone should not have tagging enabled. Exactly, both phone and the switch are configured to use tagging and to use the same VLAN # If you've got the switch doing trunk mode on that port, and the phone set up to use the right VLAN within the trunk, are you perhaps using a Cisco switch, and have accidentally set it up to use ISL encapsulation (Cisco's proprietary method) on the trunk? The phone does 802.1Q, so Cisco switches need "switchport trunk encapsulation dot1q" on the trunking interface. No, I am using the HP Pro Curve, Haven't tried Cisco yes but I woul assume that my switches are operating just fine Thanks, Wojtek -tih -- Don't ascribe to stupidity what can be adequately explained by ignorance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream and VLANs
I need tagging as I will have also PC's on the hub connected to the same port on the switch. PCs will be on a separate VLAN. As I have to tag one device I prefer to tag phones (apparently supported). Again, Cisco phones no problem (as expected) :) Anyone tried Grandstream's VLAN tagging? W - Original Message - From: "Colin Anderson" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Monday, February 28, 2005 4:53 PM Subject: RE: [Asterisk-Users] Grandstream and VLANs As it should. As a stupid work-around, you could possibly put *everything else* on a seperate VLAN from the phones and you would have kind of a reverse-VLAN which would have the net same effect, this would be fine for 10 PC's but 100? 200? fuggedaboutit. -Original Message- From: Wojciech Tryc [mailto:[EMAIL PROTECTED] Sent: Monday, February 28, 2005 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Grandstream and VLANs Yes, I guess I will have to start looking at the packets. BTW: if I set the port to which grandstream is plugged to untagged vlan and leave the default VLAN 0 on the phone then everything works just fine W - Original Message - From: "Colin Anderson" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Monday, February 28, 2005 4:38 PM Subject: RE: [Asterisk-Users] Grandstream and VLANs I'd try as "The Tyrant" always suggests and make it as simple as possible i.e. -Isolate the phone to a seperate switch that supports VLAN -Plug in a PC to the same switch -Turn OFF VLAN-ing on the switch, PC & phone -Assign a static IP to the PC & phone -Fire up Ethereal on the PC, start recording, ping the phone. -Examine the output. You should get something. -VLAN the two ports from the switch, assuming it's managed. Ping the phone again, and examine the output. Again, you should see something -VLAN the PC to the same VLAN as the switch. Make sure you change the Ethereal capture to the VLAN interface on the PC. Ping the phone. You should get nothing from the phone. -VLAN the phone and ping it. You should get something. Somewhere along the way, you should get enough information to make a deduction about what the GS is doing. It wouldn't suprise me if GS's VLANing is poopoo; everything about these phones seems to be "sacrifice quality at all cost". My users hate them. HTH. -Original Message- From: Wojciech Tryc [mailto:[EMAIL PROTECTED] Sent: Monday, February 28, 2005 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Grandstream and VLANs Yes :) It's not DHCP as the phone won't work even with statically assigned IP. It basically looks like Grandstream is tagging and/or reading the tagged packets incorectly. W - Original Message - From: "Colin Anderson" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Monday, February 28, 2005 3:47 PM Subject: RE: [Asterisk-Users] Grandstream and VLANs >I can not even get IP anymore from my DHCP Hate to ask the obvious, but is the DHCP server on the same VLAN? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream and VLANs
Yes, I guess I will have to start looking at the packets. BTW: if I set the port to which grandstream is plugged to untagged vlan and leave the default VLAN 0 on the phone then everything works just fine W - Original Message - From: "Colin Anderson" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Monday, February 28, 2005 4:38 PM Subject: RE: [Asterisk-Users] Grandstream and VLANs I'd try as "The Tyrant" always suggests and make it as simple as possible i.e. -Isolate the phone to a seperate switch that supports VLAN -Plug in a PC to the same switch -Turn OFF VLAN-ing on the switch, PC & phone -Assign a static IP to the PC & phone -Fire up Ethereal on the PC, start recording, ping the phone. -Examine the output. You should get something. -VLAN the two ports from the switch, assuming it's managed. Ping the phone again, and examine the output. Again, you should see something -VLAN the PC to the same VLAN as the switch. Make sure you change the Ethereal capture to the VLAN interface on the PC. Ping the phone. You should get nothing from the phone. -VLAN the phone and ping it. You should get something. Somewhere along the way, you should get enough information to make a deduction about what the GS is doing. It wouldn't suprise me if GS's VLANing is poopoo; everything about these phones seems to be "sacrifice quality at all cost". My users hate them. HTH. -Original Message- From: Wojciech Tryc [mailto:[EMAIL PROTECTED] Sent: Monday, February 28, 2005 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Grandstream and VLANs Yes :) It's not DHCP as the phone won't work even with statically assigned IP. It basically looks like Grandstream is tagging and/or reading the tagged packets incorectly. W - Original Message - From: "Colin Anderson" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Monday, February 28, 2005 3:47 PM Subject: RE: [Asterisk-Users] Grandstream and VLANs >I can not even get IP anymore from my DHCP Hate to ask the obvious, but is the DHCP server on the same VLAN? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream and VLANs
Yes :) It's not DHCP as the phone won't work even with statically assigned IP. It basically looks like Grandstream is tagging and/or reading the tagged packets incorectly. W - Original Message - From: "Colin Anderson" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Monday, February 28, 2005 3:47 PM Subject: RE: [Asterisk-Users] Grandstream and VLANs >I can not even get IP anymore from my DHCP Hate to ask the obvious, but is the DHCP server on the same VLAN? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream and VLANs
I am having hard time to get VLAN tagging working with Grandstream 101 phone. As soon as I enable tagging on the switch and configure the phone to tag packets with corespodning VLAN ID #. I can not even get IP anymore from my DHCP. I have to reset the phone to factory default. I've tried different Grandstream phones with various firmware without any luck. Obviously, the same idea works great with Ciscos Any ideas? Regards, Wojtek ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Toronto?
well, I am in Ottawa...only 50mins by air :) Wojtek - Original Message - From: "Leif Madsen" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, January 08, 2005 12:37 PM Subject: Re: [Asterisk-Users] Toronto? On Sat, 8 Jan 2005 05:40:02 -0500, Jim Van Meggelen <[EMAIL PROTECTED]> wrote: Anyone in the Toronto area interested in getting together to share notes and swap war stories? I'm in Oakville, right across from Sheridan College. So I guess I can be considered part of the GTA at least. But you already knew that :) Leif Madsen. http://www.leifmadsen.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Presentations
At AstriCON Steven Sokol said that copy of the presentations should be available on-line within 2 weeks. Did anyone got their user name and password to access them? Regards, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 released
What's the URL. I have 7960 with the old firmware, it works fine..but I wouldn't mind to update to the latest/ Wojtek - Original Message - From: "Shaun Ewing" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, August 17, 2004 2:28 AM Subject: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 released > Hi All, > > Just a heads up - I was looking around the Cisco FTP a little while > ago and noticed that the SIP 7.2 images for Cisco IP Phone 7940/7960 > were released yesterday (16th August). > > No new features - all bug fixes according to the release notes. I've > already started using it. > > I thought those of you running the Cisco phones and the appropriate > access who didn't yet know would like to know. > > -Shaun > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX problem; one end sounds like on fast forward
Hi, I have some issues with communication between to * servers. They are connected over DSL (3Mbps). One is behind NAT and the other on routable network. Almost every time caller will hear the other end like fast forward while the other end will have perfect quality. It doesn't matter if we use SIP phones (Cisco and Grandstreams) or analog sets via Sipura-2K. If I call the city through Mediatrix 1204 the quality is perfect. I am suspecting that this problem is related to jitter, but can not resolve it. I've tried using ulaw and ilbc with similar results. Both sites are configured to use IAX trunking and both have X101P to provide clocking (on one end the X101P is in red-alarm state as the line is not plugged in into X101P). I am tempted to switch to SIP for interoffice communication but first I want to try few more things.. Any suggestions? Regards, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
Basically outgoing calls through zap channels doesn't detect that the other end answered. In my cdr I see hang-up no answer, plus the console shows that the channel is ringing..while I am actually talking to someone. Incoming calls seems to be fine. Wojtek - Original Message - From: "Wojciech Tryc" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, June 24, 2004 7:27 PM Subject: Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming! > I have similar problem with outbound calls... > Wojtek > - Original Message - > From: "Brent Franks" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Thursday, June 24, 2004 7:16 PM > Subject: RE: [Asterisk-Users] tdm (and x100p?) echo - fix is coming! > > > > > > > -Original Message- > > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > > [EMAIL PROTECTED] On Behalf Of Rich Adamson > > > Sent: Thursday, June 24, 2004 5:01 PM > > > Be careful with that thought... here's the three lines that were > > > manually changed for testing purposes only (these would have been > > prior to > > > yesterday's change to chan_zap.c): > > > ~1195: x = 800; > > > ~1636: strcpy(p->echorest, "ww"); > > > ~1637: strcpy(p->echorest + 2, > > > > > > Changing x = 400 to x = 800 fixed the echo problem, but caused > > outbound > > > dialing to totally fail. The pstn line would be seized, but the dtmf > > > sent to the CO was less then acceptable. > > > > > > Changing lines 1636 (from "w" to "ww") and line 1637 (from "1" to "2") > > > brought the outbound dialing back into a functional state. Since I'm > > > not a programmer, I don't really know what those lines are doing. > > > > > > Mark then used that info to write the code for implementing > > > echotrainging=800 as a configurable option. > > > > > > Does today's code support changing all three values? (Since the > > example > > > in the config files suggest two specific choices, I'd bet that using > > > a value of 600 or 1200 or whatever does cause an issue with the > > outbound > > > dialing, etc.) > > > > My report > > > > With our current setup we have an Adtran TotalAccess 750 connected to a > > T100P. There are 5 incoming FXO lines from Verizon. > > > > We use about 15 Polycom SIP IP500 phones. > > > > I updated to today's CVS and still noticed an echo in the middle of > > nearly every call. The echo would come in after 2 or 3 minutes, last > > for 30 seconds and then disappear. I will report on our user's > > experiences tomorrow. > > > > Regards, > > > > - Brent > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
I have similar problem with outbound calls... Wojtek - Original Message - From: "Brent Franks" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, June 24, 2004 7:16 PM Subject: RE: [Asterisk-Users] tdm (and x100p?) echo - fix is coming! > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Rich Adamson > > Sent: Thursday, June 24, 2004 5:01 PM > > Be careful with that thought... here's the three lines that were > > manually changed for testing purposes only (these would have been > prior to > > yesterday's change to chan_zap.c): > > ~1195: x = 800; > > ~1636: strcpy(p->echorest, "ww"); > > ~1637: strcpy(p->echorest + 2, > > > > Changing x = 400 to x = 800 fixed the echo problem, but caused > outbound > > dialing to totally fail. The pstn line would be seized, but the dtmf > > sent to the CO was less then acceptable. > > > > Changing lines 1636 (from "w" to "ww") and line 1637 (from "1" to "2") > > brought the outbound dialing back into a functional state. Since I'm > > not a programmer, I don't really know what those lines are doing. > > > > Mark then used that info to write the code for implementing > > echotrainging=800 as a configurable option. > > > > Does today's code support changing all three values? (Since the > example > > in the config files suggest two specific choices, I'd bet that using > > a value of 600 or 1200 or whatever does cause an issue with the > outbound > > dialing, etc.) > > My report > > With our current setup we have an Adtran TotalAccess 750 connected to a > T100P. There are 5 incoming FXO lines from Verizon. > > We use about 15 Polycom SIP IP500 phones. > > I updated to today's CVS and still noticed an echo in the middle of > nearly every call. The echo would come in after 2 or 3 minutes, last > for 30 seconds and then disappear. I will report on our user's > experiences tomorrow. > > Regards, > > - Brent > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax detection
Please ignore my problem, I just added faxdetection to zapata.conf and everything is back to normal. Thanks, W - Original Message - From: "Wojciech Tryc" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, June 24, 2004 6:37 PM Subject: [Asterisk-Users] fax detection > Everything but fax detection seems to be fixed in the latest CVS. > Anyincoming fax on Zap channel does not get detected. Anyone? > Thanks, > Wojtek > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fax detection
Everything but fax detection seems to be fixed in the latest CVS. Anyincoming fax on Zap channel does not get detected. Anyone? Thanks, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: -- [Asterisk-Users] Serious issues with current CVS?
Same here, I also lost DTMF on some SIP devices (Grandstream phones) and fax detection on Zap devices. W. > Is anybody else having serious issues with the current version from CVS? I > just compiled and installed it and: > > 1) I was able to establish one and only one call before things went weird. > 2) It stopped responding to IAX calls after the first. Completely ignored > any subsequent commands, including hangup. > 3) It stopped responding to CLI commands. > 4) The only way to kill it was to use kill. > > Is anybody else experiencing this? > > Thanks, > > Steven > > Steven Sokol > Owner/Manager > Sokol & Associates, LLC > > Phone: 816.822.1807 > IaxTel: 700.613.9004 > Web:http://www.sokol-associates.com > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] latest CVS DTMF with Grandstream broken
Just installed the latest CVS and the * does not recognize DTMF from grandstream devices. The Sipura 2k ports are fine. Also, it seems that I lost fax receiving functionality. The incoming fax just rings and gets my IVR it doesn't get detected I downgraded to resolve DTMF with Grandstream and everything is back to normal, still working on fax... Am I missing something? Thanks, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enum problem with latest cvs
Any suggestions on how to resolve this problem? :) Thanks, Wojtek - Original Message - From: "Aaron J. Angel" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, June 21, 2004 12:01 AM Subject: RE: [Asterisk-Users] enum problem with latest cvs > Try RFC3761. It specifies E2U+ under section 2.4.2. It obsoletes > RFC2916, and nothing has superseded it yet. > > Brian K. West [EMAIL PROTECTED] wrote: > > 3762 is for h323 only: > > > > RFC 3762 - Telephone Number Mapping (ENUM) Service > > Registration for H.323 > > > > 2916 is a bit more general: > > > > RFC 2916 - E.164 number and DNS > > > > Then we have: > > > > RFC 2915 - The Naming Authority Pointer (NAPTR) DNS Resource Record > > > > I was pointing out that E2U+IAX2 was backwards.. but then > > again asterisk > > doesn't really care about that... at this point. > > > > bkw > > > > > > - Original Message - > > From: "Duane" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Sunday, June 20, 2004 9:48 PM > > Subject: Re: [Asterisk-Users] enum problem with latest cvs > > > > > >> Brian K. West wrote: > >> > >>> but then again what do I know.. I have only been using > > enum for about a > > year > >>> now. > >> > >> RFC's change, if you want to stick to the standards you have to keep > >> up with them... > >> > >> 3762 > 2916 > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enum problem with latest cvs
This is only intermittent problem!!?!? Wojtek - Original Message - From: "Brian K. West" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, June 20, 2004 3:50 PM Subject: Re: [Asterisk-Users] enum problem with latest cvs > Your regexp is WRONG > > 1.1.enum.blah.net naptr = 2 40 "u" "iax2+E2U" > "!^\\+(.*)$!iax2:[EMAIL PROTECTED]/\\1!" . > > Thats a valid enum naptr record. > > It would translate into iax2:[EMAIL PROTECTED]/11 > > bkw > > - Original Message - > From: "Wojciech Tryc" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Sunday, June 20, 2004 2:27 PM > Subject: [Asterisk-Users] enum problem with latest cvs > > > > Hi, > > I posted an error message I was getting while using enum with the latest > > CVS, but the problem disappered. > > Well, it seems to be intermitten. > > The messages below: > > > > Jun 20 15:23:30 WARNING[1218565440]: enum.c:186 parse_naptr: Regex > > compilation error (regex = "!^+16131234567$"). > > Jun 20 15:23:30 WARNING[1218565440]: enum.c:264 enum_callback: Failed to > > parse naptr :( > > Jun 20 15:23:30 WARNING[1218565440]: dns.c:141 dns_parse_answer: Failed to > > parse result > > Jun 20 15:23:30 WARNING[1218565440]: dns.c:183 ast_search_dns: Parse error > > > > What is internesting is that this is happening only with 1 number, I have > 2 > > other numbers registered and everything works fine with the other 2. > > > > Regards, > > Wojtek > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enum problem with latest cvs
The number is 1-613-823-1716 and the enum service is e164.org. The most interesting part is that this is intermittent problem, sometimes it works sometimes it doesn't work. Again, any other lookups works just fine. Thanks, Wojtek - Original Message - From: "Aaron J. Angel" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, June 20, 2004 3:52 PM Subject: RE: [Asterisk-Users] enum problem with latest cvs > [EMAIL PROTECTED] wrote: > > Hi, > > I posted an error message I was getting while using enum with > > the latest CVS, but the problem disappered. > > Well, it seems to be intermitten. > > The messages below: > > > > Jun 20 15:23:30 WARNING[1218565440]: enum.c:186 parse_naptr: > > Regex compilation error (regex = "!^+16131234567$"). > > Jun 20 15:23:30 WARNING[1218565440]: enum.c:264 > > enum_callback: Failed to parse naptr :( Jun 20 15:23:30 > > WARNING[1218565440]: dns.c:141 dns_parse_answer: Failed to > > parse result Jun 20 15:23:30 WARNING[1218565440]: dns.c:183 > > ast_search_dns: Parse error > > > > What is internesting is that this is happening only with 1 > > number, I have 2 other numbers registered and everything works fine > > with the other 2. > > If that's the case, it's likely not a problem with Asterisk. Did you check > the syntax of the regexp in the NAPTR record? Without knowing the number > being looked up and the ENUM service being used, not much can be done to > troubleshoot. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] enum problem with latest cvs
Hi, I posted an error message I was getting while using enum with the latest CVS, but the problem disappered. Well, it seems to be intermitten. The messages below: Jun 20 15:23:30 WARNING[1218565440]: enum.c:186 parse_naptr: Regex compilation error (regex = "!^+16131234567$"). Jun 20 15:23:30 WARNING[1218565440]: enum.c:264 enum_callback: Failed to parse naptr :( Jun 20 15:23:30 WARNING[1218565440]: dns.c:141 dns_parse_answer: Failed to parse result Jun 20 15:23:30 WARNING[1218565440]: dns.c:183 ast_search_dns: Parse error What is internesting is that this is happening only with 1 number, I have 2 other numbers registered and everything works fine with the other 2. Regards, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enum problems with the latest CVS
Please ignore this message, everything is back to normal :) Wojtek - Original Message - From: "Wojciech Tryc" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, June 19, 2004 11:35 AM Subject: [Asterisk-Users] enum problems with the latest CVS > Hi, > I just recompiled * with the latest CVS. > I am using enum in my extensions to dial first over the internet, if > applicable. > Everything was working perfectly, but now after installing the latest CVS I > am getting the following errors and enum lookup doesn't work. > > Jun 19 11:29:41 WARNING[1218565440]: enum.c:186 parse_naptr: Regex > compilation error (regex = "!^+1613999$"). > Jun 19 11:29:41 WARNING[1218565440]: enum.c:264 enum_callback: Failed to > parse naptr :( > Jun 19 11:29:41 WARNING[1218565440]: dns.c:141 dns_parse_answer: Failed to > parse result > Jun 19 11:29:41 WARNING[1218565440]: dns.c:183 ast_search_dns: Parse error > > I am just wondering if anyone expereinced similar problem, any suggestion > will be appreciated. > Regards, > Wojtek > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] enum problems with the latest CVS
Hi, I just recompiled * with the latest CVS. I am using enum in my extensions to dial first over the internet, if applicable. Everything was working perfectly, but now after installing the latest CVS I am getting the following errors and enum lookup doesn't work. Jun 19 11:29:41 WARNING[1218565440]: enum.c:186 parse_naptr: Regex compilation error (regex = "!^+1613999$"). Jun 19 11:29:41 WARNING[1218565440]: enum.c:264 enum_callback: Failed to parse naptr :( Jun 19 11:29:41 WARNING[1218565440]: dns.c:141 dns_parse_answer: Failed to parse result Jun 19 11:29:41 WARNING[1218565440]: dns.c:183 ast_search_dns: Parse error I am just wondering if anyone expereinced similar problem, any suggestion will be appreciated. Regards, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Primustel a.k.a. Lingo $20/month unlimited service
I had similar problem with D-Link box and Voicetronix as well as with Mediatrix. Wojtek - Original Message - From: "Ryan Courtnage" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, June 11, 2004 11:02 AM Subject: Re: [Asterisk-Users] Primustel a.k.a. Lingo $20/month unlimited service > On 10-Jun-04, at 7:47 PM, Philippe Daoust wrote: > > > If it's the same service that they offer up here in Canada it's MGCP > > based. > > I'd be interested to know if * can work with this (I assume it should > > but I > > have never tried it). > > FYI - We've been trying to run the FXS port on one of these DLinks > (also Primus), into the FXO port of a TDM400P. We had a reoccurring > problem where after several hours, the line would go to a 'dead' state. > (ie: after several hours, if you try to make an outgoing call, you > will hear nothing but white-noise ... the call will never go through). > > I suspect it has something to do with signaling (we had set the zapata > config to use kewl-start). If anyone gets this working properly in a > digium/* environment, I'd appreciate your feedback. > > Cheers > Ryan > > > The unit they sent me is the same D-Link. It's > > HUGE!!! The largest ATA I have ever seen... > > > > BTW, Primus has been offering this service in Canada for about 2-3 > > months > > now (called "Talk Broadband" here). > > > ... > Ryan Courtnage > Coalescent Systems Inc > 403.244.8089 > www.voxbox.ca > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * freezing
Hi, I am running * with 1 Zaptel 101 and 4 port fXO by Voicetronix. I also have Mediatrix 1204 and few Sipura's 2000. Everything was working just perfect before I've added the Voicetronix OpenPort 4. Now, most of the time when Voicetronix device is used the console freezes and no new calls are being accepted. Calls in progress are fine. My problem is that I can not even see if anyone is using the * and if I can kill it and restart it. Very seldom, calls going through Digium card create similar behaviour. Is this related to IRQ? Anyone here uses Voictronix together with Digium board? Any suggestions will be appreciated. I am ready to remove the Voicetronix board but I am going to loose 4 lines so I would prefer to resolve it some how. I am running almost the latest CVS Here is the output from lsmod and /proc/interrupts as well vpbscan and vpb config [EMAIL PROTECTED] asterisk]# /sbin/lsmod Module Size Used byNot tainted soundcore 6404 0 (autoclean) vpb 139136 1 wcfxo 9344 1 zaptel179712 6 [wcfxo] lp 8996 0 (autoclean) parport37056 0 (autoclean) [lp] iptable_filter 2412 0 (autoclean) (unused) ip_tables 15096 1 [iptable_filter] autofs 13268 0 (autoclean) (unused) e100 60644 1 keybdev 2944 0 (unused) mousedev5492 0 (unused) hid22148 0 (unused) input 5856 0 [keybdev mousedev hid] usb-uhci 26348 0 (unused) usbcore78784 1 [hid usb-uhci] ext3 70784 2 jbd51892 2 [ext3] aic7xxx 141236 3 sd_mod 13452 6 scsi_mod 107128 2 [aic7xxx sd_mod] [EMAIL PROTECTED] asterisk]# more /proc/interrupts CPU0 0: 26902057 XT-PIC timer 1:342 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 0 XT-PIC usb-uhci 8: 1 XT-PIC rtc 9: 268986324 XT-PIC usb-uhci, wcfxo 11:9093965 XT-PIC aic7xxx, eth0 12: 2 XT-PIC PS/2 Mouse 14: 0 XT-PIC ide0 NMI: 0 ERR: 0 [EMAIL PROTECTED] vpb-detect]# ./vpbscan CARD1:UNKNOWN:irq=10 sub=56345654 BOARDS:1 [EMAIL PROTECTED] vpb-detect]# ./vpbconf Cards detected:1 BOARD 1 vpb_pconf[0][0] = 0 vpb_pconf[0][1] = 0 vpb_pconf[0][2] = 0 vpb_pconf[0][3] = 0 vpb_pconf[0][4] = 0 vpb_pconf[0][5] = 0 vpb_pconf[0][6] = 0 vpb_pconf[0][7] = 0 vpb_pconf[0][8] = 0 vpb_pconf[0][9] = 0 vpb_pconf[0][10] = 0 vpb_pconf[0][11] = 0 MODEL : VPB4 DATE : 13/02/2004 REVISION : 20.03 SERIAL NUMBER : 40701438 STATIONS[1]: TRUNKS[1]: 0 1 2 3 4 5 6 7 8 9 10 11 Any help will be greatly appreciated, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Primustel a.k.a. Lingo $20/month unlimited service
They don't provide "soft" accounts. You need to use their D-Link box which connects back to them using MGCP. Overall service is reasonable, acceptable for home users but definitely not good enough for business use. I am just about to send their units back. Thanks, Wojtek - Original Message - From: "Stephan Wik" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, June 10, 2004 4:46 AM Subject: Re: [Asterisk-Users] Primustel a.k.a. Lingo $20/month unlimited service > > On 10 Jun 2004, at 09:53, Simon Dorfman wrote: > > > $20 monthly plan with unlimited local and long-distance calling in > > North > > America (US & Canada) and Western Europe. Plus first three months > > free and > > free equipment. It doesn't say what hardware they send you. > > > > Sounds like a very good deal. > > > > I searched the list and voip-wiki and couldn't find any reviews about > > their > > service. Has anyone tried them? How is the service? Does it work > > with *? > > I just spoke with their tech support who says you have to use their > 'hardware' to connect. He had no idea what I was talking about when I > mentioned IAX or SIP :-( > > Stephan > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoicePulse problem
It seems that VoicePulse is down, incoming calls get busy, outgoing are timing out as * can not register with them. Could anyone confirm that? Thanks, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel 1-800 gateway down?
same with their 700 network w - Original Message - From: "Mark Musone" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, June 08, 2004 11:24 AM Subject: [Asterisk-Users] iaxtel 1-800 gateway down? > Does anyone know if the 1-800 iaxtel gateway is down? > I've been trying to use it all day today and asterisk says it's ringing: > > Channel (ContextExtensionPri ) State Appl. > Data > IAX2[iaxtel]/1 ( s1 ) Ringing AppDial > (Outgoing Line) > SIP/2201-a253 (home 1476626 1 )Ring Dial > IAX2/XXX:[EMAIL PROTECTED]/[EMAIL PROTECTED] > > > But I never hear a ringing on the actual phone, and it seems to stay in > this state (i.e. never gets to bridge mode) for a long time..to a point > that ijust hang up. > > > Thanks, > > Mark > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204
The Mediatrix box will not registered with * as the user name and password for sip are not yet implemented in their firmware. All what you have to do is to protect the box from the internet (firewall) and access is like: exten => _1905XXX,1,Dial(SIP/[EMAIL PROTECTED]) exten => _1905XXX,1,Congestion This way you basically have a pool of 4 outgoing lines. You can however route properly incoming calls. I hope this will help you, Regards, Wojtek - Original Message - From: "Gonzalo Gasca" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, June 07, 2004 9:45 PM Subject: [Asterisk-Users] Mediatrix 1204 > Actually im trying to set up a Mediatrix 1204 to place outgoing calls,i just cand do internal ones, i would like to know if someone could help me with this issue, i declared in sip.conf line1 to line4 for each 1204 port > > SIP.conf > > [100]; My SIP agent > type=friend ; This device takes and makes calls > username=100 ; Username on device > secret=100 ; Password for device > host=dynamic ; This host is not on the same IP addr every time > context=sip ; Inbound calls from this host go here > mailbox=100 ; Activate the message waiting light if this voicemailbox has messages in it > callerid="Gonzalo Gasca" <100> ; Caller ID > > [line1] > type=friend ; This device takes and makes calls > username=line1 ; Username on device > host=110.10.200.10 ; This host is not on the same IP addr every time > context=sip > callerid="Line 1" ; Caller ID > > > extensions.conf > > > [sip] > ignorepat => 9 > exten => _9,1,Dial(SIP/line1) > exten => :9,2,Congestion > > But it just put the box in busy and interchange rtp G711 packets with my client SJphone form sjlabs > I would like a helping hand! > -- > ___ > Get your free email from http://www.hackermail.com > > Powered by Outblaze > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phone with multiple accounts on same instance of asterisk
same here, I 4 extensions from 2 different servers without any problems (Cisco 7960) Wojtek - Original Message - From: "John Fraizer" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, June 04, 2004 1:33 PM Subject: Re: [Asterisk-Users] IP Phone with multiple accounts on same instance of asterisk > Patrick Lidstone (Personal e-mail) wrote: > > > Please excuse me if this is a niaive question... > > > > I have Cisco 7940 (but same applies to Snom's too), and it would be > > convenient to have multiple extensions on the same phone registered > > against the same asterisk instance. (E.g. one extension which is > > associated with work, one extension which is associated with personal > > calls etc). However, when I do this, Asterisk/the phone seems to get > > hopelessly confused - incoming calls do not get routed to the correct > > extension. I think this might be related to the fact that I have a > > single IP address associated with multiple extensions in my SIP.conf. > > > > Is this is known limitation of asterisk? Or am I simply implementing my > > dialplan/routing incorrectly? Suggestions for a workaround which allow > > the separation of work and home contexts gratefully received... > > > > Patrick > > > Patrick, > > I have 6 different extensions, all from the same * server, on my 7960 > and it works just fine. You most likely have a misconfiguration > somewhere that is causing the calls to be routed to the wrong line on > your phone. > > John > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp wont compile.
then run ldconfig or restart your machine...:) W> - Original Message - From: "Sam Bingner" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, May 29, 2004 12:26 AM Subject: RE: [Asterisk-Users] spandsp wont compile. > Add the path to it to /etc/ld.so.conf > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Vlok Stone > Sent: Friday, May 28, 2004 7:14 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] spandsp wont compile. > > > got it to load but now it errors when starting asterisk. complains of no > libspandsp.so.0 and its there. this fax thing is kickin my friggin fax!! > > On Fri, 2004-05-28 at 13:27, Vlok Stone wrote: > > I can't get spandsp to compile. when I go to the */apps directory i > > continually fails. > > Makefile:80: warning: overriding commands for target `app_rxfax.so' > > Makefile:77: warning: ignoring old commands for target `app_rxfax.so' > > cc -fPIC -c -o app_rxfax.o app_rxfax.c > > app_rxfax.c:45: error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' > > undeclared here (not in a function) > > make: *** [app_rxfax.o] Error 1 > > > > I chamged the Makefile to include > > app_rxfax.so : app_rxfax.o > > $(CC) $(SOLINK) -o $@ $< -lspandsp -ltiff > > > > > app_rxfax.so : app_rxfax.c > > gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o > > app_rxfax. o app_rxfax.c > > > > > app_txfax.so : app_txfax.o > > $(CC) $(SOLINK) -o $@ $< -lspandsp -ltiff > > > > > app_txfax.o: app_txfax.c > > gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o > > app_txfax.o app_txfax.c > > > > > > any ideas? > > thanks in advance. > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fax via IAX2
I just implemented fax on my digium cards. So far seems to be working reliable. How about receiving faxes over IAX? I have an account with VoicePulse and would like to be able to get faxes through my incoming number. Anyone got it working? My switch doesn't detect incoming fax and just plays the greeting to the sender :) Also, how can I identify myself to the sender? Is there a variable which I have to set to display my name and number on sender's fax machine during the session? Please advise, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and PostgreSQL
Steven, How reliable is the current build? Do you support mySQL at this point? Thanks, Wojtek - Original Message - From: "Steven Sokol" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, May 27, 2004 12:24 PM Subject: RE: [Asterisk-Users] Asterisk and PostgreSQL > > Hi to all!! > > I'm successful to connect Asterisk to PostgreSQL database... > > If it's possible, can anyone learn me how to store sip user in > > PostgreSQL database and how to configure voicemail?? > > > > Check out the upcoming ast_data extension to Asterisk. It will allow you to > connect to PostgreSQL or any other data source you like. > > http://svn.asteriskdocs.org/res_data > > The current build support Postgresql for IAX, SIP, Extensions and Voicemail. > Zap and other configurations will be added shortly. > > Steven > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_dbmysql and ODBC Voicemail
It won't compile W > > - Original Message - > From: "Mike Machado" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Friday, May 14, 2004 12:43 PM > Subject: [Asterisk-Users] app_dbmysql and ODBC Voicemail > > > > > > I have done a little work on asterisk and database integration. Below is > > a link to app_dbmysql, modeled after Brian's app_dbodbc but for pure > > MySQL. > > > > I also ported the mysql-vm-routines.h to ODBC in case anyone is > > interested. > > > > > > You can get both of these from: > > > > http://www.cheapnet.net/~mike/asterisk > > > > > > They were working as of yesterday CVS, but today CVS will not compile > > and I have not looked into why. Let me know if you have any problems or > > feedback with either of them. > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicetronix's OpenPort4 ANyone?
Thanks Ben.
Wojtek
- Original Message -
From: "Ben Kramer" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, May 12, 2004 9:05 PM
Subject: Re: [Asterisk-Users] Voicetronix's OpenPort4 ANyone?
>
> Hi Wojtek,
>
> you can call a single port like this:
> exten => _9XXX,1,Dial(vpb/1-1/${EXTEN:${TRUNKMSD}})
> Or if you have groups defined in your vpb.conf you could so something
> like this:
> exten => _9XXX,1,Dial(vpb/g1/${EXTEN:${TRUNKMSD}})
>
> Cheers,
>
> Ben.
>
> On Wed, 2004-05-12 at 21:13, Wojciech Tryc wrote:
> > I am looking for opinions and samples on how to call their ports from
the
> > extensions.conf file.
> > Regards,
> > Wojtek
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> --
> Ben Kramer <[EMAIL PROTECTED]>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicetronix's OpenPort4 ANyone?
I am looking for opinions and samples on how to call their ports from the extensions.conf file. Regards, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Pulse and Incoming numbers problem
I just got an account with Voice Pulse and connected to them using IAX2. No problem at all with outgoing calls, however I can not receive any. After further investigation I discovered that the numbers they assinged to me were already in use!!! I am not getting much help from them, their support over e-mail is a joke. Anyone knows a number for their support? Regards, Wojtek
Re: [Asterisk-Users] Mediatrix 1204 (4x FXO)
Their current firmware doesn't allow to write to the section for SIP registration. I am able to communicate with it by dialing [EMAIL PROTECTED]. Also, you have to protect this box with Firewall otherwise the whole world will be able to call through it. Regards, Wojtek - Original Message - From: Dawid Mielnik To: [EMAIL PROTECTED] Sent: Friday, May 07, 2004 9:40 AM Subject: RE: [Asterisk-Users] Mediatrix 1204 (4x FXO) And what problem do you have with registering ? Jeremy Jones has recently posted his SNMP walkthrough from a mediatrix 1104 - you might reference that, configuring 1204 should be very similar to that of 1104. Regards, Dave -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Wojciech TrycSent: Thursday, May 06, 2004 5:27 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Mediatrix 1204 (4x FXO) I have successfully implemented 1204 in semi production environment. Just want to share that it works very well, through the firewall (NATed). Unfortunately, it can not register with the server (and authenticate) but otherwise everything is fine. The audio quality is very good. Regards, Wojtek
Re: [Asterisk-Users] Mediatrix 1204 (4x FXO)
> > Don't know how far you've tried to take the 1204 in terms of functions, > but we did the same thing over a two month period and found: > > 1. handling outbound calls on a "per pstn line" basis (eg, directing > certain calls to certain pstn lines) is very non-standard and subject > to future failures as code changes happen in * and the 1204. Correct, but I don't need to have access on per channel basis. > 2. ring-cadence detect is done on the first ring after the 1204 reboot > and applied to all four ports. If the pstn lines happen to come from > different Central Offices (with slightly different cadences), callerid > and other such timing sensitive functions will fail. I believe that you can actually change that, you have to specify time in ms not a number of rings. > 3. security is less then acceptable. If the 1204 is exposed to the > Internet, anyone can make calls, change settings, etc. Correct, but in real production wouldn't you keep it behind the Firewall? > 4. the per-port cost is substantially higher then many other products > "if" you consider the cost of keeping the firmware reasonably current > as standards evolve. Yes, but SIP connectivity (instead of PCI) adds lots of flexibility > 5. the box does not follow published sip standards; only selected pieces. I am sure that they will release new firmware with better support for SIP > 6. diagnosing problems and monitoring operational functions in a real-world > production environment is less then acceptable. Agreed > 7. support is limited to whatever your reseller provides, which is less > then acceptable if your reseller is not familiar with *. This is reality of the 21st century :) > > We also found the voice quality to be very good, echo cancellation was > good, etc. With relatively easy firmware tweeks to interoperate with * > and standards better, it would be a nice pstn interface; however, they > seem to not have any interest in going there. :) Regards, Wojtek > > Rich > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL and VoiceMail again
To All, I am experiencing very strange behavior. It compiles just fine, on startup I can see that it is connecting and authenticating properly (to mySQL), however it's not using the DB. I can not access any mailboxes while using mySQL module. Can not connect to check for messages, users can not leave me any messages. Do you have any suggestions? Regards, Wojtek - Original Message - From: "Mike Machado" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, May 06, 2004 12:34 AM Subject: Re: [Asterisk-Users] MySQL and VoiceMail again > I have it installed. Its working just fine. > > On Wed, 2004-05-05 at 19:46, Wojciech Tryc wrote: > > Thanks, will try. > > How about the one included in the standard distribution (asterisk-addons)? > > Anyone got it up and running? > > W. > > - Original Message - > > From: "Michael Shuler" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Wednesday, May 05, 2004 3:25 PM > > Subject: RE: [Asterisk-Users] MySQL and VoiceMail again > > > > > > > Use the patches from this site... They work much better > > > > > > http://svn.asteriskdocs.org/res_data/ > > > > > > > > > > > > Michael Shuler, C.E.O. > > > BitWise Systems, Inc. > > > 1301 W. Pioneer Parkway > > > Peoria, IL 61615 > > > Office: (217) 585-0357 > > > Cell: (309) 657-6365 > > > Fax: (309) 213-3500 > > > E-Mail: [EMAIL PROTECTED] > > > Customer Service: (877) 976-0711 > > > > > > > -Original Message- > > > > From: [EMAIL PROTECTED] > > > > [mailto:[EMAIL PROTECTED] On Behalf Of > > > > Wojciech Tryc > > > > Sent: Wednesday, May 05, 2004 1:17 PM > > > > To: [EMAIL PROTECTED] > > > > Subject: [Asterisk-Users] MySQL and VoiceMail again > > > > > > > > > > > > Hi, > > > > At first I would like to express how much I like Asterisk. > > > > Amazing product. > > > > > > > > I compiled Asterisk with mySQL support for CDR and Voicemail. > > > > Everything > > > > seems to be fine, I can see that Asterisk connects to mysql > > > > and logs CDRs. I > > > > can also see that the VOicemail app is also logged in, > > > > however I can not > > > > access any mailboxes. > > > > Similar messages to others, > > > > app_voicemail.c:3011 vm_execmain: Couldn't read username > > > > Can not leave messages, can not check messages... > > > > I have removed the whole section with mailbox definitions from > > > > voicemail.conf > > > > I am running the latest CVS (as of today). > > > > Anyone actually got it to work? > > > > Any help will be greatly appreciated, > > > > Regards, > > > > Wojtek > > > > > > > > ___ > > > > Asterisk-Users mailing list > > > > [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > To UNSUBSCRIBE or update options visit: > > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > -- > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
