Re: [asterisk-users] Mysql identifier not found

[Jonas Kellens]

Hello;

is it normal that connid and resutid have values of 73 and 74 ??

How come this value increases ? What does this mean for values 1 to 70 ??


[May  7 08:52:41] -- Executing [h@sub:10] NoOp("SIP/kal3-024f", 
"") in new stack
[May  7 08:52:41] -- Executing [h@sub:11] NoOp("SIP/kal3-024f", 
"clear MySQL-connections") in new stack
[May  7 08:52:41] -- Executing [h@sub:12] MYSQL("SIP/kal3-024f", 
"Clear 74") in new stack
[May  7 08:52:41] WARNING[15003]: app_mysql.c:194 find_identifier: 
Identifier 74, identifier_type 2 not found in identifier list
[May  7 08:52:41] WARNING[15003]: app_mysql.c:510 aMYSQL_clear: Invalid 
result identifier 74 passed in aMYSQL_clear
[May  7 08:52:41] -- Executing [h@sub:13] MYSQL("SIP/kal3-024f", 
"Disconnect 73") in new stack
[May  7 08:52:41] WARNING[15003]: app_mysql.c:194 find_identifier: 
Identifier 73, identifier_type 1 not found in identifier list
[May  7 08:52:41] WARNING[15003]: app_mysql.c:527 aMYSQL_disconnect: 
Invalid connection identifier 73 passed in aMYSQL_disconnect
[May  7 08:52:41] -- Executing [h@sub:14] NoOp("SIP/kal3-024f", 
"clear MySQL-connections") in new stack
[May  7 08:52:41] -- Executing [h@sub:15] NoOp("SIP/kal3-024f", 
"end") in new stack



There are currently only 8 calls going on...



Kind regards,
Jonas.


 Original Message 
Subject:Re: [asterisk-users] Mysql identifier not found
Date:   Sat, 05 May 2012 12:06:38 +0200
From:   Jonas Kellens 
To: 	Asterisk Users Mailing List - Non-Commercial Discussion 





I ask this because I find the MySQL status information a bit alarming 
(2946 connections) :



mysql> status
--
mysql  Ver 14.12 Distrib 5.0.95, for redhat-linux-gnu (x86_64) using 
readline 5.1


Connection id:2922
Current database:
Current user:root@localhost
SSL:Not in use
Current pager:stdout
Using outfile:''
Using delimiter:;
Server version:5.0.95 Source distribution
Protocol version:10
Connection:Localhost via UNIX socket
Server characterset:latin1
Db characterset:latin1
Client characterset:latin1
Conn.  characterset:latin1
UNIX socket:/var/lib/mysql/mysql.sock
Uptime:6 hours 54 min 31 sec

Threads: 3  Questions: 4919496  Slow queries: 0  Opens: 47  Flush 
tables: 1  Open tables: 41  Queries per second avg: 197.800

--

mysql> show status like 'Conn%';
+---+---+
| Variable_name | Value |
+---+---+
| Connections   | 2946  |
+---+---+
1 row in set (0.00 sec)



Jonas.


On 05/05/2012 11:53 AM, Jonas Kellens wrote:

Hello,

notice in the console output beneath that there is a resultid 6 but it 
can not be cleared :



[May  5 11:46:27] -- Executing [s@sub:3] 
MYSQL("SIP/vart-0336", "Connect connid localhost dialplan host 
Asterisk") in new stack
[May  5 11:46:27] -- Executing [s@sub:4] 
MYSQL("SIP/vart-0336", "Query resultid 4 DELETE FROM pickuptbl 
WHERE pickmark LIKE "%SIP/vart2-0336%"") in new stack
[May  5 11:46:27] -- Executing [s@sub:5] 
MYSQL("SIP/vart-0336", "Clear 6") in new stack
[May  5 11:46:27] WARNING[17803]: app_mysql.c:194 find_identifier: 
Identifier 6, identifier_type 2 not found in identifier list
[May  5 11:46:27] WARNING[17803]: app_mysql.c:510 aMYSQL_clear: 
Invalid result identifier 6 passed in aMYSQL_clear
[May  5 11:46:27] -- Executing [s@sub:6] 
MYSQL("SIP/vart-0336", "Disconnect 4") in new stack
[May  5 11:46:27] -- Executing [s@sub:7] 
Return("SIP/vart-0336", "") in new stack



How come ??


Kind regards,
Jonas.


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Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.

[Steve Edwards]

On Sun, 6 May 2012, Nunya Biznatch wrote:

Thanks for the info. It got me digging deeper. I definitely don't want 
to screw this one up, but I've got to pinch pennies to get this done, so 
don't want to buy anything that would just be nice to have. ...but if I 
have to get it, that's what I'll do.


Aside from capacity, think about maintenance.

If you 'front' your Asterisk servers with Kamailio running on 2 servers 
(even if these servers are also your Asterisk servers) you have the 
ability to take an Asterisk server out of production just by reconfiguring 
Kamailio and waiting the calls in progress to finish.


Then you can install patches, replace failing disks, etc, etc, etc.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.

[Nunya Biznatch]

Thanks for the info. It got me digging deeper. I definitely don't want 
to screw this one up, but I've got to pinch pennies to get this done, so 
don't want to buy anything that would just be nice to have. ...but if I 
have to get it, that's what I'll do.


Have any of you seen this? 
ftp://download.intel.com/design/intarch/PAPERS/318862.pdf


It's a whitepaper from Intel where they load tested Asterisk on various 
Intel Processors. They were trying to show the benefit of compiling 
Asterisk using their compiler vs. gcc. It's from January 2008. They used 
Astertest as the test base. With a dual Xeon 5335 @ 2GHz (dual quad 
cores), and using a gcc compiled Asterisk, they were able to process 673 
concurrent calls with GSM to iLBC transcoding and 552 calls with GSM to 
Speex transcoding.


Looking at http://cpubenchmark.net, I see a dual Xeon 5335 @ 2GHz has a 
Passmark score of 5,095. A more modern single E5-2630 processor has more 
than double the score at 10,401.


...and those results were with whatever version of Asterisk was out and 
about in January 2008. Would it be 1.4? From what I read here 
http://www.voip-info.org/wiki/view/Asterisk+dimensioning, Asterisk 1.6 
is 3-4 times better in performance than 1.4, and 1.10 is 2-3 times 
faster than 1.8.


Also, keeping in mind while yes I have 800 SIP phones, only 200 will be 
active concurrently at peak times based on current call traffic data, 
and I'm adding 50% to cover myself and looking to build to support 300 
concurrent calls. Finally, throw in the fact the main Asterisk Server 
will not be doing any transcoding. The only transcoding will be in the 
PRI Gateway server, and with 3 PRI's, I only need the power to transcode 
69 concurrent calls from G.711 to G.722.


The next concern is the raw number of actively registered phones. I 
guess this is something I don't understand what the repercussions are, 
and I know the unknown is always what bites you. What happens? I 
wouldn't think that's a lot of open port traffic to worry about?



Thanks Again?


On 5/6/2012 3:19 PM, Mitul Limbani wrote:


For 100% High Availibility and Hot Failover, I would recommend one of 
those Red-fone Fonebridges.


Also getting 800 Phones all register on single server is crazy, add a 
SIP proxy to distribute load evenly between 2 Ast boxes.


For Wireless you might consider using DECT phones from Snom instead of 
std 802.11 based wifi phones. Giving QoS on wifi is a big pain.


Hope that helps,

Regards,
Mitul Limbani
Enterux Solutions

On May 6, 2012 11:34 PM, "Nunya Biznatch" > wrote:


I'm about to receive approval to design and deploy an
Asterisk-based phone system for my company. I will immediately
have to start writing specifications. I'm working on the hardware
design and the architecture right now. I'd like a second, third,
fourth, 1,000th opinion.

800 SIP phones. All will be G.722. I expect 200 concurrent calls,
with 20% leaving to the outside world. There will be another 200
analog lines that will for the time being remain on the TDM PBX
switch they reside on, and will be whittled down and converted to
SIP as time and attrition allows. These are primarily fax machines
and conference "spider" phones. Those are included in my 200
concurrent calls number. I'm looking to get as close to 5-9's
reliability as I can, with 4-9's mandatory. Proper power filtering
and backup is already available.


Here's what I'm thinking for the architecture:

Server 1: PRI Gateway 1 - Support 2 outside PRI trunks for local
and long distance, plus a third PRI connecting to the existing TDM
PBX.

Server 2: PRI Gateway 2 - Support 1 PRI trunk for local and long
distance with room for another, plus a second PRI connecting to
the existing TDM PBX.

Reason for two PRI Gateways is for redundancy and fail-over, but
processor capabilities is a concern. I expect in about two years
I'll be ready to decommission the TDM PBX, but will be left with
about 80 Analog lines across the multiple buildings on my campus.
I expect I'll end up purchasing channel banks to support the
remaining analog lines, and distribute across the campus using
existing copper plant.


Server 3: Asterisk Master Server

Server 4: Asterisk Slave Server

I'm considering a clustered environment, but I believe a fail-over
solution would be easier to implement in the short term. This
means each system needs to handle all traffic by itself. These
servers will be used for Asterisk and Voice-mail. Conferencing
will be enabled, but I'm not considering it in the build. If I see
conferencing becoming a factor, I will build another server and
offload that service.


Server 5: Boot Server - DHCP, RADIUS, SNTP, DNS, LDAP, FTP, HTTPS,
SNMP, etc...

This service will provide the phone network all the basic
services. This is a stand-alone phone netw

Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.

[Antonio Goméz Soto]

I have deployed multiple installations with 1500 phones per server using 
standard HP DL-380's.
It's not that crazy. They can handle it pretty easily.

Antonio.




Op 06-05-12 23:19, Mitul Limbani schreef:
> For 100% High Availibility and Hot Failover, I would recommend one of those 
> Red-fone Fonebridges.
> 
> Also getting 800 Phones all register on single server is crazy, add a SIP 
> proxy to distribute load evenly between 2 Ast boxes.
> 
> For Wireless you might consider using DECT phones from Snom instead of std 
> 802.11 based wifi phones. Giving QoS on wifi is a big pain.
> 
> Hope that helps,
> 
> Regards,
> Mitul Limbani
> Enterux Solutions
> 
> On May 6, 2012 11:34 PM, "Nunya Biznatch"  > wrote:
> 
> I'm about to receive approval to design and deploy an Asterisk-based 
> phone system for my company. I will immediately have to start writing 
> specifications. I'm working on the hardware design and the architecture right 
> now. I'd like a second, third,
> fourth, 1,000th opinion.
> 
> 800 SIP phones. All will be G.722. I expect 200 concurrent calls, with 
> 20% leaving to the outside world. There will be another 200 analog lines that 
> will for the time being remain on the TDM PBX switch they reside on, and will 
> be whittled down and
> converted to SIP as time and attrition allows. These are primarily fax 
> machines and conference "spider" phones. Those are included in my 200 
> concurrent calls number. I'm looking to get as close to 5-9's reliability as 
> I can, with 4-9's mandatory.
> Proper power filtering and backup is already available.
> 
> 
> Here's what I'm thinking for the architecture:
> 
> Server 1: PRI Gateway 1 - Support 2 outside PRI trunks for local and long 
> distance, plus a third PRI connecting to the existing TDM PBX.
> 
> Server 2: PRI Gateway 2 - Support 1 PRI trunk for local and long distance 
> with room for another, plus a second PRI connecting to the existing TDM PBX.
> 
> Reason for two PRI Gateways is for redundancy and fail-over, but 
> processor capabilities is a concern. I expect in about two years I'll be 
> ready to decommission the TDM PBX, but will be left with about 80 Analog 
> lines across the multiple buildings on
> my campus. I expect I'll end up purchasing channel banks to support the 
> remaining analog lines, and distribute across the campus using existing 
> copper plant.
> 
> 
> Server 3: Asterisk Master Server
> 
> Server 4: Asterisk Slave Server
> 
> I'm considering a clustered environment, but I believe a fail-over 
> solution would be easier to implement in the short term. This means each 
> system needs to handle all traffic by itself. These servers will be used for 
> Asterisk and Voice-mail.
> Conferencing will be enabled, but I'm not considering it in the build. If 
> I see conferencing becoming a factor, I will build another server and offload 
> that service.
> 
> 
> Server 5: Boot Server - DHCP, RADIUS, SNTP, DNS, LDAP, FTP, HTTPS, SNMP, 
> etc...
> 
> This service will provide the phone network all the basic services. This 
> is a stand-alone phone network primarily because it would be too costly to 
> upgrade the entire data network to support both voice and data. The phone 
> network will not initially
> have Internet Access. This server will be the server all the phones talk 
> to for pulling their configs.
> 
> I'm considering a second Boot Server for redundancy, but since the phones 
> should store their configs, I'm not seeing this as horribly critical. Am I 
> smoking something?
> 
> 
> Finally, I'll have a Windows-based workstation that will be used to 
> remote into all the services, for administration, etc...
> 
> I need to plan to use FreePBX on all Asterisk Servers, but I don't intend 
> to install it until I'm in regular MAC maintenance mode.
> 
> I have no plans at this time to build out any databases. I just plan to 
> use whatever Asterisk has. If it ever comes to that, I would make those 
> separate servers as well.
> 
> My goal is to build Asterisk Servers and PRI Gateways capable of 
> supporting 150% of what I anticipate, which would come out to 300 concurrent 
> calls. Again, all phones will use G.722. The PRI Gateway servers will do the 
> heavy lifting of converting
> G.711 traffic from the PRIs to G722, and connect to the Asterisk Servers 
> via IAX2 trunk.
> 
> It's my intention to build each server myself with high-quality off the 
> shelf components. I'd like all servers to be as close to identical as 
> possible, as I intend to keep spares on hand to facilitate quick repair and 
> minimize downtime. I'm
> considering RAID 1 + 0 (mirrored and stripped drives) for all servers. I 
> am considering dual redundant power supplies.
> 
> For a processor, I'm currently looking at the i7-3770K @ 3.5GHz or very 
> similar. Its Passmark compares to the Xeon E5-2630 @ 2.3GHz, but is ha

Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.

[Mitul Limbani]

For 100% High Availibility and Hot Failover, I would recommend one of those
Red-fone Fonebridges.

Also getting 800 Phones all register on single server is crazy, add a SIP
proxy to distribute load evenly between 2 Ast boxes.

For Wireless you might consider using DECT phones from Snom instead of std
802.11 based wifi phones. Giving QoS on wifi is a big pain.

Hope that helps,

Regards,
Mitul Limbani
Enterux Solutions
On May 6, 2012 11:34 PM, "Nunya Biznatch"  wrote:

>  I'm about to receive approval to design and deploy an Asterisk-based
> phone system for my company. I will immediately have to start writing
> specifications. I'm working on the hardware design and the architecture
> right now. I'd like a second, third, fourth, 1,000th opinion.
>
> 800 SIP phones. All will be G.722. I expect 200 concurrent calls, with 20%
> leaving to the outside world. There will be another 200 analog lines that
> will for the time being remain on the TDM PBX switch they reside on, and
> will be whittled down and converted to SIP as time and attrition allows.
> These are primarily fax machines and conference "spider" phones. Those are
> included in my 200 concurrent calls number. I'm looking to get as close to
> 5-9's reliability as I can, with 4-9's mandatory. Proper power filtering
> and backup is already available.
>
>
>  Here's what I'm thinking for the architecture:
>
> Server 1: PRI Gateway 1 - Support 2 outside PRI trunks for local and long
> distance, plus a third PRI connecting to the existing TDM PBX.
>
> Server 2: PRI Gateway 2 - Support 1 PRI trunk for local and long distance
> with room for another, plus a second PRI connecting to the existing TDM PBX.
>
> Reason for two PRI Gateways is for redundancy and fail-over, but processor
> capabilities is a concern. I expect in about two years I'll be ready to
> decommission the TDM PBX, but will be left with about 80 Analog lines
> across the multiple buildings on my campus. I expect I'll end up purchasing
> channel banks to support the remaining analog lines, and distribute across
> the campus using existing copper plant.
>
>
>  Server 3: Asterisk Master Server
>
> Server 4: Asterisk Slave Server
>
> I'm considering a clustered environment, but I believe a fail-over
> solution would be easier to implement in the short term. This means each
> system needs to handle all traffic by itself. These servers will be used
> for Asterisk and Voice-mail. Conferencing will be enabled, but I'm not
> considering it in the build. If I see conferencing becoming a factor, I
> will build another server and offload that service.
>
>
>  Server 5: Boot Server - DHCP, RADIUS, SNTP, DNS, LDAP, FTP, HTTPS, SNMP,
> etc...
>
> This service will provide the phone network all the basic services. This
> is a stand-alone phone network primarily because it would be too costly to
> upgrade the entire data network to support both voice and data. The phone
> network will not initially have Internet Access. This server will be the
> server all the phones talk to for pulling their configs.
>
> I'm considering a second Boot Server for redundancy, but since the phones
> should store their configs, I'm not seeing this as horribly critical. Am I
> smoking something?
>
>
> Finally, I'll have a Windows-based workstation that will be used to remote
> into all the services, for administration, etc...
>
> I need to plan to use FreePBX on all Asterisk Servers, but I don't intend
> to install it until I'm in regular MAC maintenance mode.
>
> I have no plans at this time to build out any databases. I just plan to
> use whatever Asterisk has. If it ever comes to that, I would make those
> separate servers as well.
>
> My goal is to build Asterisk Servers and PRI Gateways capable of
> supporting 150% of what I anticipate, which would come out to 300
> concurrent calls. Again, all phones will use G.722. The PRI Gateway servers
> will do the heavy lifting of converting G.711 traffic from the PRIs to
> G722, and connect to the Asterisk Servers via IAX2 trunk.
>
> It's my intention to build each server myself with high-quality off the
> shelf components. I'd like all servers to be as close to identical as
> possible, as I intend to keep spares on hand to facilitate quick repair and
> minimize downtime. I'm considering RAID 1 + 0 (mirrored and stripped
> drives) for all servers. I am considering dual redundant power supplies.
>
> For a processor, I'm currently looking at the i7-3770K @ 3.5GHz or very
> similar. Its Passmark compares to the Xeon E5-2630 @ 2.3GHz, but is half
> the price.
>
> I have no idea what amount of memory to consider, so I am thinking 8GB per
> machine.
>
> PCI-E is what I plan for all the cards.
>
> Debian is the Linux flavor
>
> A new network will be deployed using PoE layer-2 managed switches. Battery
> backup capable of providing 8 hours will be installed as required. There
> will be multiple VLANs in the network as I have multiple dissimilar offices
> I need to keep separated from each other.

Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.

[Paul Belanger]

On 12-05-06 02:00 PM, Nunya Biznatch wrote:

I'm about to receive approval to design and deploy an Asterisk-based
phone system for my company. I will immediately have to start writing
specifications. I'm working on the hardware design and the architecture
right now. I'd like a second, third, fourth, 1,000th opinion.

800 SIP phones. All will be G.722. I expect 200 concurrent calls, with
20% leaving to the outside world. There will be another 200 analog lines
that will for the time being remain on the TDM PBX switch they reside
on, and will be whittled down and converted to SIP as time and attrition
allows. These are primarily fax machines and conference "spider" phones.
Those are included in my 200 concurrent calls number. I'm looking to get
as close to 5-9's reliability as I can, with 4-9's mandatory. Proper
power filtering and backup is already available.


Here's what I'm thinking for the architecture:

Server 1: PRI Gateway 1 - Support 2 outside PRI trunks for local and
long distance, plus a third PRI connecting to the existing TDM PBX.

Server 2: PRI Gateway 2 - Support 1 PRI trunk for local and long
distance with room for another, plus a second PRI connecting to the
existing TDM PBX.

Reason for two PRI Gateways is for redundancy and fail-over, but
processor capabilities is a concern. I expect in about two years I'll be
ready to decommission the TDM PBX, but will be left with about 80 Analog
lines across the multiple buildings on my campus. I expect I'll end up
purchasing channel banks to support the remaining analog lines, and
distribute across the campus using existing copper plant.


Server 3: Asterisk Master Server

Server 4: Asterisk Slave Server

I'm considering a clustered environment, but I believe a fail-over
solution would be easier to implement in the short term. This means each
system needs to handle all traffic by itself. These servers will be used
for Asterisk and Voice-mail. Conferencing will be enabled, but I'm not
considering it in the build. If I see conferencing becoming a factor, I
will build another server and offload that service.

800 SIP phones on one server? I wouldn't want to do it. Add a SIP proxy 
to your design and have it handle all your SIP.  Then you can load 
balance across multiple asterisk boxes.  You'll be thankful you did this 
at the start, as it will allow you to increase resources more easily.




Server 5: Boot Server - DHCP, RADIUS, SNTP, DNS, LDAP, FTP, HTTPS, SNMP,
etc...

This service will provide the phone network all the basic services. This
is a stand-alone phone network primarily because it would be too costly
to upgrade the entire data network to support both voice and data. The
phone network will not initially have Internet Access. This server will
be the server all the phones talk to for pulling their configs.

I'm considering a second Boot Server for redundancy, but since the
phones should store their configs, I'm not seeing this as horribly
critical. Am I smoking something?

Finally, I'll have a Windows-based workstation that will be used to
remote into all the services, for administration, etc...


Why?


I need to plan to use FreePBX on all Asterisk Servers, but I don't
intend to install it until I'm in regular MAC maintenance mode.

It is ashame you are going this far with your setup to rely on FreePBX. 
 For something this complex, you are setting your self up for some 
heartache.



I have no plans at this time to build out any databases. I just plan to
use whatever Asterisk has. If it ever comes to that, I would make those
separate servers as well.

My goal is to build Asterisk Servers and PRI Gateways capable of
supporting 150% of what I anticipate, which would come out to 300
concurrent calls. Again, all phones will use G.722. The PRI Gateway
servers will do the heavy lifting of converting G.711 traffic from the
PRIs to G722, and connect to the Asterisk Servers via IAX2 trunk.

It's my intention to build each server myself with high-quality off the
shelf components. I'd like all servers to be as close to identical as
possible, as I intend to keep spares on hand to facilitate quick repair
and minimize downtime. I'm considering RAID 1 + 0 (mirrored and stripped
drives) for all servers. I am considering dual redundant power supplies.

For a processor, I'm currently looking at the i7-3770K @ 3.5GHz or very
similar. Its Passmark compares to the Xeon E5-2630 @ 2.3GHz, but is half
the price.

I have no idea what amount of memory to consider, so I am thinking 8GB
per machine.

PCI-E is what I plan for all the cards.

Debian is the Linux flavor

A new network will be deployed using PoE layer-2 managed switches.
Battery backup capable of providing 8 hours will be installed as
required. There will be multiple VLANs in the network as I have multiple
dissimilar offices I need to keep separated from each other. We will
also have 802.11 SIP phones, and will be deploying a campus-wide WiFi
network used only by the phone system. Yes, I crunched the number

[asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.

[Nunya Biznatch]

I'm about to receive approval to design and deploy an Asterisk-based 
phone system for my company. I will immediately have to start writing 
specifications. I'm working on the hardware design and the architecture 
right now. I'd like a second, third, fourth, 1,000th opinion.


800 SIP phones. All will be G.722. I expect 200 concurrent calls, with 
20% leaving to the outside world. There will be another 200 analog lines 
that will for the time being remain on the TDM PBX switch they reside 
on, and will be whittled down and converted to SIP as time and attrition 
allows. These are primarily fax machines and conference "spider" phones. 
Those are included in my 200 concurrent calls number. I'm looking to get 
as close to 5-9's reliability as I can, with 4-9's mandatory. Proper 
power filtering and backup is already available.



Here's what I'm thinking for the architecture:

Server 1: PRI Gateway 1 - Support 2 outside PRI trunks for local and 
long distance, plus a third PRI connecting to the existing TDM PBX.


Server 2: PRI Gateway 2 - Support 1 PRI trunk for local and long 
distance with room for another, plus a second PRI connecting to the 
existing TDM PBX.


Reason for two PRI Gateways is for redundancy and fail-over, but 
processor capabilities is a concern. I expect in about two years I'll be 
ready to decommission the TDM PBX, but will be left with about 80 Analog 
lines across the multiple buildings on my campus. I expect I'll end up 
purchasing channel banks to support the remaining analog lines, and 
distribute across the campus using existing copper plant.



Server 3: Asterisk Master Server

Server 4: Asterisk Slave Server

I'm considering a clustered environment, but I believe a fail-over 
solution would be easier to implement in the short term. This means each 
system needs to handle all traffic by itself. These servers will be used 
for Asterisk and Voice-mail. Conferencing will be enabled, but I'm not 
considering it in the build. If I see conferencing becoming a factor, I 
will build another server and offload that service.



Server 5: Boot Server - DHCP, RADIUS, SNTP, DNS, LDAP, FTP, HTTPS, SNMP, 
etc...


This service will provide the phone network all the basic services. This 
is a stand-alone phone network primarily because it would be too costly 
to upgrade the entire data network to support both voice and data. The 
phone network will not initially have Internet Access. This server will 
be the server all the phones talk to for pulling their configs.


I'm considering a second Boot Server for redundancy, but since the 
phones should store their configs, I'm not seeing this as horribly 
critical. Am I smoking something?



Finally, I'll have a Windows-based workstation that will be used to 
remote into all the services, for administration, etc...


I need to plan to use FreePBX on all Asterisk Servers, but I don't 
intend to install it until I'm in regular MAC maintenance mode.


I have no plans at this time to build out any databases. I just plan to 
use whatever Asterisk has. If it ever comes to that, I would make those 
separate servers as well.


My goal is to build Asterisk Servers and PRI Gateways capable of 
supporting 150% of what I anticipate, which would come out to 300 
concurrent calls. Again, all phones will use G.722. The PRI Gateway 
servers will do the heavy lifting of converting G.711 traffic from the 
PRIs to G722, and connect to the Asterisk Servers via IAX2 trunk.


It's my intention to build each server myself with high-quality off the 
shelf components. I'd like all servers to be as close to identical as 
possible, as I intend to keep spares on hand to facilitate quick repair 
and minimize downtime. I'm considering RAID 1 + 0 (mirrored and stripped 
drives) for all servers. I am considering dual redundant power supplies.


For a processor, I'm currently looking at the i7-3770K @ 3.5GHz or very 
similar. Its Passmark compares to the Xeon E5-2630 @ 2.3GHz, but is half 
the price.


I have no idea what amount of memory to consider, so I am thinking 8GB 
per machine.


PCI-E is what I plan for all the cards.

Debian is the Linux flavor

A new network will be deployed using PoE layer-2 managed switches. 
Battery backup capable of providing 8 hours will be installed as 
required. There will be multiple VLANs in the network as I have multiple 
dissimilar offices I need to keep separated from each other. We will 
also have 802.11 SIP phones, and will be deploying a campus-wide WiFi 
network used only by the phone system. Yes, I crunched the numbers. This 
will be significantly cheaper than upgrading the entire existing data 
network to support the new phone system. ...and to be quite honest, I 
don't trust our network folks, and know adding that layer of bureaucracy 
will only negatively impact the customer experience. I was a network 
engineer for a top-three telecom company for many years, so I do have a 
point of reference to make those statements.


...yes, I am on

[asterisk-users] fake auth rejection??

[Ira]

Very occasionally in my logs I see things like this. In this case 7 
lines starting with the first line and each ending with one of the 
group of 7. Took about 10 seconds for the 7 tries.


[2012-05-03 16:58:27] NOTICE[31850] chan_sip.c: Sending fake auth 
rejection for device "unknown"


;tag=aTZ1eFu5Gi
;tag=MQ7X2xPoIy
;tag=FnfiJEymn6
;tag=tVPso6QAEp
;tag=tWHOjRp11z
;tag=DK42h3mLn1
;tag=CPW3Z9lDvN

Should I be worried?

Ira


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Re: [asterisk-users] TDM400P: Lifetime & Replacement

[Greg Woods]

On Sun, 2012-05-06 at 12:46 -0400, Andrew Latham wrote:

> 
> Sounds like you did a kernel update and did not rebuild DAHDI.

I haven't done a kernel update on this particular machine in quite some
time, since long before the card started failing. It is still running
Fedora 14, so there aren't even any kernel updates available for it.

--Greg



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Re: [asterisk-users] Why SendDTMF is not working?

[Shahid H]

Hey guys,

I have managed to get to work  Thanks for the help..

I just registered a new account at sipgate.co.uk and test it on asterisk...
and DTMF worked well :)

It seem voip.ms dont work well when sending DTMF to UK.

Do anyone know UK/Europe voip provider to allow you change any callerID as
you like without validation?

I know voip.ms does it and sipgate don't allow it.

Thanks!

On Sun, May 6, 2012 at 5:08 PM, Shahid H  wrote:

> Here is another debug log:
>
>  == Using SIP RTP CoS mark 5
> -- Executing [123@test2:1] Dial("SIP/test2-0008",
> "SIP/+44776@voipms,,D(1ww2ww3ww4)") in new stack
>   == Using SIP RTP CoS mark 5
> -- Called SIP/+44776XX@voipms
> -- SIP/voipms-0009 is making progress passing it to
> SIP/test2-0008
> -- SIP/voipms-0009 answered SIP/test2-0008
> -- Sending DTMF '1ww2ww3ww4' to the called party.
> -- Locally bridging SIP/test2-0008 and SIP/voipms-0009
>
> When DTMF is finish then "Locally bridging" is executed...
>
> On the softphone it say "State: Early Media" while it sending DTMF even
> though I cant hear DTMF sound.. after 10 seconds State changed to "Up" (I
> can hear talking to myself).
>
>
>
> On Sun, May 6, 2012 at 4:18 PM, Shahid H  wrote:
>
>> When I changed back to dtmfmode=rfc2833 and I cant hear the DTMF
>> sound.. completely silent.
>>
>> Indeed I have put disallow=all before the allow=ulaw  allow=alaw
>>
>> "sip show channels" in the CLI  show during a call:
>>
>> 78.129.xxx.xx   +447715d909406db14d2  0x4 (ulaw)   No
>>   Tx: ACK
>> 94.192.xxx.xx   test  MTNlNGNkYjlhODA  0x4 (ulaw)
>>   No   Rx: ACK
>>
>> Still no luck to get DTMF to work :(
>>
>> Thanks
>> Shahid
>>
>>
>> On Sun, May 6, 2012 at 2:54 PM, Eric Wieling  wrote:
>>
>>> Now you have a totally different issue.  8-)
>>>
>>> While the call is up do a "sip show channels" in the CLI.  This will
>>> show you the ACTUAL codec for the call.  Likely the call was still using
>>> GSM.  Did you remember to put a disallow=all before the allow= lines?
>>>
>>> I recommend dtmfmode=rfc2833 with whatever codec you want to use.
>>> Inband DTMF will sound broken and distorted if it is sent over most codecs.
>>>
>>>
>>> -Original Message-
>>> From: asterisk-users-boun...@lists.digium.com [mailto:
>>> asterisk-users-boun...@lists.digium.com] On Behalf Of Shahid H
>>> Sent: Sunday, May 06, 2012 9:16 AM
>>> To: Markus
>>> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: Re: [asterisk-users] Why SendDTMF is not working?
>>>
>>> Thanks for the suggestion Markus. Here what I did:
>>>
>>> In the logger.config I have added 'dtmf':
>>>
>>> console => notice,warning,error,dtmf
>>>
>>> and then in sip.conf:
>>>
>>> allow=ulaw
>>> allow=alaw
>>> ; allow=gsm
>>> dtmfmode=inband
>>>
>>> I've added a test to call my mobile:
>>>
>>> exten => 123,1,Dial(SIP/+4477XXX@voipms,,D(1ww2ww3ww4))
>>> exten => 123,n,Hangup()
>>>
>>> then restarted asterisk and logged into console (asterisk -r)
>>>
>>> I've call my mobile using softphone, I did not see 1,2,3,4 digits being
>>> sent on the console but I can hear broken/unclear DTMF on the mobile...
>>>
>>> however when I press digits on the softphone I can hear DTMF clear how
>>> it should be on my mobile and on the console it is showing DTMF:
>>>
>>> astrisk*CLI> [May  6 14:13:06] DTMF[28559]: channel.c:3082 __ast_read:
>>> DTMF begin '4' received on SIP/test-001c [May  6 14:13:06] DTMF[28559]:
>>> channel.c:3092 __ast_read: DTMF begin passthrough '4' on SIP/test-001c
>>> [May  6 14:13:06] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '4'
>>> received on SIP/test-001c, duration 120 ms [May  6 14:13:06]
>>> DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '4' on
>>> SIP/test-001c [May  6 14:13:06] DTMF[28559]: channel.c:3066 __ast_read:
>>> DTMF end passthrough '4' on SIP/test-001c [May  6 14:13:07]
>>> DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '5' received on
>>> SIP/test-001c [May  6 14:13:07] DTMF[28559]: channel.c:3092 __ast_read:
>>> DTMF begin passthrough '5' on SIP/test-001c [May  6 14:13:07]
>>> DTMF[28559]: channel.c:2997 __ast_read: DTMF end '5' received on
>>> SIP/test-001c, duration 120 ms [May  6 14:13:07] DTMF[28559]:
>>> channel.c:3037 __ast_read: DTMF end accepted with begin '5' on
>>> SIP/test-001c [May  6 14:13:07] DTMF[28559]: channel.c:3066 __ast_read:
>>> DTMF end passthrough '5' on SIP/test-001c [May  6 14:13:08]
>>> DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '6' received on
>>> SIP/test-001c [May  6 14:13:08] DTMF[28559]: channel.c:3092 __ast_read:
>>> DTMF begin passthrough '6' on SIP/test-001c [May  6 14:13:08]
>>> DTMF[28559]: channel.c:2997 __ast_read: DTMF end '6' received on
>>> SIP/test-001c, duration 120 ms [May  6 14:13:08] DTMF[28559]:
>>> channel.c:3037 __ast_read: DTMF end accepted with begin '6' on
>>> S

Re: [asterisk-users] TDM400P: Lifetime & Replacement

[Andrew Latham]

On Sun, May 6, 2012 at 12:42 PM, Greg Woods  wrote:
> I have a Digium TDM400P card that appears to have died. The first noted
> symptoms were that dahdi would fail to reload on boot. On closer
> inspection, the card looks totally dead; no lights on at all. I have
> tried moving it to a different PCI slot, and removing the other PCI card
> (a 3com 10/100 NIC) completely.  I have not tried removing the PCI-E
> graphics card, of course, because I can't boot the system without it,
> but that is unlikely to be fruitful anyway.
>
> So the questions are: first, what is the expected lifetime of one of
> these cards? It just passed its 5th birthday. Is that as long as it
> could be expected to last?
>
> Second, since the parts of this card are very expensive, I am wondering
> if these symptoms likely mean that the main board of the card is dead,
> but the FXS and FXO modules might still be good. In that case, I could
> just get a new main card and move the modules to the new main card. The
> problem is that I can't find any TDM400P cards anywhere, all I can find
> are TDM410P's. Will the modules I have (assuming they are still good)
> work with a TDM410P?
>
> Last question: the TDM410 card is available in PCI and PCIx1 forms. I do
> have a free PCIx1 slot. Is there any advantage in one over the other?
>
> Thanks,
> --Greg

Sounds like you did a kernel update and did not rebuild DAHDI.

I have never witnessed a failed Tormenta or Digium card.  I have read
about them.

-- 
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Re: [asterisk-users] TDM400P: Lifetime & Replacement

[Kevin P. Fleming]

On 05/06/2012 11:42 AM, Greg Woods wrote:

Second, since the parts of this card are very expensive, I am wondering
if these symptoms likely mean that the main board of the card is dead,
but the FXS and FXO modules might still be good. In that case, I could
just get a new main card and move the modules to the new main card. The
problem is that I can't find any TDM400P cards anywhere, all I can find
are TDM410P's. Will the modules I have (assuming they are still good)
work with a TDM410P?


Yes, the module are compatible with a TDM410P, or any other Digium card 
supporting analog modules except the TDM2400P.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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[asterisk-users] TDM400P: Lifetime & Replacement

[Greg Woods]

I have a Digium TDM400P card that appears to have died. The first noted
symptoms were that dahdi would fail to reload on boot. On closer
inspection, the card looks totally dead; no lights on at all. I have
tried moving it to a different PCI slot, and removing the other PCI card
(a 3com 10/100 NIC) completely.  I have not tried removing the PCI-E
graphics card, of course, because I can't boot the system without it,
but that is unlikely to be fruitful anyway.

So the questions are: first, what is the expected lifetime of one of
these cards? It just passed its 5th birthday. Is that as long as it
could be expected to last?

Second, since the parts of this card are very expensive, I am wondering
if these symptoms likely mean that the main board of the card is dead,
but the FXS and FXO modules might still be good. In that case, I could
just get a new main card and move the modules to the new main card. The
problem is that I can't find any TDM400P cards anywhere, all I can find
are TDM410P's. Will the modules I have (assuming they are still good)
work with a TDM410P?

Last question: the TDM410 card is available in PCI and PCIx1 forms. I do
have a free PCIx1 slot. Is there any advantage in one over the other?

Thanks,
--Greg



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Re: [asterisk-users] Why SendDTMF is not working?

[Shahid H]

Here is another debug log:

 == Using SIP RTP CoS mark 5
-- Executing [123@test2:1] Dial("SIP/test2-0008",
"SIP/+44776@voipms,,D(1ww2ww3ww4)") in new stack
  == Using SIP RTP CoS mark 5
-- Called SIP/+44776XX@voipms
-- SIP/voipms-0009 is making progress passing it to
SIP/test2-0008
-- SIP/voipms-0009 answered SIP/test2-0008
-- Sending DTMF '1ww2ww3ww4' to the called party.
-- Locally bridging SIP/test2-0008 and SIP/voipms-0009

When DTMF is finish then "Locally bridging" is executed...

On the softphone it say "State: Early Media" while it sending DTMF even
though I cant hear DTMF sound.. after 10 seconds State changed to "Up" (I
can hear talking to myself).



On Sun, May 6, 2012 at 4:18 PM, Shahid H  wrote:

> When I changed back to dtmfmode=rfc2833 and I cant hear the DTMF
> sound.. completely silent.
>
> Indeed I have put disallow=all before the allow=ulaw  allow=alaw
>
> "sip show channels" in the CLI  show during a call:
>
> 78.129.xxx.xx   +447715d909406db14d2  0x4 (ulaw)   No
>   Tx: ACK
> 94.192.xxx.xx   test  MTNlNGNkYjlhODA  0x4 (ulaw)
>   No   Rx: ACK
>
> Still no luck to get DTMF to work :(
>
> Thanks
> Shahid
>
>
> On Sun, May 6, 2012 at 2:54 PM, Eric Wieling  wrote:
>
>> Now you have a totally different issue.  8-)
>>
>> While the call is up do a "sip show channels" in the CLI.  This will show
>> you the ACTUAL codec for the call.  Likely the call was still using GSM.
>>  Did you remember to put a disallow=all before the allow= lines?
>>
>> I recommend dtmfmode=rfc2833 with whatever codec you want to use.
>> Inband DTMF will sound broken and distorted if it is sent over most codecs.
>>
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] On Behalf Of Shahid H
>> Sent: Sunday, May 06, 2012 9:16 AM
>> To: Markus
>> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Why SendDTMF is not working?
>>
>> Thanks for the suggestion Markus. Here what I did:
>>
>> In the logger.config I have added 'dtmf':
>>
>> console => notice,warning,error,dtmf
>>
>> and then in sip.conf:
>>
>> allow=ulaw
>> allow=alaw
>> ; allow=gsm
>> dtmfmode=inband
>>
>> I've added a test to call my mobile:
>>
>> exten => 123,1,Dial(SIP/+4477XXX@voipms,,D(1ww2ww3ww4))
>> exten => 123,n,Hangup()
>>
>> then restarted asterisk and logged into console (asterisk -r)
>>
>> I've call my mobile using softphone, I did not see 1,2,3,4 digits being
>> sent on the console but I can hear broken/unclear DTMF on the mobile...
>>
>> however when I press digits on the softphone I can hear DTMF clear how it
>> should be on my mobile and on the console it is showing DTMF:
>>
>> astrisk*CLI> [May  6 14:13:06] DTMF[28559]: channel.c:3082 __ast_read:
>> DTMF begin '4' received on SIP/test-001c [May  6 14:13:06] DTMF[28559]:
>> channel.c:3092 __ast_read: DTMF begin passthrough '4' on SIP/test-001c
>> [May  6 14:13:06] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '4'
>> received on SIP/test-001c, duration 120 ms [May  6 14:13:06]
>> DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '4' on
>> SIP/test-001c [May  6 14:13:06] DTMF[28559]: channel.c:3066 __ast_read:
>> DTMF end passthrough '4' on SIP/test-001c [May  6 14:13:07]
>> DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '5' received on
>> SIP/test-001c [May  6 14:13:07] DTMF[28559]: channel.c:3092 __ast_read:
>> DTMF begin passthrough '5' on SIP/test-001c [May  6 14:13:07]
>> DTMF[28559]: channel.c:2997 __ast_read: DTMF end '5' received on
>> SIP/test-001c, duration 120 ms [May  6 14:13:07] DTMF[28559]:
>> channel.c:3037 __ast_read: DTMF end accepted with begin '5' on
>> SIP/test-001c [May  6 14:13:07] DTMF[28559]: channel.c:3066 __ast_read:
>> DTMF end passthrough '5' on SIP/test-001c [May  6 14:13:08]
>> DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '6' received on
>> SIP/test-001c [May  6 14:13:08] DTMF[28559]: channel.c:3092 __ast_read:
>> DTMF begin passthrough '6' on SIP/test-001c [May  6 14:13:08]
>> DTMF[28559]: channel.c:2997 __ast_read: DTMF end '6' received on
>> SIP/test-001c, duration 120 ms [May  6 14:13:08] DTMF[28559]:
>> channel.c:3037 __ast_read: DTMF end accepted with begin '6' on
>> SIP/test-001c [May  6 14:13:08] DTMF[28559]: channel.c:3066 __ast_read:
>> DTMF end passthrough '6' on SIP/test-001c
>>
>> Thanks!
>>
>> On Sun, May 6, 2012 at 1:03 PM, Markus  wrote:
>>
>>
>>Am 06.05.2012 13:46, schrieb Shahid H:
>>
>>
>>Hello,
>>
>>I am having a problem with SendDTMF - it is not sending
>> the numbers
>>properly during the phone call.. I want the numbers key to
>> to be
>>pressed/sent automatically after 3 seconds during a phone
>> call.
>>
>>
>>
>>Log the ac

Re: [asterisk-users] Why SendDTMF is not working?

[Shahid H]

When I changed back to dtmfmode=rfc2833 and I cant hear the DTMF
sound.. completely silent.

Indeed I have put disallow=all before the allow=ulaw  allow=alaw

"sip show channels" in the CLI  show during a call:

78.129.xxx.xx   +447715d909406db14d2  0x4 (ulaw)   No
Tx: ACK
94.192.xxx.xx   test  MTNlNGNkYjlhODA  0x4 (ulaw)
No   Rx: ACK

Still no luck to get DTMF to work :(

Thanks
Shahid

On Sun, May 6, 2012 at 2:54 PM, Eric Wieling  wrote:

> Now you have a totally different issue.  8-)
>
> While the call is up do a "sip show channels" in the CLI.  This will show
> you the ACTUAL codec for the call.  Likely the call was still using GSM.
>  Did you remember to put a disallow=all before the allow= lines?
>
> I recommend dtmfmode=rfc2833 with whatever codec you want to use.   Inband
> DTMF will sound broken and distorted if it is sent over most codecs.
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Shahid H
> Sent: Sunday, May 06, 2012 9:16 AM
> To: Markus
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Why SendDTMF is not working?
>
> Thanks for the suggestion Markus. Here what I did:
>
> In the logger.config I have added 'dtmf':
>
> console => notice,warning,error,dtmf
>
> and then in sip.conf:
>
> allow=ulaw
> allow=alaw
> ; allow=gsm
> dtmfmode=inband
>
> I've added a test to call my mobile:
>
> exten => 123,1,Dial(SIP/+4477XXX@voipms,,D(1ww2ww3ww4))
> exten => 123,n,Hangup()
>
> then restarted asterisk and logged into console (asterisk -r)
>
> I've call my mobile using softphone, I did not see 1,2,3,4 digits being
> sent on the console but I can hear broken/unclear DTMF on the mobile...
>
> however when I press digits on the softphone I can hear DTMF clear how it
> should be on my mobile and on the console it is showing DTMF:
>
> astrisk*CLI> [May  6 14:13:06] DTMF[28559]: channel.c:3082 __ast_read:
> DTMF begin '4' received on SIP/test-001c [May  6 14:13:06] DTMF[28559]:
> channel.c:3092 __ast_read: DTMF begin passthrough '4' on SIP/test-001c
> [May  6 14:13:06] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '4'
> received on SIP/test-001c, duration 120 ms [May  6 14:13:06]
> DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '4' on
> SIP/test-001c [May  6 14:13:06] DTMF[28559]: channel.c:3066 __ast_read:
> DTMF end passthrough '4' on SIP/test-001c [May  6 14:13:07]
> DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '5' received on
> SIP/test-001c [May  6 14:13:07] DTMF[28559]: channel.c:3092 __ast_read:
> DTMF begin passthrough '5' on SIP/test-001c [May  6 14:13:07]
> DTMF[28559]: channel.c:2997 __ast_read: DTMF end '5' received on
> SIP/test-001c, duration 120 ms [May  6 14:13:07] DTMF[28559]:
> channel.c:3037 __ast_read: DTMF end accepted with begin '5' on
> SIP/test-001c [May  6 14:13:07] DTMF[28559]: channel.c:3066 __ast_read:
> DTMF end passthrough '5' on SIP/test-001c [May  6 14:13:08]
> DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '6' received on
> SIP/test-001c [May  6 14:13:08] DTMF[28559]: channel.c:3092 __ast_read:
> DTMF begin passthrough '6' on SIP/test-001c [May  6 14:13:08]
> DTMF[28559]: channel.c:2997 __ast_read: DTMF end '6' received on
> SIP/test-001c, duration 120 ms [May  6 14:13:08] DTMF[28559]:
> channel.c:3037 __ast_read: DTMF end accepted with begin '6' on
> SIP/test-001c [May  6 14:13:08] DTMF[28559]: channel.c:3066 __ast_read:
> DTMF end passthrough '6' on SIP/test-001c
>
> Thanks!
>
> On Sun, May 6, 2012 at 1:03 PM, Markus  wrote:
>
>
>Am 06.05.2012 13:46, schrieb Shahid H:
>
>
>Hello,
>
>I am having a problem with SendDTMF - it is not sending the
> numbers
>properly during the phone call.. I want the numbers key to
> to be
>pressed/sent automatically after 3 seconds during a phone
> call.
>
>
>
>Log the actual DTMF to your console, set in logger.conf:
>
>console => something,something,dtmf
>  
>
>Then try again and check if you see the actual DTMF. If you do and
> it still doesn't work, try
>
>dtmfmode=inband
>
>for your voipms peer.
>
>rfc2833 has been working always unreliable for me.
>
>Also, I'm doing DTMF like this:
>
>exten => 5000,n,Dial(SIP/123456@provider,,D(ww1ww2ww3ww4))
>
>Just use more w's to generate your 3 seconds pause. No need for
> SendDTMF.
>
>For more debugging just call yourself on your UK mobile from a
> softphone and press digits and watch the console and listen on your mobile
> if you hear the DTMF.
>
>
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Jo

Re: [asterisk-users] Calendar Integration Problem

[Bharat Lalcheta]

Hii All,

Thanks for helps. Problem solved. Its due to exchange side authentication
problem. In exchange IIS server ,for EWS site only window authentication
enabled which is not supported by neon library. Just enabled basic
authentication for EWS site on IIS and its working fine now.

Thanks for help..

Bharat Lalcheta

On Sun, May 6, 2012 at 5:14 PM, Michel Verbraak  wrote:

>  On 30-04-12 11:09, Bharat Lalcheta wrote:
>
> Hiii all,
>
> I am using asterisk 1.8.9.2 and compile all modules related to calendar.
>
> neon version is 0.29.6. OS is ubuntu 11.10.
>
> I configured ical for zimbra, caldav for google mail and ews for exchange
> 2010 calendar.
>
> ical and caldav setup working fine and i am getting my calendar events
> perfectly. But for exchange 2010 calendar i am getting following error.
>
> "Unable to communicate with Exchange Web Service at '
> https://ex1.domain.com/EWS/Exchange.asmx': Could not authenticate to
> server: ignored NTLM challenge, GSSAPI authentication error: Unspecified
> GSS failure.  Minor code may provide more information: Credentials cache
> file '/tmp/krb5cc_0' not found"
>
> my calendar.conf is as follows
>
> [calendar3]
> type = ews   ; type of calendar--currently supported: ical,
> caldav, exchange, or ews
> url = https://ex1.domain.com/EWS/Exchange.asmx ; URL to MS Exchange EWS
> user = myn...@domain.com  ; Exchange username
> secret = xx   ; Exchange password
> refresh = 10 ; refresh calendar every n minutes
> timeframe = 20
>
> calendar show status command shows following output
>
> Calendar Type   Status
>     --
> calendar3ewsfree
>
> Please help me out for solve above problem.
> Thanks in advance
> --
> Bharat Lalcheta
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>  Did you compile neon against openssl or the default internal ssl? I used
> against openssl.
> Make sure you have the rootca from the exchange server in /etc/ssl/certs
>
> The message/warning looks like the Exchange server expects a kerberos
> authentication. I have no experience with the EWS calendar module and using
> kerberos to authenticate.
>
> Hope this info helps.
>
> Michel.
>
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> _
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Bharat Lalcheta
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Re: [asterisk-users] Why SendDTMF is not working?

[Eric Wieling]

Now you have a totally different issue.  8-)

While the call is up do a "sip show channels" in the CLI.  This will show you 
the ACTUAL codec for the call.  Likely the call was still using GSM.  Did you 
remember to put a disallow=all before the allow= lines?

I recommend dtmfmode=rfc2833 with whatever codec you want to use.   Inband DTMF 
will sound broken and distorted if it is sent over most codecs.


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shahid H
Sent: Sunday, May 06, 2012 9:16 AM
To: Markus
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Why SendDTMF is not working?

Thanks for the suggestion Markus. Here what I did:

In the logger.config I have added 'dtmf': 

console => notice,warning,error,dtmf

and then in sip.conf:

allow=ulaw
allow=alaw
; allow=gsm
dtmfmode=inband

I've added a test to call my mobile:

exten => 123,1,Dial(SIP/+4477XXX@voipms,,D(1ww2ww3ww4))
exten => 123,n,Hangup()

then restarted asterisk and logged into console (asterisk -r)

I've call my mobile using softphone, I did not see 1,2,3,4 digits being sent on 
the console but I can hear broken/unclear DTMF on the mobile... 

however when I press digits on the softphone I can hear DTMF clear how it 
should be on my mobile and on the console it is showing DTMF:

astrisk*CLI> [May  6 14:13:06] DTMF[28559]: channel.c:3082 __ast_read: DTMF 
begin '4' received on SIP/test-001c [May  6 14:13:06] DTMF[28559]: 
channel.c:3092 __ast_read: DTMF begin passthrough '4' on SIP/test-001c [May 
 6 14:13:06] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '4' received on 
SIP/test-001c, duration 120 ms [May  6 14:13:06] DTMF[28559]: 
channel.c:3037 __ast_read: DTMF end accepted with begin '4' on 
SIP/test-001c [May  6 14:13:06] DTMF[28559]: channel.c:3066 __ast_read: 
DTMF end passthrough '4' on SIP/test-001c [May  6 14:13:07] DTMF[28559]: 
channel.c:3082 __ast_read: DTMF begin '5' received on SIP/test-001c [May  6 
14:13:07] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin passthrough '5' on 
SIP/test-001c [May  6 14:13:07] DTMF[28559]: channel.c:2997 __ast_read: 
DTMF end '5' received on SIP/test-001c, duration 120 ms [May  6 14:13:07] 
DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '5' on 
SIP/test-001c [May  6 14:13:07] DTMF[28559]: channel.c:3066 __ast_read: 
DTMF end passthrough '5' on SIP/test-001c [May  6 14:13:08] DTMF[28559]: 
channel.c:3082 __ast_read: DTMF begin '6' received on SIP/test-001c [May  6 
14:13:08] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin passthrough '6' on 
SIP/test-001c [May  6 14:13:08] DTMF[28559]: channel.c:2997 __ast_read: 
DTMF end '6' received on SIP/test-001c, duration 120 ms [May  6 14:13:08] 
DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '6' on 
SIP/test-001c [May  6 14:13:08] DTMF[28559]: channel.c:3066 __ast_read: 
DTMF end passthrough '6' on SIP/test-001c

Thanks!

On Sun, May 6, 2012 at 1:03 PM, Markus  wrote:


Am 06.05.2012 13:46, schrieb Shahid H:


Hello,

I am having a problem with SendDTMF - it is not sending the 
numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.



Log the actual DTMF to your console, set in logger.conf:

console => something,something,dtmf
  

Then try again and check if you see the actual DTMF. If you do and it 
still doesn't work, try

dtmfmode=inband

for your voipms peer.

rfc2833 has been working always unreliable for me.

Also, I'm doing DTMF like this:

exten => 5000,n,Dial(SIP/123456@provider,,D(ww1ww2ww3ww4))

Just use more w's to generate your 3 seconds pause. No need for 
SendDTMF.

For more debugging just call yourself on your UK mobile from a 
softphone and press digits and watch the console and listen on your mobile if 
you hear the DTMF.






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Re: [asterisk-users] Why SendDTMF is not working?

[Shahid H]

Thanks for the suggestion Markus. Here what I did:

In the logger.config I have added 'dtmf':

console => notice,warning,error,dtmf

and then in sip.conf:

allow=ulaw
allow=alaw
; allow=gsm
dtmfmode=inband

I've added a test to call my mobile:

exten => 123,1,Dial(SIP/+4477XXX@voipms,,D(1ww2ww3ww4))
exten => 123,n,Hangup()

then restarted asterisk and logged into console (asterisk -r)

I've call my mobile using softphone, I did not see 1,2,3,4 digits being
sent on the console but I can hear broken/unclear DTMF on the mobile...

however when I press digits on the softphone I can hear DTMF clear how it
should be on my mobile and on the console it is showing DTMF:

astrisk*CLI> [May  6 14:13:06] DTMF[28559]: channel.c:3082 __ast_read: DTMF
begin '4' received on SIP/test-001c
[May  6 14:13:06] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin
passthrough '4' on SIP/test-001c
[May  6 14:13:06] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '4'
received on SIP/test-001c, duration 120 ms
[May  6 14:13:06] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted
with begin '4' on SIP/test-001c
[May  6 14:13:06] DTMF[28559]: channel.c:3066 __ast_read: DTMF end
passthrough '4' on SIP/test-001c
[May  6 14:13:07] DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '5'
received on SIP/test-001c
[May  6 14:13:07] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin
passthrough '5' on SIP/test-001c
[May  6 14:13:07] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '5'
received on SIP/test-001c, duration 120 ms
[May  6 14:13:07] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted
with begin '5' on SIP/test-001c
[May  6 14:13:07] DTMF[28559]: channel.c:3066 __ast_read: DTMF end
passthrough '5' on SIP/test-001c
[May  6 14:13:08] DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '6'
received on SIP/test-001c
[May  6 14:13:08] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin
passthrough '6' on SIP/test-001c
[May  6 14:13:08] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '6'
received on SIP/test-001c, duration 120 ms
[May  6 14:13:08] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted
with begin '6' on SIP/test-001c
[May  6 14:13:08] DTMF[28559]: channel.c:3066 __ast_read: DTMF end
passthrough '6' on SIP/test-001c

Thanks!

On Sun, May 6, 2012 at 1:03 PM, Markus  wrote:

> Am 06.05.2012 13:46, schrieb Shahid H:
>
>  Hello,
>>
>> I am having a problem with SendDTMF - it is not sending the numbers
>> properly during the phone call.. I want the numbers key to to be
>> pressed/sent automatically after 3 seconds during a phone call.
>>
>
> Log the actual DTMF to your console, set in logger.conf:
>
> console => something,something,dtmf
>   
>
> Then try again and check if you see the actual DTMF. If you do and it
> still doesn't work, try
>
> dtmfmode=inband
>
> for your voipms peer.
>
> rfc2833 has been working always unreliable for me.
>
> Also, I'm doing DTMF like this:
>
> exten => 5000,n,Dial(SIP/123456@**provider,,D(ww1ww2ww3ww4))
>
> Just use more w's to generate your 3 seconds pause. No need for SendDTMF.
>
> For more debugging just call yourself on your UK mobile from a softphone
> and press digits and watch the console and listen on your mobile if you
> hear the DTMF.
>
>
>
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Re: [asterisk-users] Problem with SendDTMF

[Eric Wieling]

Try using Dial(SIP/+44797XX@voipms,30,D(ww0788XX)t)

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shahid H
Sent: Saturday, May 05, 2012 11:20 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem with SendDTMF

Hello,

I am having a problem with SendDTMF - it is not sending the numbers properly 
during the phone call.. I want the numbers key to to be pressed/sent 
automatically after 3 seconds during a phone call.

I use software phone to test it... when I dialed 501, I cant hear anything for 
about 10 seconds (this is because of SendDTMF)  and then I can hear the 
operator saying to enter the numbers but SendDTMF already did it?!

Asterisk server are connected to voip.ms provider. 

I have spent many hours trying to get to work, how to fix this issue?

See the configuration and debug log below:

extensions.conf

[test]
exten => 501,1,Set(CALLERID(num)=004471XXX)
exten => 501,n,Dial(SIP/+44797XX@voipms,30,M(sendnumber)t)
exten => 501,n,Hangup()

[macro-sendnumber]
exten => s,1,Wait(3)
exten => s,n,SendDTMF(www0w7w8w8wXwXwXwXwXwX)

sip.conf
==
[general]
context=default
tcpbindaddr=0.0.0.0
dtmfmode = rfc2833
register => x:vxx...@london.voip.ms:5060

[test]
type=peer
secret=2xxx
host=dynamic
context=test

[voipms]
canreinvite=no
host=london.voip.ms
secret=xx
type=peer
username=135xxx ;your account
disallow=all
allow=gsm
; allow=g729 ; Uncomment if you support G729 fromuser=135xxx insecure=invite 
trustrpid=yes sendrpid=yes nat=yes
dtmfmode=rfc2833




debug:
=
  == Using SIP RTP CoS mark 5
-- Executing [501@test:1] Set("SIP/test-0026", 
"CALLERID(num)=004471XX") in new stack
-- Executing [501@test:2] Dial("SIP/test-0026", 
"SIP/+4479XX@voipms,30,M(sendnumber)t") in new stack
  == Using SIP RTP CoS mark 5
-- Called +44797XX@voipms
-- SIP/voipms-0027 is making progress passing it to SIP/test-0026
-- SIP/voipms-0027 answered SIP/test-0026
-- Executing [s@macro-sendnumber:1] Wait("SIP/voipms-0027", "3") in new 
stack
-- Executing [s@macro-sendnumber:2] SendDTMF("SIP/voipms-0027", 
"www0w7w8wXwXwXwXw4wXwXwX") in new stack




Thanks!

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Re: [asterisk-users] Why SendDTMF is not working?

[Markus]

Am 06.05.2012 13:46, schrieb Shahid H:

Hello,

I am having a problem with SendDTMF - it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.


PS: You are only allowing the GSM codec for your voipms peer, why? Maybe 
that is the reason why DTMF isn't working properly. You should try alaw 
and ulaw, respectively both.





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Re: [asterisk-users] Why SendDTMF is not working?

[Markus]

Am 06.05.2012 13:46, schrieb Shahid H:

Hello,

I am having a problem with SendDTMF - it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.


Log the actual DTMF to your console, set in logger.conf:

console => something,something,dtmf
   

Then try again and check if you see the actual DTMF. If you do and it 
still doesn't work, try


dtmfmode=inband

for your voipms peer.

rfc2833 has been working always unreliable for me.

Also, I'm doing DTMF like this:

exten => 5000,n,Dial(SIP/123456@provider,,D(ww1ww2ww3ww4))

Just use more w's to generate your 3 seconds pause. No need for SendDTMF.

For more debugging just call yourself on your UK mobile from a softphone 
and press digits and watch the console and listen on your mobile if you 
hear the DTMF.




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[asterisk-users] Why SendDTMF is not working?

[Shahid H]

Hello,

I am having a problem with SendDTMF - it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.

I use software phone to test it... when I dialed 501, I cant hear anything
for about 10 seconds (this is because of SendDTMF)  and then I can hear the
operator saying to enter the numbers but SendDTMF already did it?!

Asterisk server are connected to voip.ms provider.

I have spent many hours trying to get to work, how to fix this issue?

See the configuration and debug log below:

extensions.conf

[test]
exten => 501,1,Set(CALLERID(num)=004471XXX)
exten => 501,n,Dial(SIP/+44797XX@voipms,30,M(sendnumber)t)
exten => 501,n,Hangup()

[macro-sendnumber]
exten => s,1,Wait(3)
exten => s,n,SendDTMF(www0w7w8w8wXwXwXwXwXwX)

sip.conf
==
[general]
context=default
tcpbindaddr=0.0.0.0
dtmfmode = rfc2833
register => x:vxx...@london.voip.ms:5060

[test]
type=peer
secret=2xxx
host=dynamic
context=test

[voipms]
canreinvite=no
host=london.voip.ms
secret=xx
type=peer
username=135xxx ;your account
disallow=all
allow=gsm
; allow=g729 ; Uncomment if you support G729
fromuser=135xxx
insecure=invite
trustrpid=yes
sendrpid=yes
nat=yes
dtmfmode=rfc2833


debug:
=
  == Using SIP RTP CoS mark 5
-- Executing [501@test:1] Set("SIP/test-0026",
"CALLERID(num)=004471XX") in new stack
-- Executing [501@test:2] Dial("SIP/test-0026",
"SIP/+4479XX@voipms,30,M(sendnumber)t") in new stack
  == Using SIP RTP CoS mark 5
-- Called +44797XX@voipms
-- SIP/voipms-0027 is making progress passing it to
SIP/test-0026
-- SIP/voipms-0027 answered SIP/test-0026
-- Executing [s@macro-sendnumber:1] Wait("SIP/voipms-0027", "3") in
new stack
-- Executing [s@macro-sendnumber:2] SendDTMF("SIP/voipms-0027",
"www0w7w8wXwXwXwXw4wXwXwX") in new stack


If you need any more information or debug, let me know!

Thanks!
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Re: [asterisk-users] Calendar Integration Problem

[Michel Verbraak]

On 30-04-12 11:09, Bharat Lalcheta wrote:


Hiii all,

I am using asterisk 1.8.9.2 and compile all modules related to calendar.

neon version is 0.29.6. OS is ubuntu 11.10.

I configured ical for zimbra, caldav for google mail and ews for 
exchange 2010 calendar.


ical and caldav setup working fine and i am getting my calendar events 
perfectly. But for exchange 2010 calendar i am getting following error.


"Unable to communicate with Exchange Web Service at 
'https://ex1.domain.com/EWS/Exchange.asmx': Could not authenticate to 
server: ignored NTLM challenge, GSSAPI authentication error: 
Unspecified GSS failure.  Minor code may provide more information: 
Credentials cache file '/tmp/krb5cc_0' not found"


my calendar.conf is as follows

[calendar3]
type = ews   ; type of calendar--currently supported: 
ical, caldav, exchange, or ews

url = https://ex1.domain.com/EWS/Exchange.asmx ; URL to MS Exchange EWS
user = myn...@domain.com   ; 
Exchange username

secret = xx   ; Exchange password
refresh = 10 ; refresh calendar every n minutes
timeframe = 20

calendar show status command shows following output

Calendar Type   Status
    --
calendar3ewsfree

Please help me out for solve above problem.

Thanks in advance
--
Bharat Lalcheta


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Did you compile neon against openssl or the default internal ssl? I used 
against openssl.

Make sure you have the rootca from the exchange server in /etc/ssl/certs

The message/warning looks like the Exchange server expects a kerberos 
authentication. I have no experience with the EWS calendar module and 
using kerberos to authenticate.


Hope this info helps.

Michel.
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Re: [asterisk-users] Asterisk 1.6.2.22 backtrace

[Tzafrir Cohen]

On Fri, May 04, 2012 at 11:53:41PM +0200, Jonas Kellens wrote:
> I have selected " don't optimize " in the menuselect for better
> information in the trace and now you tell me that it's still useless
> ?

(As mentioned in a separate post, this is not related to the missing
debug symbols from the external library libgcc_s)

Have you reproduced the crash with that rebuilt Asterisk? If you rebuild
(and later install) Asterisk with different build options, the original
core dump becomes useless, as the code has changed (and may reside in
different addresses).

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Asterisk 1.6.2.22 backtrace

[Tzafrir Cohen]

On Fri, May 04, 2012 at 08:34:49PM +0200, Jonas Kellens wrote:
> Hello,
> 
> what does it mean when you read in the backtrace file :
> 
> Reading symbols from /lib64/libgcc_s.so.1...(no debugging symbols
> found)...done.

No debugging symbols are avaialble for libgcc_s . Libgcc is an external
library, which is part of gcc and includes code generated by it. Its
debug information would probably be included in a package such as
libgcc-debuginfo . Try installing it.

That said, debug information for external libraries is often not that
important for Asterisk crashes.

> Loaded symbols for /lib64/libgcc_s.so.1

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   Tzafrir Cohen
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Re: [asterisk-users] chan_dahdi with asterisk 1.4 and new Linux versions

[Tzafrir Cohen]

On Fri, May 04, 2012 at 09:24:56AM -0700, bilal ghayyad wrote:

> 
> What is happening with me that when I used fedora core 16, I compiled
> and installed dahdi 2.6 and then compiled and installed asterisk 1.4
> and it did not create chan_dahdi. I tried to select it by running make
> menuselect and I discover that it is not possible !! By the way: this
> problem is not existed with old linux versions .. 

Have you re-run ./configure ?

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Re: [asterisk-users] Calendar Integration Problem

[Marek Cervenka]

Dne 30.4.2012 11:09, Bharat Lalcheta napsal(a):


Hiii all,

I am using asterisk 1.8.9.2 and compile all modules related to calendar.

neon version is 0.29.6. OS is ubuntu 11.10.

I configured ical for zimbra, caldav for google mail and ews for 
exchange 2010 calendar.


ical and caldav setup working fine and i am getting my calendar events 
perfectly. But for exchange 2010 calendar i am getting following error.


"Unable to communicate with Exchange Web Service at 
'https://ex1.domain.com/EWS/Exchange.asmx': Could not authenticate to 
server: ignored NTLM challenge, GSSAPI authentication error: 
Unspecified GSS failure.  Minor code may provide more information: 
Credentials cache file '/tmp/krb5cc_0' not found"


my calendar.conf is as follows

[calendar3]
type = ews   ; type of calendar--currently supported: 
ical, caldav, exchange, or ews

url = https://ex1.domain.com/EWS/Exchange.asmx ; URL to MS Exchange EWS
user = myn...@domain.com   ; 
Exchange username

secret = xx   ; Exchange password
refresh = 10 ; refresh calendar every n minutes
timeframe = 20



try
user = domain.com/myname   ; 
Exchange username
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