Hello list.
The VoiceMailMain()-application has an advanced menu. Can I get a
Voicemail-application that has less functionality ?
I only want the user to listen to new voicemail-messages (and delete
them), not the functionality with the folders and redirecting messages
to other mailboxes...
Hello Mike,
the semi-column did not really work :
Jonas Kellens jo...@mail1.be:i...@mail2.be: malformed address:
:i...@mail2.be may not follow Jonas Kellens jo...@mail1.be
and same result with a comma :
Jonas Kellens jo...@mail1.be,i...@mail2.be: malformed address:
,i...@mail2.be may
is not flexible at all...
Jonas.
On 06/01/2010 04:44 PM, Jared Smith wrote:
On Mon, 2010-05-31 at 22:08 +0200, Jonas Kellens wrote:
Is there yet a seperator that actually works to define multiple mail
addresses ?
Not that I'm aware of. I simply create an alias on the mail server
So Asterisk sends a user to the Voicemail()-application and creates a
mail statement with attachment and sends it to the Exim-proces.
Therefore Asterisk looks into the realtime MySQL-DB voicemail_users for
the mail-address of the voicemail-user/mailbox. Exim will then also
check the MySQL-DB
I am no programmer, and very happy with what Asterisk holds in it. Just
hoped that mailing multiple mail-addresses was an easy configuration...
On 06/01/2010 06:06 PM, Steve Howes wrote:
On 1 Jun 2010, at 16:53, Jonas Kellens wrote:
Sounds... p
Perhaps you could contribute
Hello list,
google returns a discussion on the dev-list when I search for how to
mail a voicemail to multiple mail addresses.
Is there yet a seperator that actually works to define multiple mail
addresses ?
Kind regards,
Jonas.
--
Giorgio,
have you made progress on this topic ? Because I'm interested in this too.
Jonas.
On 04/15/2010 01:18 PM, Giorgio Incantalupo wrote:
Hello,
I'm trying to setup shared lines with Asterisk 1.4.26 and Snom phones.
It seems that Asterisk works correctly (I get State:
Hello list,
I am confronted with the following problem :
making a call only leasts for about 30 seconds, then the call is ended.
The CLI shows :
[May 21 14:31:50] WARNING[25345]: chan_sip.c:1980 retrans_pkt: Maximum
retries exceeded on transmission 954539948-506...@192.168.1.100 for
seqno
For those having the same problem, my solution was to upgrade to the
newest firmware on the Linksys WAG160.
It seemed a NAT-problem because NAT-ting was not correctly handled by
the firmware.
Jonas.
On 05/21/2010 02:41 PM, Jonas Kellens wrote:
Hello list,
I am confronted with the following
What does this mean :
[May 20 09:57:28] NOTICE[14916]: chan_sip.c:15644
handle_request_subscribe: Sending fake auth rejection for user
sip:c...@ast_pub_ip;tag=wetpp2qb3f
[May 20 09:57:28] NOTICE[14916]: chan_sip.c:15644
handle_request_subscribe: Sending fake auth rejection for user
Hello list,
I read on voip-info.org that Asterisk 1.4 support T38 passthrough.
So I guess this means that I can have a Grandstream HT503 with T38
support and an analogue faxmachine on the other side of my Asterisk and
a T38-account with a ITSP on the other side of my Asterisk machine, right ?!
as it does not
mess up anything, but it still is a WARNING.
Jonas.
On 05/13/2010 04:26 PM, Doug Lytle wrote:
Jonas Kellens wrote:
exten = s,n,NoOp(fetchid = ${fetchid})
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
The only different between yours
the ringtone back ?!
Jonas.
On 05/12/2010 05:37 PM, Jonas Kellens wrote:
Yes, 20 in Queue is timeout... works fine.
Also with the Ringing() command, there is no dialtone... It's just
silence... With or without the r-option, always the same.
When there is no Queue in between the 2 dial-commands
It does not make a difference because it is the same result : All the
other queries go well, just the last one gives this 'WARNING'.
Using 1.4.25.1
Jonas.
On 05/14/2010 11:46 AM, Doug Lytle wrote:
Jonas Kellens wrote:
But it does not make a difference...
I'm running Asterisk
Hello list,
I have the following problem with MySQL-queries : it seems that the
resultid and connid are not cleared !
[macro-GetMailboxFromSIPuserID]
exten = s,1,MYSQL(Connect connid localhost xxx xxx xxx)
exten = s,n,MYSQL(Query resultid ${connid} SELECT\ extensie FROM\
tbl_SIPaccounts\
Hello list,
when I sent an incoming call first to a queue and after the timeout to a
dial-command, while the correspondent's phone rings there is no ringtone
for the caller...
So it goes like this :
1. dial(SIP/account1,20)
2. queue(myqueue20)
3. dial(SIP/account2)
In step 1 there is
.
On 05/12/2010 10:47 AM, Vardan wrote:
I think he need use r option in Dial command, while how I understand in
Queue he need musiconhold.
Dial(SIP/account2,,r)
Vardan
Ishfaq Malik wrote:
On 12/05/10 09:08, Jonas Kellens wrote:
Hello list,
when I sent an incoming call first to a queue
Yes, 20 in Queue is timeout... works fine.
Also with the Ringing() command, there is no dialtone... It's just
silence... With or without the r-option, always the same.
When there is no Queue in between the 2 dial-commands, then the ringtone
is there as it should be !
So when I change to
Hello list,
I need to set Voice QoS and SIP QoS for YeaLink. The possible values are
0 ~ 63.
With Grandstream I can fill in DiffServ 46, which is EF. That's what I want.
With Snom I fill in 184, which corresponds to EF or DSCP 46 (according
to their wiki)
But what value do I want to fill
Jared,
the Portech SIMbox is registering to a DNS name. The firewall is off and
the NAT is a Zyxel NBG-419 router.
No mather what port I set, It is not working :
--- SIP read from my_public_ip:5070 ---
REGISTER sip:sip.sipserver.tld SIP/2.0
Via: SIP/2.0/UDP
Hello list,
using asterisk 1.4.25.1 and realtime queues.
I would like to use the parameter 'membermacro' so I've added a field in
my mysql-table queues, but this is not working.
Anyone knows how I can execute a macro when the queue is answered by a
queuemember ?? Also the command queue()
Hello list,
can a conversation be recorded without the caller or callee having to
press some combination that is defined in features.conf ??
Like in queues.conf you have the ability to record a conversation with
MixMonitor when the caller is connected to an agent/member of the queue.
Can
Hello list,
is it save to send manager events from a remote website (php) to an
Asterisk-server ? Is there some tutorial on how to implement a tight
safety policy ?
Jonas.
--
_
-- Bandwidth and Colocation Provided by
What read/write rights do I need to issue this sip prune realtime peer
command in manager.conf ??
Jonas.
Jonathan Thurman wrote:
If you have a web interface for updating information you could always use AMI
to issue the prune/reload after committing the changes.
-Jonathan
--
When I
comment out the port-parameter (then it defaults to 5060), it is still
the same...
[Apr 22 09:32:49]
--- Transmitting (NAT) to my_pub_ip:5064 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.23:5064;branch=z9hG4bKc46696a2b5;received=my_pub_ip
From: "SIM
3-1"
I'm using
Firefox on Fedora but I don't think the problems lies there.
All goes well when the gateway is connected directly to the internet...
It's when it is behind NAT the 401 is sent from Asterisk...
It must be some NAT-thing combination in how the
GSM-gateway/Zyxel-router sends the
: "use STUN".
Even if the NAT rewrites the IP-address/port combination, why is it a
problem for the Portech and not for the IP-phones (Grandstream
Snom) ? They all communicate on port 5060 -- 5064 (several
SIP-accounts)
Jonas.
Jared Smith wrote:
On Thu, 2010-04-22 at 17:45 +0
the Portech is behind
NAT...
Jonas.
bruce bruce wrote:
Take out the router/firewall and connect directly to the
net to test your NAT problem theory.
On Thu, Apr 22, 2010 at 12:15 PM, Jonas
Kellens jonas.kell...@telenet.be
wrote:
Jared,
thank you for your answer.
As I said in my
Hello list,
has anyone experience with the Portech MV-374 GSM-gateway ?
I'm trying to register the SIP-accounts to a public SIP-server but that
fails.
When trying to register to a local Asterisk-server, all goes well.
So anyone knows what special setting I need to make to register my
Seq: 2354 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="103001vc",
nonce="3c911c4a"
Content-Length: 0
How come there is a register attempt that is "Una
, Jonas
Kellens jonas.kell...@telenet.be
wrote:
With tcpdump
I saw that there were packets coming in from the GSM-gateway to the
public Asterisk-server.
I saw nothing on the Asterisk-CLI that told me that there were attempts
to register, but a "sip debug" shows this :
--
When there is
a register with bad password, then this is the SIP response :
[Apr 20 16:44:29]
--- Transmitting (NAT) to my_public_ip:5061 ---
SIP/2.0 403 Forbidden (Bad auth)
Via: SIP/2.0/UDP
192.168.1.22:5061;branch=z9hG4bKbdc15a66904d1239;received=my_public_ip
From: "test"
mapping rules ?!
Jonas.
bruce bruce wrote:
Try changing port=5064 to port=5060 in your Asterisk
config file. Portech will negotiate it's port with Asterisk itself.
On Tue, Apr 20, 2010 at 10:50 AM, Jonas
Kellens jonas.kell...@telenet.be
wrote:
When
there is
a register with bad
Jonathan,
'sip show peers' works just fine...
asterisk*CLI sip show peers
Name/username Host Dyn Nat ACL Port
Status Realtime
testcorp4 (Unspecified) D N 0
UNREACHABLE Cached RT
testcorp3/testcorp3 192.168.1.100 D N 5061 OK (25
ms) Cached RT
Only you see the 'Realtime'-column, and the
Hello list,
Using Asterisk 1.4.25.1
Using realtime sip_buddies
I notice that when changing the sip_buddie name (field 'name' and
'username') or secret, this is not implemented until a sip reload.
When changing the secret, the old secret is still the one to use until
a sip reload.
When
Hello Steve,
I don't really understand what you mean.
Do I need to 'sip prune realtime all' after every change ??
Is rtcachefriends=yes a wrong setting ??
Kind regards,
Jonas.
Steve Howes wrote:
On 17 Apr 2010, at 10:25, Jonas Kellens wrote:
When changing the secret
Is rtcachefriends=yes a wrong setting ??
No, not if you want caching enabled. I enable sip realtime caching on all of my systems.
What if I do not enable caching ? What would be the effect on my
realtime configuration with sip_buddies in my mysql-DB ?
Kind regards,
Hello,
I have a question concerning SNOM M9 base station.
If my customer places a SNOM M9 base station in place A and a SNOM M9
base station in place B, which is 100 meters further... will a SNOM M9
handheld from base station A register to base station B when it enters
its DECT-environment.
Hello list,
with
the SNOM M9 DECT base station and handhelds, how can the range
best be expanded ?
Is there a DECT repeater that can be used ??
Is there a way to put some 'dumb' base station somewhere else on the
network to expand the range ?
Kind regards,
Jonas.
--
) -- Asterisk-server
-- ITSP -- rest of the world
The same TOS-settings for sip and audio are set in the Zoiper softphone.
On the workstation there is some minimal web browsing, no hardcore
downloading or file transfer.
Kind regards.
On Tue, 2010-03-23 at 17:21 +0100, jonas kellens wrote:
Hello
Hello list,
how does StopMixMonitor know which 'monitoring channel' to stop when
there are multiple conversations that are being monitored/recorded ??
I want to use StopMixMonitor in a macro, called from within
applicationmap (features.conf).
Jonas.
--
Hello list !
I have the following problem at a customer :
Their is a firewall in between the internal network (with IP-phones) and
the public Asterisk-server.
I see the following message when sip debug enabled :
[Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] --- (11
headers 11
with incoming traffic on the firewall at my
customer's site.
Jonas.
On Wed, 2010-03-24 at 06:39 -0400, Alex Balashov wrote:
Have a look at rtp.conf.
On 03/24/2010 06:33 AM, jonas kellens wrote:
Hello list !
I have the following problem at a customer :
Their is a firewall in between
Netstat is indeed a nice tip to view the RTP-connections between the
public Asterisk-server and the firewall on location.
On Wed, 2010-03-24 at 08:33 -0500, Danny Nicholas wrote:
You should be able to establish a very narrow range (4 ports per line)
by monitoring the ports with netstat and
Hello list,
what can I do to minimalize the jitter in SIP-calls at server level ?
If at local network level, there is a VoIP-router and their is a
physical network dedicated to IP-phones, but there is still jitter.
When using a Hosted Asterisk server, which settings on the
Asterisk-server can
Hello,
I'm trying to call different SIP-accounts to connect them to a
conference.
This is my call-file :
Channel: SIP/test3SIP/test1
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: from-conf
Extension: 1000
I get the following in the CLI :
[Mar 22 14:40:26] -- Attempting call on
Hello list.
Can I do a sip debug on a per call base ?
Or is there a way that I could write the SIP invite messages + SDP (to
view the codecs presented) to a file ?
Anyone tried this before with a somewhat automated Wireshark script
maybe ??
Jonas.
--
Hello list,
I have nat=no and qualify=no in my sip peer definition and still my CLI
is flooded with :
[Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985
handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (30ms /
2000ms)
[Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985
That is indeed an option, thank you.
It also went away by restarting Asterisk, but this is not desirable in
production environment.
On Fri, 2010-03-12 at 11:05 +0100, Alexandru Oniciuc wrote:
Edit logger.conf and set the desired log level.
To disable the messages below just remove the
If I have this is sip.conf :
[general]
disallow=all
allow=g729
allow=alaw
The prefered codecs set in my Grandstream phone is G729, alaw.
In the sip peer definition I have commented out 'disallow=' and
'allow='.
The prefered codecs set in the Zoiper softphone is alaw, gsm.
In the sip peer
I would also add the following :
sip.conf has :
[general]
disallow=all
allow=g729
allow=alaw
allow=gsm
And again the same in the sip peer definition :
disallow=all
allow=g729
allow=alaw
allow=gsm
sip debug shows :
[Mar 12 15:28:23] Found audio description format G729 for ID 18
[Mar 12
Yes I am for most of my SIP peers.
On Fri, 2010-03-12 at 10:51 -0600, Warren Selby wrote:
On Fri, Mar 12, 2010 at 4:25 AM, jonas kellens
jonas.kell...@telenet.be wrote:
Are you using SIP realtime?
--
Thanks,
--Warren Selby
http://www.selbytech.com
I'm using Asterisk 1.4.25.1, can I do something like :
== extconfig.conf ==
hints = mysql,asterisk,hints
(got the info from https://issues.asterisk.org/view.php?id=16059 )
On Wed, 2010-03-10 at 18:26 +0100, jonas kellens wrote:
Hello list,
Can I do something like this for BLF
How can I set the prefered codec between 2 calling parties ??
My Grandstream supports G729, alaw and gsm... in this order.
The Zoiper softphone has alaw and gsm as codecs... in that order.
Although there should be a matching codec found, my Grandstream can not
call the Zoiper softphone.
CLI
= no
allowsubscribe=yes
limitonpeer = yes
notifyringing=yes
notifyhold=yes
then come the registrations...
Jonas.
On Fri, 2010-03-12 at 11:47 +0530, Prince Singh wrote:
Post your Asterisk's sip.conf
On Thu, Mar 11, 2010 at 10:39 PM, jonas kellens
jonas.kell...@telenet.be wrote:
How can I
Hello list.
An incoming call goes to the queue. Then is routed to a free
SIP-member1. When this SIP-member1 transfers the call to another
SIP-member2, and this SIPmember-2 rejects the call, then the
communication is lost.
How can I make the call go back to the SIP-member1 ? Or maybe back to
the
Hello list,
Can I do something like this for BLF functionality :
[test-blf]
exten = _XX,hint,Macro(GetSIPaccount,${EXTEN})
exten = _XX,hint,SIP/${SIPACCOUNT}
GetSIPaccount is a macro that looks at a MySQL-database for the realtime
table sip_buddies where the SIPusername is taken from.
It
Hello !
My macro to avoid voicemail of a cellphone is not really working. Can
you take a look at it :
This is the macro :
[macro-testgsm]
exten = s,1,NoOp(inside macro testgsm)
exten = s,n,Wait(2)
exten = s,n,Read(INPUT,,1,1,1)
exten = s,n,GoToIf($[${INPUT}==1]?exit:hangup)
exten =
Hello list.
ChanIsAvail returns 20 for ${AVAILSTATUS}. What does this '20' mean ??
...
exten = 1,n,ChanIsAvail(SIP/sin10)
exten = 1,n,NoOp(chanisavail == ${AVAILSTATUS})
...
[Mar 4 15:10:16] -- Executing [...@sin:7]
ChanIsAvail(IAX2/testlocal-14088, SIP/sin10) in new stack
[Mar 4
On Tue, 2010-03-02 at 14:42 -0500, Fred Posner wrote:
On Mar 2, 2010, at 2:37 PM, jonas kellens wrote:
Does Asterisk know when it hits a voicemailbox ?
When calling to a cell-phone or GSM, after some rings and no pickup you
arrive at a voicemailbox.
If Asterisk does not know it's
Thank you for your answer, Nic.
It seems that by putting rtcachefriend=yes, the qualify works as
expected and even changes made to my realtime MySQL-DB take affect
immediately without the need of a reload (I changed the username and
name).
However the old username and name are still valuable and
I'd like to add to my thread that realtime SIP peers do not seem to be
surviving a sip reload.
step 1 : 2 realtime SIP peers are registered to Asterisk, they can make
a phone call to each other.
step 2 : I do a 'sip reload'
step 3 : the 2 realtime SIP peers are no longer able to phone to each
On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote:
In my experience, yes, that is normal behaviour. Generally any SIP phone
will try to reconnect with the server within 2 mins anyway.
In the Zoiper softphone, it is set to 3600 seconds... I don't want my
customers have to do a lot of
On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote:
If you are changing RealTime config in your DB you need to do a sip
prune realtime either directly from asterisk cli or using AMI. You
really do not need to do a SIP reload when changing the config of one
sip extension.
I notice that
Does Asterisk know when it hits a voicemailbox ?
When calling to a cell-phone or GSM, after some rings and no pickup you
arrive at a voicemailbox.
If Asterisk does not know it's a voicemailbox that has answered the
call, the voicemailbox will contain 60minutes of 'silence'. This is very
Hello list,
is it necessary to properly answer() an incoming call ?
I don't want to answer a call because the caller has to pay even if the
attached SIP-phones do not answer the phone call. Because I answer() the
incoming call, the caller has to pay for 60 seconds of 'ringtone'.
On the other
[Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify
is incompatible with dynamic uncached realtime. Please either turn
rtcachefriends on or turn qualify off on peer 'gerrie'
Am I correct that when I turn on rtcachefriends in sip.conf,
database-changes in my MySQL-DB will not
Hello list,
when installing Dahdi, the following error comes up :
You do not appear to have the sources for the 2.6.18-164.11.1.el5xen kernel
installed.
make[1]: *** [modules] Error 1
The running kernel version :
-bash-3.2# uname -a
Linux vds.hosting.net 2.6.18-164.11.1.el5xen #1 SMP Wed
I have followed the instructions on voip-info.org for Realtime SIP
peers, but I get this notice :
[Feb 22 20:05:32] NOTICE[15298]: chan_sip.c:15889
handle_request_register: Registration from
'sip:test...@192.168.1.150;transport=UDP' failed for '192.168.1.105' -
No matching peer found
The CLI
] == Binding sippeers to mysql/Asterisk/sip_buddies
My database-name is just 'Asterisk', my bad.
So... what am I missing for this realtime SIP to work ??
Jonas
On Mon, 2010-02-22 at 20:13 +0100, jonas kellens wrote:
I have followed the instructions on voip-info.org for Realtime SIP
peers, but I
The problem was that I had a different value for 'name' and 'username'.
How can I have the 'name' different from the 'username' ??? Why do these
2 need to be the same ??
Jonas.
On Mon, 2010-02-22 at 20:36 +0100, jonas kellens wrote:
Little fault in my mailing :
The CLI shows :
[Feb 22 19
Dear Juan,
thank you for your answer. The reason why registration failed was a
mismatch between the 'username'-field and the 'name'-field.
If I put both values to the same, it works... But why do these 2 need to
be the same ? I would rather have a different 'name' and
'username'-parameter.
not doable.
Jonas.
On Thu, 2010-02-18 at 20:15 +0100, jonas kellens wrote:
How about something like :
[mycontext]
exten = 100,1,NoOp(calling 100)
exten = 100,n,NoOp(going realtime)
switch = Realtime/mycont...@realtime_extensions ; from here on we use
realtime
And then my MySQL-DB contains
Anyone know if my example of combining extensions.conf and realtime
extensions is doable ??
Kind regards,
Jonas.
On Thu, 2010-02-18 at 20:15 +0100, jonas kellens wrote:
How about something like :
[mycontext]
exten = 100,1,NoOp(calling 100)
exten = 100,n,NoOp(going realtime)
switch
Hello list !
Can realtime dialplan be combined with 'hardcoded' dialplan in
extensions.conf ??
Does a context need completely be written or in extensions.conf or in
the mysql-table 'extensions_table' ? Or can I combine the two with the
'switch'-statement ??
Kind regards,
Jonas.
--
:55 -0500, Jared Smith wrote:
On Thu, 2010-02-18 at 19:46 +0100, jonas kellens wrote:
Does a context need completely be written or in extensions.conf or in
the mysql-table 'extensions_table' ? Or can I combine the two with the
'switch'-statement ??
You can certainly combine the two
Hello list.
I would like to set the CDR(src)-variable to the SIPphone that is
initiating the call.
When calling out, the src-variable is always the public telephone
number.
I get the ERROR : ast_cdr_setvar: Attempt to set the 'src' read-only
variable!
Is there some way to implement this ??
Hello list !
I'm having one way audio on incoming and outgoing calls. Outgoing audio
works fine, incoming audio is not working.
My setup is the following :
incoming calls :
PSTN -- FXOport -- HT503 -- WANport -- Asterisk -- WANport -- HT503 (the
same) -- FXSport -- DECTphone
outgoing calls :
multiple outgoing peer definitions without having an
incoming call be matched against these peers
Thank you !
Jonas.
On Fri, 2010-01-29 at 16:51 +, Robert Lister wrote:
On Fri, 2010-01-29 at 15:09 +0100, jonas kellens wrote:
Hello list !
Having troubles with multiple
Hello list !
Having troubles with multiple registrations to one and the same ITSP.
This sip.conf :
register = user1:pass...@sip.itsp
register = user2:pass...@sip.itsp
; outgoing conversations
[user1-out]
type=peer
host=sip.ITSP
username=user1
secret=secret1
fromuser=user1
dtmfmode=rfc2833
-server ?!
Jonas.
On Fri, 2010-01-29 at 16:51 +, Robert Lister wrote:
On Fri, 2010-01-29 at 15:09 +0100, jonas kellens wrote:
Hello list !
Having troubles with multiple registrations to one and the same ITSP.
This sip.conf :
register = user1:pass...@sip.itsp
register = user2
Hello list,
anyone have a manual for the webGUI of the above phone ? Just bought a
Polycom Soundpoint IP300 and on the site of Polycom I see a user manual
and a administrator's manual but none of these 2 guides explain the
fields in the webGUI.
Trying to understand the difference between the
Hello list,
I'm using an IVR where the caller chooses between 1. sales 2. support.
When choosing 1 the caller is directed to the sales-queue when choosing
2 the caller is directed to the support-queue.
Then the caller is directed to a free agent.
I notice in the CDR-rapports that the destination
I'm using Asterisk 1.4.27.
In queues.conf I do not find this option. I have added it, reloaded
Asterisk, but still the destination is '1' or '2'.
Does it make a difference of my queue members are just SIP-accounts in
stead of agents ?
member = SIP/VCsupport,1,Jonas
member =
Hello Danny,
what do you mean by 'all the CDR fields' ?
The Destination-field shows '1' or '2'. The dstchannel shows the correct
SIP-channel. But this is not the same as the 'real' destination namely
the SIP-account of my SIP-phone.
Jonas.
On Wed, 2010-01-27 at 08:22 -0600, Danny Nicholas
On Sat, 2010-01-23 at 21:19 -0500, Alex Balashov wrote:
What is the situation with Asterisk and fax over IP ? Can Asterisk
receive a fax over a POTS or ISDN line ?? Do I then need a Digium
TDM-card and an FXO-module or a T38-gateway ?
Despite what anyone may say about Fax over IP
Dear members of the list,
a customer of mine has some questions and I would like to pose some of
them further to you guys.
What is the situation with Asterisk and fax over IP ? Can Asterisk
receive a fax over a POTS or ISDN line ?? Do I then need a Digium
TDM-card and an FXO-module or a
Hello list,
I have the following in my voicemail.conf :
[general]
format=wav|wav49
When I receive a WAV-file from my Asterisk-server, I am unable to play
the file... There is no player on my Fedora that wants to play the file.
When I make my Asterisk-server unreachable, the incoming call goes
Hello list.
Is it possible in the Asterisk dialplan to send a 503 Service
Unavailable of 603 Decline after having answered the call with Answer()
in the dialplan ??
Suppose that I first want to check the call in a MySQL-database, while I
put some MoH, and then let the call go through or send
Thank you for your answer.
So if I use early media (not putting answer() at the beginning of my
dialplan), how can I send a 503 or 603 from the dialplan ??
Kind regards,
Jonas.
On Tue, 2010-01-12 at 12:05 -0600, Kevin P. Fleming wrote:
jonas kellens wrote:
Is it possible in the Asterisk
hearing music on hold in stead of a ring tone
when an agent's phone calls. Moh stops when the agents picks up.
So I'm not quite there yet. Any feedback is appreciated.
Jonas.
On Wed, 2010-01-06 at 15:43 +0100, jonas kellens wrote:
Please can someone help me with my queue-problems ?
When
for as long as the conversation of the first caller
takes.
It is as off there is nobody to pick up the phone, as in reality the
queue member is busy with an other conversation.
Jonas.
On Mon, 2010-01-04 at 09:51 +0100, jonas kellens wrote:
Hello list !
I have some configuration issues with queues
Hello list !
I have some configuration issues with queues, but I'm sure they are
minor and for someone who has already configured queues it could be
trivial.
This is my queue configuration :
[VC_support_queue]
musicclass = default
strategy = ringall
timeout = 20
retry = 5
wrapuptime=15
[Dec 31 10:39:45] WARNING[17884]: pbx.c:2518 __ast_pbx_run: Invalid
extension '11', but no rule 'i' in context ...[snip]...
When testing IVR and pressing 1 from my Grandstream SIP-phone, the
above message is printed on the Asterisk CLI.
How come Asterisk receives my 1 as 11 ??
Settings in my
Francesco,
Marking only RTP or only SIP info makes my DTMF to be correctly received
by Asterisk (read: only once).
It works fine now.
Thanks.
Jonas.
On Thu, 2009-12-31 at 10:53 +0100, Francesco Peeters wrote:
Jonas.
It may be me, but it looks like Asterisk correctly interprets the
Hello list.
My phone rings, I pick up, and the conversation is terminated. Every
time.
The setup :
Grandstream GXP2010 -- SIPproxy (Endian Firewall) -- Asterisk Server
-- ITSP
Could it be the SIP proxy of my Endian firewall ??
I have 4 accounts on the Grandstream which listen on port 5060 --
I have set SIP debug on but it is too much output to post on the
mailinglist. I have tried to understand the SIP-messages between my
Grandstream and my Asterisk-server and my Asterisk server and the ITSP.
This is some output that's a bit shorter :
debug log :
[Dec 27 12:11:32] DEBUG[14035]
Hello list !
This is my call to the macro :
exten = 36,n,Macro(mymacro,${KNUMMER},yo)
This is where {ARG1} and {ARG2} are used inside the macro :
exten = s,n,Voicemail(${ar...@${arg2},s)
This the output on the CLI :
VoiceMail(IAX2/zoiper-13958, 908001@|s)
The variable ${KNUMMER} is set in
Calling my home numbers has always worked. Till now. The Asterisk CLI
show the following :
[Dec 23 10:53:22] NOTICE[25159]: chan_sip.c:12640 handle_response_invite:
Failed to authenticate on INVITE to 'sip:092xx9...@85.xx.xx.xx;tag=as5b139383'
And after restarting Asterisk, the CLI is
Where is your definition of codecs ??
On Tue, 2009-12-22 at 11:26 +0200, zehra yildiz wrote:
Hello All,
I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone.
The softphone can call the other one but I can' t hear any voice. My
configuration files are below:
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