I got it !!
host=192.168.0.151
port=5060
type=friend
nat=yes
qualify=yes
fromdomain=192.168.0.151
insecure=invite,port
dtmfmode=auto
disallow=all
allow=alaw&g729 -<-here! make a tention at the order! G729 is
not allowed !
i reorder it get work!!
thks a lot,all !
On 26 March 2010
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SIP: voipat...@vuc.onsip.com - Enter 22622# and your PIN# if you have
no PIN you can listen using 1#
iNum - +883 51007 039 9924
hi,all
when i send a call to other server use SIP trunk,
i got this below,
<--- SIP read from 222.46.18.52:5060 --->
SIP/2.0 403 Forbidden
what's rong with is?
> Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others.
> Created by Mark Spencer
> Asterisk comes with ABSOLUTELY
it doesn't seems to be a problem of communication between A and B
>-- Executing [...@macro-dialout-trunk:19]
Dial("SIP/192.168.0.151-088e7938",
"ZAP/g0/15921256331|300|M(setmusic^none)Tt") in new stack
> == Everyone is busy/congested at this time (1:0/0/1)
That's says it's more a problem with
If you didn't have this problem before I'll check up for any changes lately
(i suppose you have done so, but ask this just to be safe)
I see you have lots of agents and also lots of hard disk space, so I guess
disk space is not an issue. Please check it anyway.
how many concurrent calls you have?
Jim-
>> Jim-
>>
>>> There will be up to 150 phones so there will be 300
>>> channels when they are all on the phone at one time.
>>>
>>> I will be using a current 1.4 version.
>>
>> That's a lot of channels for Asterisk... IIRC the TC400B transcoding card is
>> rated at up to 96 G729 channels.
>>
sean darcy wrote:
> Using spandsp-0.0.6-pre17, SendFax on 1.6.1.18 and ReceiveFax on
> 1.6.2.8. Sip.conf on both sides has t38pt_udptl = yes.
>
> -- Executing [...@fax-tx-test:3] SendFAX("SIP/side-sip-0009",
> "/var/spool/asterisk/fax/20091113_1455.tif") in new stack
> [Mar 20 17:05:34]
Dear sir,
We have been using asterisk for 4 years. Now we have got problems which
occurs during the attended transfer.
But we are not always getting this problem. Sometimes it happens. But now we
cannot understand why this is happening?
problem is:"Failed to play transfer sound! "
The log of ast
>> I think you would be more successful and have more control if you wrote
>> it as an AGI. Then you could set a timer that would interrupt the
>> process and you could do what you like from there (hangup?). I think
>> you are asking too much of the dialplan.
> I would tend to leap into an AG
i have a prablom here,
i want to send a call from A to B use sip trunk ,
the call can sended B,but not work to PSTN.
the message from B server. help pls,what's rong?
>
> <--- SIP read from 192.168.0.176:5060 --->
> INVITE sip:15921256...@192.168.0.151 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.176:
Any probs with the circuits?
Try and upgrade?
On 17/03/2010, Russell Brown wrote:
>
>
> I'm seeing both inbound and outgoing call failures on our ISDN-30 lines
> that only seem to go away when I do a "zap restart" or in extremis
> restart Asterisk (1.4.25 with a Digium TE205P and zaptel 1.4.12.1
On Mar 25, 2010, at 4:10 PM, Jeff Brower wrote:
> Jim-
>
>> There will be up to 150 phones so there will be 300
>> channels when they are all on the phone at one time.
>>
>> I will be using a current 1.4 version.
>
> That's a lot of channels for Asterisk... IIRC the TC400B transcoding card is
Hi!
> well here is what i did to solve it but i still don't know why i had
> to or why my current config works.
Really, you should take this to a FreePBX forum or mailing list.
Do a "locate amportal" and you might be a bit wiser, but please do keep
your posts here on topic.
Philipp
--
_
On Thu, Mar 25, 2010 at 09:58:17PM +, Ott Rose wrote:
>
> well here is what i did to solve it but i still don't know why i had to or
> why my current config works.
>
> i edited /etc/rc.local
You don't need to have anything in /etc/rc.local if you have a proper
/etc/init.d/dahdi .
Maybe peo
Daryl-
>> I'm involved in discussions with my carrier right now and am
>> wondering if anyone has interconnected Asterisk to
>> Metasphere via SIP?
>>
>
>
> Yes, we're served by a Metaswitch usng SIP. Works fine.
Metasphere is MetaSwitch's PC/server based system, not to be confused with
their l
Hi!
> i have recently connected my (working) asterisk 1.2 server, with two
> 1.4 asterisk servers (one using SIP the other using IAX), since then
> (i believe) people starts complaining about a high background noise
The best idea is probably to start out by looking at the codecs. If you
happen
Jim-
> There will be up to 150 phones so there will be 300
> channels when they are all on the phone at one time.
>
> I will be using a current 1.4 version.
That's a lot of channels for Asterisk... IIRC the TC400B transcoding card is
rated at up to 96 G729 channels.
Can you clarify your recordi
well here is what i did to solve it but i still don't know why i had to or why
my current config works.
i edited /etc/rc.local
old
touch /var/lock/subsys/local
/usr/local/sbin/amportal start
new
touch /var/lock/subsys/local
/usr/sbin/amportal start
keep in mind that my server that i have b
Hi!
> I am testing the Openstage phones from Siemens but I can not find a
> solution on how to update the caller-id after a successful attended
> transfer.
When I tested the OpenStage 60 recently I did not get that to work
either, but this was with a medium aged Asterisk 1.4.17. Not sure where
Hi!
> > Try using DIALSTATUS.
>
> Thank you!
>
> but DIALSTATUS IS used for Dial. not for queue
Look at HANGUPCAUSE. It translates in a non-biunique fashion,
unfortunately, but Asterisk is ISDN/PSTN centric and does not provide
direct access to the SIP error code.
Philipp
--
_
> From: steve-li...@geekinter.net
> Date: Thu, 25 Mar 2010 18:41:41 +
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] new server install errors starting asterisk
>
> Sorry to keep jumping back to the previously ignored attempts to help, but
> does that file exist?
i
2010/3/25 Zeeshan Zakaria :
> Tzafrir, so you have actually worked with more than 192 concurrent zap
> channels, which means more than 8 spans, on a single server, and can verify
> that it actually works without freezing asterisk.
As I have written before - I did use 8 E1 in one machine quite oft
2010/3/25 Steve Edwards :
> On Thu, 25 Mar 2010, Tzafrir Cohen wrote:
>
> [snipping a lot of interesting technical and historical details]
>
>> As you can see, there's actually a limit at the DAHDI level.
>> DAHDI_MAX_SPANS, which is 128. Likewise there's DAHDI_MAX_CHANS which is
>> 1024. That's as
Sorry to keep jumping back to the previously ignored attempts to help, but does
that file exist?
S
On 25 Mar 2010, at 16:46, Ott Rose wrote:
> so i went back to 1.6.1.18 and didn't have any issue with the install.
> following the same setups as before with 1.6.2.
>
> finished the install and
Hi Guys,
i have recently connected my (working) asterisk 1.2 server, with two 1.4
asterisk servers (one using SIP the other using IAX), since then (i believe)
people starts complaining about a high background noise when using the
handset on Polycom phones (but when using the speaker it's fine, and
Tzafrir, so you have actually worked with more than 192 concurrent zap
channels, which means more than 8 spans, on a single server, and can verify
that it actually works without freezing asterisk.
I really need specs for this system. I'll recompile zaptel, no problem, but
it'll save me one extra s
On Thu, 25 Mar 2010, Tzafrir Cohen wrote:
[snipping a lot of interesting technical and historical details]
> As you can see, there's actually a limit at the DAHDI level.
> DAHDI_MAX_SPANS, which is 128. Likewise there's DAHDI_MAX_CHANS which is
> 1024. That's as many channels that you can have.
> Thank you for your reply.
>
>
> The first proposed solution has resolved the problem for a test in the local
> network. Another test is planned today later with a client in the same NAT
> and another in the public internet with a public static ip address.
>
> Do you have any advice for that ca
On Thu, Mar 25, 2010 at 10:24:41AM -0600, Carlos Chavez wrote:
> On Thu, 2010-03-25 at 17:11 +0100, Christian Victor wrote:
> > Hi James,
> >
> > we did sucessfully run two TE410P with 8xE1 in a decent Intel P4-3200
> > machine with quite heavy line usage. No codec conversion course.
> >
> > I do
- "Asterisk" wrote:
> Hi Steve,
>
> Yes, that's true. It seems that Asterisk gets it with great delay. For
> instance:
>
> Asterisk says:
> ==
>
> Mar 25 11:26:54 VERBOSE[1385] logger.c: Reliably Transmitting (no NAT)
> to 172.11.11.2:5060:
> OPTIONS sip:mytestph...@172.11.11.2
After a power interruption, asterisk doesn't seem to be routing calls and
there seems to be a premature timeout and hangups occurring. I am clueless
where to look. Can someone in the know, look at the following log and
enlighten me if there's a problem, or if it looks normal. From the calling
phone
so i went back to 1.6.1.18 and didn't have any issue with the install.
following the same setups as before with 1.6.2.
finished the install and now i have an issue were asterisk doesn't start on
reboot and also the flash panal doesn't run. i get this in freepbx. Could not
reload the FOP operat
On 3/25/2010 8:13 AM, David Gibbons wrote:
> Hi All
>
> I'm involved in discussions with my carrier right now and am wondering if
> anyone has interconnected Asterisk to Metasphere via SIP?
>
Yes, we're served by a Metaswitch usng SIP. Works fine.
-Daryl
--
_
Does anybody know specs of their server, cards, and versions of asterisk,
zaptel and libpri?
On 2010-03-25 12:28 PM, "Carlos Chavez" wrote:
On Thu, 2010-03-25 at 17:11 +0100, Christian Victor wrote:
> Hi James,
>
> we did sucessfully run t...
If you ever take the DCAP training they use a
On Thu, 2010-03-25 at 17:11 +0100, Christian Victor wrote:
> Hi James,
>
> we did sucessfully run two TE410P with 8xE1 in a decent Intel P4-3200
> machine with quite heavy line usage. No codec conversion course.
>
> I don't believe that there is a hard limit of E1s coded into Asterisk.
> But the
Chris, I am running 8 spans on a few servers. It is when you go beyond that.
If anyone is running 9 or more spans successfully, please let us know their
configuration, it'll be very helpful.
--
Zeeshan A Zakaria
On 2010-03-25 12:19 PM, "Christian Victor" wrote:
Hi James,
we did sucessfully run
Hi James,
we did sucessfully run two TE410P with 8xE1 in a decent Intel P4-3200
machine with quite heavy line usage. No codec conversion course.
I don't believe that there is a hard limit of E1s coded into Asterisk.
But the maximum lines you can squeeze out of your specific hardware
depends on so
Zeeshan A Zakaria wrote:
>On Wed, Mar 24, 2010 at 5:42 PM, James Lamanna wrote:
[snip]
>>
>> The specs of the machine are, Dual Xeon 2.80Ghz (both single core but w/HT)
>> 4GB memory.
>> Running asterisk 1.4.26.3 (32-bit)
>> with libpri-1.4.7 and zaptel-1.4.12.9
>
>So I think it is not your T1 car
Hi All
I'm involved in discussions with my carrier right now and am wondering if
anyone has interconnected Asterisk to Metasphere via SIP?
Thanks
Dave
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Hi,
we have a problem with Asterisk that is described in
https://issues.asterisk.org/view.php?id=15915. According to the last post in
the bug report, a workaround is using of static linking. When I tried (I
enabled option Compiler Flags/STATIC_BUILD) it I got the following error
message:
/usr
On 25 Mar 2010, at 14:02, Ott Rose wrote:
> well i followed the same directions i used like 3 weeks ago with 1.6.0 and
> didn't have any issue. Not sure what went wrong. That why i posted it.
>
> how can it work one time and not the next.
Does the file exist? If not, then something is diffe
well i followed the same directions i used like 3 weeks ago with 1.6.0 and
didn't have any issue. Not sure what went wrong. That why i posted it.
how can it work one time and not the next.
> From: steve-li...@geekinter.net
> Date: Thu, 25 Mar 2010 13:28:57 +
> To: asterisk-users@lists.di
On 25 Mar 2010, at 13:08, Ott Rose wrote:
> Can't find indications config file indications.conf.
Thats the last line. Probably the problem... Amazing what reading instructions
does...
S
--
_
-- Bandwidth and Colocation Provide
Hello,
Thank you for your reply.
The first proposed solution has resolved the problem for a test in the local
network. Another test is planned today later with a client in the same NAT
and another in the public internet with a public static ip address.
Do you have any advice for that case?
-
I have a Diameter server I want to integrate it with asterisk for CCR in a
prepaid scenario
did anyone have implemented it
I did saw that one guy was working with cdr_diameter but the project seems
to be suspended
--
Regards
Tushar Jain
"two roads diverged in a wood, and I - I took the one les
> Date: Thu, 25 Mar 2010 11:30:49 +1300
> From: li...@venturevoip.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] new server install errors starting asterisk
>
> Just try running:
>
> asterisk -vcd
here you go.
[r...@phoneserver src]# asterisk -vcd
A
Loic Didelot wrote:
> I am testing the Openstage phones from Siemens but I can not find a
> solution on how to update the caller-id after a successful attended
> transfer. Of course, I mean an attended transfer by using the phones
> functionality, not something defined in asterisks features.conf.
I can't find this in the wiki/email history..but I'm sure it's based asked
before.
The port range define in rtp.conf - is that for connections initiated by
asterisk? Or the port range asterisk listens on? Or both?
Thanks!
MD
--
24 mar 2010 kl. 16.48 skrev Karl Fife:
>>> Steve Edwards wrote:
>>>
It may not be as intended, but from a "user" standpoint, it seems
logical
and convenient to establish "policy" in [general] and make exceptions in
the entities as needed.
>>>
>>> Right... for when you have
On Wed, Mar 24, 2010 at 05:49:30PM -0400, sean darcy wrote:
> Tzafrir Cohen wrote:
> > On Mon, Mar 22, 2010 at 09:38:26PM -0400, sean darcy wrote:
> >> 1.6.2:
> >>
> >> -- Executing [...@incoming-pstn-line:4] VoiceMail("DAHDI/4-1",
> >> "1...@default,u") in new stack
> >> -- Playing
>
Hi Steve,
Yes, that's true. It seems that Asterisk gets it with great delay. For instance:
Asterisk says:
==
Mar 25 11:26:54 VERBOSE[1385] logger.c: Reliably Transmitting (no NAT) to
172.11.11.2:5060:
OPTIONS sip:mytestph...@172.11.11.2 SIP/2.0
Via: SIP/2.0/UDP 172.11.0.201:5060;bra
On 25 Mar 2010, at 10:18, Asterisk wrote:
> How is it possible that the peer becames UNREACHABLE eventhough Wireshark
> logged its proper response?
Wireshark received it, doesn't mean Asterisk did. what does a sip debug in
Asterisk show?
S
--
__
Hi guys,
I have one tricky question regarding SIP peers becoming unreachable.
I was logging the network traffic with Wireshark, and this is what it logged:
10:22:33.319719000 == sent from Asterisk to the phone:
OPTIONS sip:mytestph...@172.11.8.30 SIP/2.0
Via: SIP/2.0/UDP 172.32.0.201:5060;branch
Hello All,
I do have asterisk installed for a call centre with aheeva application and
i would like to know how to configure the sound for the inbound calls and if
there is any possibility for agent to receive a file with the phone number
and name of clients: For your information there is no probl
Hello,
I am testing the Openstage phones from Siemens but I can not find a
solution on how to update the caller-id after a successful attended
transfer. Of course, I mean an attended transfer by using the phones
functionality, not something defined in asterisks features.conf.
Any idea on how to ac
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi, Alyed.
On Mon, 22 Mar 2010, Alyed wrote:
> you are right, under [channels] is where it's supposed to be my
> mistake, i guess i was thinking in sip.conf :)
Perfect :-)
>> However, the following doubt arises to me: it would also have had
>> thi
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