On Mar 20, 2006, at 4:39 PM, Gabriel Afana wrote:
I just did a little RTP debug and this is what it shows:
== Spawn extension (304, 301, 1) exited non-zero on 'SIP/304-d9f8'
-- Accepting AUTHENTICATED call from 216.152.244.81:
requested format = ulaw,
requested prefs = (),
actual
Thanks for the response.
Yes, canreinvite is set to no on all lines.
After some testing, I was able to get sound between phones when they were
both registered to the same server. Maybe the IAX trunk is messing
something up. strange because it was working perfect last week and nothing
changed!
Hi,I've tryed it using my mobile and I've been charged.Maybe, my mobile operator(Vodafone) does not support it?Thanks again.p.s.: hi John, I love to learn(books, google, lists, ecc...) and cooperation and I can say that everytime I learn something new. :o)
2006/3/21, CC Jay [EMAIL PROTECTED]:
Not
hi,
if interested please consider the TigerNetcom box of 104 for doing the same functionality, much better piece and at considerable lower price.
for technical information on how to use it i would be happy to assist off list.
On 3/20/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Mar 19, 2006, at
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You can do the same thing with DND. Turn the value on or off, then in
your dial string, check the database value and act accordingly.
Hi Doug.
Do you know how to, when leaving office, set all incoming calls to transfer do
my coworker?
Hi,
we just noticed a strange CDR problem. We are using individual phone numbers
for all our SIP phones. During dialout we do a database lookup in order to
set the correct callerid (e.g. phone has number 100 but in external calls
this should be displayed as CID -20).
This works like a charm and
Hi,
I'm having a problem with the queue behaviour in my place:
I have two ISDN channels to the outside (Zap/1) and two channels two a
Siemens Gigaset (Zap/4). I also use a SIP gateway to call outside and
have a couple of IP phones around as well (SIP).
The Gigaset has about 5 phones connected
[EMAIL PROTECTED] a écrit :
Oops !
I have upgraded TRUNK again via SVN and all was seeming to be fine, no more
invalid IAX2 frames and able to place and receive calls.
I was happy..
But, few calls later (about 5 minutes) : INVAL frames again and no more
possibility to place or receive calls,
Hello All,i am repeating this question for the sixth time but i think i was not explaining the problem correctly. . Now i will try to explain it..I have 4 telephone lines(PSTN) in my PBX.Now I want to makegroups of the extensions to use that lines. e.g. extensions
Hello All,i am repeating this question for the sixth time but i think i was not explaining the problem correctly. . Now i will try to explain it..I have 4 telephone lines(PSTN) in my PBX.Now I want to makegroups of the extensions to use that lines. e.g. extensions
RTFM
El Martes, 21 de Marzo de 2006 10:53, Faisal Inam escribió:
Hello All,
i am repeating this question for the sixth time but i think i was not
explaining the problem correctly. . Now i will try to explain
it..
I have 4 telephone lines(PSTN) in my PBX.
Now I want
Hi
i have installed unixODBC Drivers and created DSN
but when i reload from * console
asterisk terminating the services
if i do again start the services
and change something and reload
again its hangs
is that bug in 1.2.5
ram
On 3/21/06, Aaron Daniel [EMAIL PROTECTED] wrote:
Since you say
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You can do the same thing with DND. Turn the value on or off, then in
your dial string, check the database value and act accordingly.
Hi Doug.
Do you know how to, when leaving office, set all incoming
Anton Krall wrote:
Guys.
This idea has been banging my headfor days now and I feel the need to share
with you.
Imagine this scenario: all calls come in thru a receptionist, asterisk
records all incoming calls, the receptionist's work is to transfer the calls
to internal people but some
Matt wrote:
I received an e-mail from a vendor who says:
We have recently become aware of an issue in the chan_iax2
implementation of IAX2. This issue leads to degraded audio quality.
Due to this we are urging everyone to move to SIP.
I don't want to discount what this person is talling me,
All of what you say is true, but wouldn't one expect a business who has
wrapped themselves with Asterisk would be better able to provide IAX ?
One wonders about their long term viability, given this position and the
condition of their website. Broken links, and such.
JMO
John Novack
Rich
Hi,
I'm using the Pickup() application for direct call pickup having the following
line in the dialplan:
exten = _*88XX,1,Pickup(${EXTEN:2})
It works OK, though I would like to have to get the original caller ID number
forwarded to the phone where I do the pickup and have it displayed during
Hello,
I've configured ACD with ARA asterisk-1.2.4 .
I try show queues command but no queue is shown. why
?
Can I keep the caller on queue until an agent answer
the call ?
I use ARA to configure queues and members however i
have to use agents.conf to store the agents.
I wish to configure
Is there an app or softphone for meetings that displays the
hosts screen like webex or intercall.
Jordan Novak
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On Tuesday 21 March 2006 07:19, Matt wrote:
I was going to avoid naming names :P But anyway.. yes it's
asterlink. Guys seem nice enough.. and by golly.. when I switched to
SIP the termination is crystal clear... so far I'm happy with the
service from Asterlink... just wish I could use
I don't know why you'd avoid naming names. Asterlink does have good service,
and as I said they are a smart bunch of guys. I get troubles with my SIP
registrations to them on occasion but that's it. I have absolutely no
trouble recommending them to anyone.
Hi,
Wanted to avoid naming names
On 3/21/06, Jordan Novak [EMAIL PROTECTED] wrote:
Is there an app or softphone for meetings that displays the hosts screen
like webex or intercall.
http://www.btwtech.com/wipast/
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
Mazhar Hussain wrote:
Hi,
I have upgraded my PBX to Asterisk 1.2.5 , previously I was
using Asterisk 1.0.9, and Every thing was working fine ,But now
voice mail is not working. The error I am receiving in log files is
like following,
WARNING[2413] app_voicemail.c: No entry in
We have three remote call center Asterisk servers communicating with two
central Asterisk boxes over a private IP-VPN with QoS. All systems were
running Asterisk 1.0.7 communicating via IAX2 with little or no quality
issues at all.
Once we upgraded to Asterisk 1.2.4 call quality with IAX2 was
I happened to see a demonstration of WebInterpoint for Asterisk at the
Digium booth at the recent VON show, and was impressed with the
capabilities.
Cory Andrews
++
VOIPSupply.com
A Division of b2 Technologies
454 Sonwil Drive
Buffalo, NY 14225
direct -
Hi,
I have upgraded my PBX to Asterisk 1.2.5 , previously I was using Asterisk 1.0.9, and Every thing was working fine ,But now voice mail is not working. The error I am receiving in log files is like following,
WARNING[2413] app_voicemail.c: No entry in voicemail config file for '12'
I have
Hi,
Check your context you need to specify voicemail as [EMAIL PROTECTED]
(context seems to have been more tightly enforced since version 1.2 came
out). Below is an example of one of the macro I use for extensions...
[macro-stdexten]
exten = s,1,Set(DYNAMIC_FEATURES=automon)
exten =
Hi,
at my Asterisk box, I have a few of IAX2 phones (configured with
alaw/ulaw/gsm codecs, in this order) and a PRI E1 line.
In iax.conf I hav:
disallow=all
allow=alaw
allow=ulaw
allow=gsm
During some incoming call, I read at console:
-- Executing Dial(Zap/2-1, IAX2/215|20|TtwW) in new
On Tuesday 21 March 2006 09:47, Adam Robins wrote:
We have three remote call center Asterisk servers communicating with two
central Asterisk boxes over a private IP-VPN with QoS. All systems were
running Asterisk 1.0.7 communicating via IAX2 with little or no quality
issues at all.
Once we
Check this out:
http://lists.digium.com/pipermail/asterisk-users/2006-March/143394.html
When that one will work, then yours will.
On 3/21/06, Tamás Bondár [EMAIL PROTECTED] wrote:
Hi,
I'm using the Pickup() application for direct call pickup having the following
line in the dialplan:
exten
As others have told you already, RTFM. Context is what you are looking for.
On 3/21/06, Faisal Inam [EMAIL PROTECTED] wrote:
Hello All,
i am repeating this question for the sixth time but i think i was not
explaining the problem correctly. . Now i will try to explain
it..
We upgraded all five servers to 1.2.4. We tried trunking/notrunking.
End users use an IAX2 softphone on their desktop PCs. Agents are VLANed
and all IAX2 traffic is QoS'd on all LAN and WAN legs. Calls flow from
the agents to the local Asterisk server as IAX2/ulaw. Then they went
over the
Polycoms are not the best if you want a phone that works behind NAT.
On 3/21/06, Gabriel Afana [EMAIL PROTECTED] wrote:
Thanks for the response.
Yes, canreinvite is set to no on all lines.
After some testing, I was able to get sound between phones when they were
both registered to the same
On Tuesday 21 March 2006 10:55, Adam Robins wrote:
End users use an IAX2 softphone on their desktop PCs. Agents are VLANed
If there were significant changes to chan_iax2 and these were not upgraded to
match, this could explain the trouble.
Point is that it worked fine for 6-9 months before
On Tuesday 21 March 2006 10:53, C F wrote:
As others have told you already, RTFM. Context is what you are looking for.
Uh, I'm not exactly sure how contexts will help him here.
Zaptel channel groups will help him. Contexts won't do shit here unless I'm
grossly misinterpreting what he wants.
Hi all,
is it possible to bridge a call between a Junghanns quadBRI card and a
TDM400 in the same server?
It should be I think, -- I am trying this and when an incoming call comes
in, it hangs both up at the moment the bridge is attempted
(and a subsequent 'qozap: dropped audio' error is show
All switches and routers give highest priority to traffic on IAX2 port
4569. We use DSCB values over the IP-VPN to prioritize it as well.
This did not change with the upgrade, as we can still see proper packet
coding.
The softphone is provided by our vendor Aheeva. It is the same IAX2
softphone
On 3/21/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Tuesday 21 March 2006 10:53, C F wrote:
As others have told you already, RTFM. Context is what you are looking for.
Uh, I'm not exactly sure how contexts will help him here.
Zaptel channel groups will help him. Contexts won't do shit
On Tue, 21 Mar 2006 12:18:20 +0800, Ganbaa wrote
Hi
all,
I would like to use Asterisk in the E1/R2
digital
compelled enviroment. Which card is better TE210P Dual T1/E1 card or Sangoma A102U Dual T1/E1? I heard Asterisk's
Unicall add-on can support R2 signalling. But I dont have no
Are you moving your db over to an odbc connection?
Aaron
On Tue, 21 Mar 2006, ram wrote:
Hi
i have installed unixODBC Drivers and created DSN
but when i reload from * console
asterisk terminating the services
if i do again start the services
and change something and reload
again its
On Tuesday 21 March 2006 11:19, Adam Robins wrote:
All switches and routers give highest priority to traffic on IAX2 port
4569. We use DSCB values over the IP-VPN to prioritize it as well.
This did not change with the upgrade, as we can still see proper packet
coding.
Right, I wouldn't
Title: Multiple processes
Does anyone have any ideas why my recently updated * 1.2.5 system should spawn multiple * process at seemingly random intervals?
Regards
L:ee
###This message has been scanned by F-Secure Anti-Virus for Microsoft
On Tuesday 21 March 2006 11:28, C F wrote:
I disagree that it is very easy to miss, in fact even just mentioning
it, makes it very easy to not miss, becuase it's a very understandable
feature.
Well when you know what to look for everything is easy to find. :-)
How will groups without
Ha -- this looks useful
Just was trying to do a *8 on an IAXy phone...realized it didn't work
across protocols
If I implement this, I'll have to code in *8 into my extensions.conf instead
of relying on the default built in 'steal' ?
--
Chris
- Original Message -
From: Mimmus
On Tuesday 21 March 2006 10:55, C F wrote:
Polycoms are not the best if you want a phone that works behind NAT.
Are you kidding me? I used to think that anything SIP was a pain behind NAT
until I turned on 'nat=yes' for the peer/user/friend entry in sip.conf and
told the IP501 to register to
is where anyone out there having hfc-pci cards running with asterisk on
ppc-platform ?
any information on working cards, drivers, kernel, asterisk versions
would be helpful
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I'd like to use native moh instead of with mpg123... for some reason the
processes never bloody die.
For native moh to not spawn an external player, I'd need to convert the default
supplied moh sound files in /var/lib/asterisk/mohmp3 to ulaw and g729 format.
Anyone know of a free, easy way to
I'm about to start working with WiFi phones on my Asterisk
installations.
Can anyone tell me if they are using WiFi phones on wireless network
that is extended with WDS and how well the phone handles jumping from
access point to access point while on a call?
Do any WiFi phones support WPA
Whether or not a forum is a better idea isn't really depending on the
subject matter IMHO. Its success or failure depends on what the
prospective participants like better. I personally cannot stand forums.
That's a place where I have to expend energy to go there and manually
click through
Anyone have experience with the 3-08-2 release of Cisco's SIP firmware?
Are there any new features in the SIPDefault.cnf?
Thanks,
Ron
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Hi,
I disagree that contexts are not for outgoing calls, how else do you
restrict certain user to local calls only without using contexts?? On
the subject of grouping extensions I use pickup groups so that any
person can answer any phone in their immediate area by using a '*8' (as
long as
On Tuesday 21 March 2006 11:28, C F wrote:
I disagree that it is very easy to miss, in fact even just mentioning
it, makes it very easy to not miss, becuase it's a very understandable
feature.
Well when you know what to look for everything is easy to find. :-)
How will groups without
I didn't say it doens't work, I said it's not the best, and if you
want I'll repeat myslef, Polycoms are not the best behind NAT, Cisco,
or SPAs are much better. Just because you didn't run into any problems
doesn't mean that it works well with all NAT devices.
On 3/21/06, Andrew Kohlsmith [EMAIL
Andrew Kohlsmith wrote:
On Tuesday 21 March 2006 10:55, C F wrote:
Polycoms are not the best if you want a phone that works behind NAT.
Are you kidding me? I used to think that anything SIP was a pain behind NAT
until I turned on 'nat=yes' for the peer/user/friend entry in sip.conf and
On 3/21/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Tuesday 21 March 2006 11:28, C F wrote:
I disagree that it is very easy to miss, in fact even just mentioning
it, makes it very easy to not miss, becuase it's a very understandable
feature.
Well when you know what to look for
There are a number of phones that support WPA, including the UTStarcom
F1000G and F3000, and the Linksys WIP300 and WIP330, although current
availability on these products is scarce.
I know a couple of integrators that are having good success on WIFI
deployments using the D-Link DWS-1008 (8
Hi,
I use the Zyxel P-2000W v2 wireless VOIP phones with Zyxel G-1000
access points and the hand off calls fairly smoothly using a port for
the hand off and using WEP security (the Zyxel is not capable of WPA
security yet). I understand that people have problems with some
manufactures
Hello,
I installed 1.2.5 and realtime SIP. The connection
to the DB is OK
because I can get the values from the
CLI.
Here are my 3 different cases:
1- If I put an unexisting user, I get 404 and I am
not able to dial.
2- If I check "Disable registration" within Firefly
itdoes
Charles Marcus wrote:
Whether or not a forum is a better idea isn't really depending on the
subject matter IMHO. Its success or failure depends on what the
prospective participants like better. I personally cannot stand
forums. That's a place where I have to expend energy to go there and
On Tuesday 21 March 2006 12:07, Chuck Bunn wrote:
I disagree that contexts are not for outgoing calls, how else do you
restrict certain user to local calls only without using contexts?? On
Quite simply: The SIP user has a 'context' field. When a call comes *in* to
Asterisk from that SIP
Yeah, I agree with Chuck. User's on our system are put into various
contexts depending on who they can call... local, long distance, or
internal only.
Aaron
On Tue, 21 Mar 2006, Chuck Bunn wrote:
Hi,
I disagree that contexts are not for outgoing calls, how else do you restrict
certain
I think that you are sending an outgoing caller id that is not part of the
DID range. Most operators do not allow this.
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1013' ]
Are you using caller id 1013 ?
Change it to a number that is part of your trunks.
Henk
-Original Message-
Hi,
I am using the following architecture:
SER (SIP server) -- Asterisk -- PSTN Gateway
And I would like to implement a prepaid billing solution that could be controlled by asterisk. Can anyone give me a direction?
Jose Simoes
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the DB in same server
but iam using DSN to connect
using ODBC
is the not th right proces
if not kindly recomend me the process
i want to both SIP users / CDR to be from Mysql
ram
On 3/21/06, Aaron Daniel [EMAIL PROTECTED] wrote:
Are you moving your db over to an odbc connection?AaronOn Tue,
At 01:50 AM 03/21/2006, you wrote:
i am repeating this question for the sixth time but i think i was
not explaining the problem correctly. . Now i will try
to explain it..
For each line in Zapata.conf make an entry like:
context=line1
group=1,9
channel = 1
context=line2
On 3/21/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I'm about to start working with WiFi phones on my Asterisk
installations.
Can anyone tell me if they are using WiFi phones on wireless network
that is extended with WDS and how well the phone handles jumping from
access point to access
Is it possible to store voicemail recorded messages using odbc?
Fernando Lujan
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I have only been able to confirm this issue with my IAXy as it is the
only ATA I have.
I am running 1.2.5 stable.
Example:
User @ exten 100 (iaxy) receives call from PSTN (call 1).
While on the call, another user from PSTN (call 2) calls 100 (iaxy)
sending a call waiting beep to the iaxy.
On 21 Mar 2006, at 13:21, Rich Adamson wrote:
Add to that the fact that iax is actually a proprietary protocol
implementation (eg, not based on any current published/approved
standards), and the fact that only folks that run asterisk actually
use the protocol, you now have a fairly
On 21 Mar 2006, at 16:19, Adam Robins wrote:
All switches and routers give highest priority to traffic on IAX2 port
4569. We use DSCB values over the IP-VPN to prioritize it as well.
This did not change with the upgrade, as we can still see proper
packet
coding.
The softphone is provided
Ready
to scream here..
1.
After 6 months with Asterisk I'm STILL trying to understand the difference
between a SIP user, friend and peer.
2.
Exactly what resource does Asterisk use to send MWI to registered phones? I
thought it was astdb?
3. It
looks like it isn't astdb. It looks like
On Tuesday 21 March 2006 12:25, Aaron Daniel wrote:
Yeah, I agree with Chuck. User's on our system are put into various
contexts depending on who they can call... local, long distance, or
internal only.
And *all* of those people are placing calls *in* to asterisk to get into those
contexts.
On Tuesday 21 March 2006 12:14, C F wrote:
Of course contexts are for outgoing as well, how else is he going to
make sure that device a only dials out using channel/group x?
No, the dialplan determines what you do. I.e. you get to an appropriate
Dial() command which specifies the appropriate
On Mon, 2006-03-20 at 11:51 +0100, Sébastien Mortier wrote:
Hello,
I recently bought a Junghanns Octobri Card. I have some problems with
this card to make outbound calls but I can receive calls.
I have 3 lines to PSTN and 3 lines to my existing PBX
FRANCE TELECOM -- OctoBRI --
On Tue, 21 Mar 2006 12:56:13 -0500, Fernando Lujan
[EMAIL PROTECTED] wrote:
Is it possible to store voicemail recorded messages using odbc?
Fernando Lujan
see asterisk-sources/doc/README-odbcstorage
--
Justin Tunney
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Tuesday, March 21, 2006 11:36 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Problem with
chan_iax.cimplimentationcausesbadaudio?
On Tuesday 21 March 2006
Hi all,
Can someone share with me his experience in making asterisk-oh323 talk to
quintum gateway without gatekeeper.
My set up is QUINTUM GATEWAY --IPM ASTERISK (OH323)
Both are gateways.. but I don't know what authentication I will set up in
oh323.conf and how to set it up
I will
Hi all,
Can someone share with me his experience in making asterisk-oh323 talk to
quintum gateway without gatekeeper.
My set up is QUINTUM GATEWAY --IPM ASTERISK (OH323)
Both are gateways.. but I don't know what authentication I will set up in
oh323.conf and how to set it up
I
这个名单是英文. 这是我讲的一切.
Jeffery Chen wrote:
真的很遗憾。不管左派的网友还是右派的网友，在谈到计划生育的时候大都会摆出
一副冷酷的面孔。我就来说说计划生育是个什么东西。
坐在电脑面前的精英们应该知道这么一个国情常识：中国的农民是没有任何退
休金和任何形式的医疗保障的。
你们有没有想过，他们如果没有一个强有力的孩子，当他们失去劳动能力的时
候，就只能坐在家里慢慢饿死？死并不可怕，对于中国农民来说，每年的非正常死
亡不计其数：有死在矿井里的，有死在城市的工地上的，有死在收容所里的，有洪
the cisco 7920 with the latest firmware supports WPA-psk using the AKM
for auth.
it is important to turn off CDP discovery otherwise it will crash other
cisco sccp phones connected to asterisk - advaned menu: Menu, *, #, #,
Send (green phone icon) - network config and disable cdp tx
haven't
Thanks for the offer. We deleted all of our Ethereal traces once we
switched to SIP. On a bad call call there were tens of thousands of
checksum errors and packets out of sequence. This occurred both with
and without IAX2 trunking and trunktimestamps.
Complaints of poor quality were from both
If you're using a mysql backend, it may be simpler to use res_mysql to get
to it since it's designed specifically for use with mysql. Just change
your configuration file extconfig.conf and change everything that says
odbc over to mysql, and use the cdr_mysql plugin for the mysql connection.
I've been using realtime for sip users information.
I noticed that when you are doing this, if you do a 'reload' or restart
asterisk, the information in a 'sip show peers' goes away. When I do this, MWI
stops working. I always though MWI used the astdb file ('database show') to
determine where
On 3/21/06, Dave Cotton [EMAIL PROTECTED] wrote:
span=1,1,3,ccs,ami
I was going mad with a Quadbri until I changed ccs,ami to ccs,hdb3 and
it's been running 6 months now.
Dave, nice to read on this, can you explain what was going wrong when
you used ccs,ami? And how did you find out about
Hi,
I have 2 asterisk boxes connected both through internet with 8Mbit via IAX
trunking. In asterisk A I have one TE410P card with one E1 active and I
receive calls from PSTN and send them to asterisk B.
In asterisk B I have other TE410P and one port is connected to one TOPEX
GSM Gateway for
Voipers Portugal wrote:
Hi,
I am using the following architecture:
SER (SIP server) -- Asterisk -- PSTN Gateway
And I would like to implement a prepaid billing solution that could be
controlled by asterisk. Can anyone give me a direction?
I use this setup with Asterisk2Billing
Peter Fern wrote:
I've had the same problem with all boxen running the same version. We
ditched IAX2 for SIP and it has been working fine since.
Well, upgrading my remote site to 1.2.5 appears to have fixed my issues.
-Barry
Doug Lytle wrote:
Barry Flanagan wrote:
Hi,
I have a
Andrew Kohlsmith wrote:
These are *all* incoming calls as far as Asterisk is concerned. You get
dumped into a specific part of the dialplan (the context specified) and you
tell Asterisk what they can dial. Internal extensions, external peers, Zap
channels or even applications... the second
Douglas Garstang wrote:
I've been using realtime for sip users information.
I noticed that when you are doing this, if you do a 'reload' or restart
asterisk, the information in a 'sip show peers' goes away. When I do this,
MWI stops working. I always though MWI used the astdb file
Try googling the archives using the keywords rtcachefriends mwi.
You should find more info about this.
regards,
David
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Andrew Kohlsmith wrote:
On Tuesday 21 March 2006 12:25, Aaron Daniel wrote:
Yeah, I agree with Chuck. User's on our system are put into various
contexts depending on who they can call... local, long distance, or
internal only.
And *all* of those people are placing calls *in* to asterisk to
Is it possible to use a2billing for everything except the dial out, as I
want to use a2billing, to auth users and log time but I want to added custom
IVR menus after users log in, like custom speed dial numbers. A2billing
allows you to dial out no problem, but how do I get it to drop back to the
Brian Capouch wrote:
Andrew Kohlsmith wrote:
These are *all* incoming calls as far as Asterisk is concerned. You
get dumped into a specific part of the dialplan (the context
specified) and you tell Asterisk what they can dial. Internal
extensions, external peers, Zap channels or even
David Thomas wrote:
Try googling the archives using the keywords rtcachefriends mwi.
You should find more info about this.
Google doesn't work anymore; the subjects are listed just fine, but
clicking on one leads to page not found.
___
Was it difficult to install and make it work? I need to make it work in
a week or so, do you think it's possible? Did you manage to work with
SER already? Because i don't see how can I distinguish the users
because all of them come from SER, and don't Register directly into
Asterisk.
Jose
I do have rtcachefriends=yes in sip.conf, and my astdb file is full of sip
contacts.
That's not the problem.
-Original Message-
From: Barry Flanagan [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 21, 2006 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
I have rtcachefriends=yes in sip.conf.
It is caching friends because as I said in my post, astdb has all the contacts,
ie they're cached.
It's the behaviour of 'sip show peers' that's not working.
-Original Message-
From: David Thomas [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 21,
On 21 Mar 2006, at 18:35, Adam Robins wrote:
Thanks for the offer. We deleted all of our Ethereal traces once we
switched to SIP. On a bad call call there were tens of thousands of
checksum errors and packets out of sequence. This occurred both with
and without IAX2 trunking and
Hi guys,
I'm using SIP phones with Asterisk 1.2 and going fine, most of the time.
However, when the duration of a call is longer than 20minutes I often
stop hearing the other party, but that one keeps most of the time
hearing me. Does any of you know of this or similar problems?
Thing is that
I don't know for sure about the formats, but I'd try sox. I'm pretty
sure pcm/ulaw is built in...
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Tuesday, March 21, 2006 11:03 AM
To: Asterisk Users Mailing List -
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