Hi,
Call getting silenced in the middle definitely point to RTP but I think
the redialling part should be considered as well. I think that Phones are
loosing registrations or like Zeeshan mentioned could be getting blocked by
firewall - Asterisk server's firewall as well as any other firewall in f
> I do not know anything about 10.0 but 1.6.2 problem most likely can be
> fixed by a simple patch which is not being committed for unknown
> reason
> since late August 2011.
>
> https://issues.asterisk.org/jira/browse/ASTERISK-18301?focusedCommentId=183734#comment-183734
1.6.2 is in security fix
-- Forwarded message --
From: michael k
Date: Wed, Oct 19, 2011 at 10:43 AM
Subject: Outgoing call failure
To: Asterisk Users Mailing List - Non-Commercial Discussion <
asterisk-users@lists.digium.com>
Hi List,
My all incoming calls are working fine but i cant make ou
Hi List,
My all incoming calls are working fine but i cant make outgoing
calls. There was no issues for both incoming and outgoing calls till
yesterday. Can somebody tell me what is the issue is ?. I have enable the
dibugging in pri line by issue the command "pri set debug on span 1"
[trunkgroups]
[channels]
[my-phones](!)
usecallerid = yes
hidecallerid = no
callwaiting = yes
usecallingpres = yes
callwaitingcallerid = yes
threewaycalling = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
echocancel = yes
echocancelwhenbridged = yes
relaxdtmf = yes
rxgain
I think this might actually be a bug.
https://issues.asterisk.org/jira/browse/ASTERISK-18137
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Lists
Sent: Tuesday, October 18, 2011 9:23 PM
To: asterisk-us
@bakko
I do not know anything about 10.0 but 1.6.2 problem most likely can be
fixed by a simple patch which is not being committed for unknown reason
since late August 2011.
https://issues.asterisk.org/jira/browse/ASTERISK-18301?focusedCommentId=183734#comment-183734
-Vladimir
On 10/18/2011 6
Hi,
After a make menuselect I now have menuselect.makeopts,
menuselect.makedeps and menuselect-tree.
How do I get the buildsystem to use the settings in those files? Thus
far they just seem to get overwritten if I do:
$ cd asterisk-1.8.8.0-rc1
$ cp -v /tmp/menuselect* .
$ ./configure
$ make
Dear;
By the way, the asterisk version that I have is 1.8.4.2 and DAHDI version is
2.4.1.2
Here I would like to mention the following:
1) As per the telecom provider, they said they openned for us all the digits to
send (two digits, 4 digits, all the digits ...) as they said.
2) The caller i
Hello,
Is there any issue with gtalk module?
Whent I try to call asterisk gtalk user nothing happens on the asterisk
console. Asterisk 1.6.2.20
With Asterisk 10.0.0 beta 2 and the same configuration, works.
???
Thank you
Regards
--
Hi Tarek,
Yes, after running some more detailed packet captures, it seems that
the SDP sent has the sendonly media attribute. I do not know if it is
the Sonus switch, but the problem is identical to yours.
Unfortunately setting canreinvite=yes for that peer does not solve the
problem. I am guessi
> from the
> > > telecom starts
> > > > >> from 1030 and end by 1059, now
> whenever we
> > > place a call, the
> > > > >> destination see the number 5631030.
> I gave
> > > the phone
> > > > >> extensi
+1 for you Andrew - easiest fix I've had for Asterisk in a while.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham
Sent: Tuesday, October 18, 2011 4:30 PM
To: Asterisk Users Mailing List - Non-Comm
On Tue, Oct 18, 2011 at 6:21 PM, Danny Nicholas wrote:
> Hi gang,
>
> We are moving our 1.4 asterisk with DAHDI over to 10.0 with
> SIP. Everything is going nicely except that I can’t get NV_FAXDETECT to
> compile properly into 10.0. Because of this, I will have to have my
> rece
Hi gang,
We are moving our 1.4 asterisk with DAHDI over to 10.0 with
SIP. Everything is going nicely except that I can't get NV_FAXDETECT to
compile properly into 10.0. Because of this, I will have to have my
receptionist manually transfer incoming faxes. Any suggestions?
Th
Thank you for the reply.
The Asterisk is behind a firewall, but not in a dmz, been thinking of placing
it in a dmz soon, maybe that will solve the problem.
Or else, I will try your guide with wireshark.
Thank you very much.
Best regards
Aksel
Fra: asterisk-users-boun...@lists.digium.com
[m
> On 11-10-16 01:51 AM, Michael C. Robinson wrote:
> > [Oct 15 22:44:31] ERROR[29013] res_config_pgsql.c: PostgreSQL
> > RealTime:
> > Failed to connect database asterisk on 127.0.0.1:
> > [Oct 15 22:44:31] WARNING[29013] res_config_pgsql.c: PostgreSQL
> > RealTime: Couldn't establish connection. C
On 11-10-16 01:51 AM, Michael C. Robinson wrote:
[Oct 15 22:44:31] ERROR[29013] res_config_pgsql.c: PostgreSQL RealTime:
Failed to connect database asterisk on 127.0.0.1:
[Oct 15 22:44:31] WARNING[29013] res_config_pgsql.c: PostgreSQL
RealTime: Couldn't establish connection. Check debug.
[Oct 15
Hello everybody, sorry for delay
Le 16/10/2011 16:51, Tarek Sawah a écrit :
One more thing can you post your peer's configs as you have it in the
config file?
It's below, at the end of the original message. Tried as well type=peer
with no luck
Details:
[snom320](!)
type=peer
host=dynamic
c
On 10/18/2011 07:27 AM, bilal ghayyad wrote:
make progdocs, we use it to create documentation, correct?
Documentation for Asterisk developers (programmers), yes. That's why it
is called 'progdocs' (programmer documentation).
Well, how I can use this documentation if I need to search for a
@Tzafrir Cohen:
Thanks for your reply. Yeah, I'm through the source and online documentation,
it seems AEL is one of the ways that one can write dialplans. But, the default
dialplan is written in extensions.conf.
Thanks for mentioning pbx_conf.c. I'll examine that tomorrow.
I want to know the
thanks for your response
itry this but i didn't recive any email,also if there is a way to recive a
SMS in my mobile 0678XX
regards
2011/10/18 Ishfaq Malik
> On Tue, 2011-10-18 at 12:26 +, salaheddine elharit wrote:
> > hello list
> >
> >
> >
> > i have configured the voicemail in my
On Tue, 2011-10-18 at 12:26 +, salaheddine elharit wrote:
> hello list
>
>
>
> i have configured the voicemail in my server asterisk 1.4 i can use it
> without issue ,i have a question
>
>
>
> i want to receive an email in my address email when there is no
> response from 270 after 10 s
I can only make another guess. If your system is behind a firewall, try
adding 'insecure=invite' in your sip.conf's general section.
To troubleshoot such cases, do a tcpdump trace like this:
1. Run tcpdump on your server before making a call. Use command "tcpdump
port 5060 -s0 -w dumpfile.pcap
Hi all,
Just hit this problem for the first time:
WARNING[17712] chan_iax2.c: Too much delay in IAX2 calltoken timestamp
from address 10.25x.xxx.160
When I ran "iax2 show peers" everything comes up as unreachable, no
calls are passed between the servers (as would be expected) but there
is no prob
Thank you for replying
My sip.conf is set to no on canreinvite
[general]
context=default
allowguest=yes
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
pedantic=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
disallow=all
allow=alaw
;allow=ulaw
;allow=gsm
language=en
trustrpid = yes
s
I have similar problem at my home extension, but for that I know my
phone's speaker is defective, and tapping it against the desk or wall
fixes the problem.
However in your case probably it is sip configuration (sip.conf or an
included file), where canreinvite=yes where it should be canreinvit
Hello dear list.
We run a Asterisk 1.6.2.6 on testbasis (SIP), and experience every day, when
making calls, that the calls become silent.
Not every calls, but 1 out of 3-4 calls, becomes silent suddenly during the
conversation.
When we then hangup, and redial immediately, the calls do not go thr
On Tuesday 18 October 2011, bilal ghayyad wrote:
> We contacted the Telecom provider and they confirmed multiple times that
> the DID service is enabled, but again still the caller id does not appear
> as we need (it is always appearing as the primary number). I tried to set
> the CALLERDID(num) to
hello list
i have configured the voicemail in my server asterisk 1.4 i can use it
without issue ,i have a question
i want to receive an email in my address email when there is no response
from 270 after 10 s
could you please verify the code below and tell me what is wrong
thanks and reg
Dear All;
make progdocs, we use it to create documentation, correct?
Well, how I can use this documentation if I need to search for a topic or
settings? Any example?
Regards
Bilal
--
_
-- Bandwidth and Colocation Provided by h
Bialal,
what hw you are using and what is the h/w files. ?
On Tue, Oct 18, 2011 at 5:23 PM, bilal ghayyad wrote:
> Dears;
>
> We contacted the Telecom provider and they confirmed multiple times that
> the DID service is enabled, but again still the caller id does not appear as
> we need (it
Dears;
We contacted the Telecom provider and they confirmed multiple times that the
DID service is enabled, but again still the caller id does not appear as we
need (it is always appearing as the primary number). I tried to set the
CALLERDID(num) to be 40, 1040, 5631040 and 065631040 without an
Hi,
I'd been thinking about such a huge conferencing system for about last few
months. Like Steve suggested, my concept is almost similar but instead of
making a central hub conference junction between multiple Conferences I was
thinking of making a peer2peer runtime connection between conferences
On Tue, Oct 18, 2011 at 12:08:04AM -0700, Sazzad Kamal wrote:
> Hello,
>
> I want to know the way Asterisk parses configuration files. So that I can
> find out, what Asterisk does when I write, 'allow=g711' in extensions.conf
> file.
>
> As I've come to know, Asterisk configuration files are ki
Hi folks,
Please discard the e-mail since it seems to be a problem of the E1 provider
and nothign related to asterisk.
Apologies for the noise,
Samuel
On 17 October 2011 18:32, samuel wrote:
> Hi folks,
>
> I'm having an issue with an asterisk 1.4.36 with an E1 card that is not
> forwarfing th
I see, it is pbx_ael.c . But I'm unable to see the 'allow=codec' part any where.
From: Sazzad Kamal
To: "asterisk-users@lists.digium.com"
Sent: Tuesday, October 18, 2011 1:08 PM
Subject: How does Asterisk parse extensions.conf file?
Hello,
I want to know the
Hey,
I don't think you are doing it right. The memebers/channels you need to spy
should be added in SPYGROUP and not the channel which is spying. i.e your
code maybe something like this.
exten => 4368,1,Answer()
exten => 4368,n,NoOp(${CHANNEL})
exten => 4368,n,Set(SPYGROUP=my-g
Hi list,
I have write down my code on which chanspy not working when I make a group
with name of spy. Please help me where is the issue on that.
a) caller will call this number to join konference and spy group
exten => 4368,1,Answer()
exten => 4368,n,NoOp(${CHANNEL})
exten => 436
Hello,
I want to know the way Asterisk parses configuration files. So that I can find
out, what Asterisk does when I write, 'allow=g711' in extensions.conf file.
As I've come to know, Asterisk configuration files are kind of DSLs. I thought
it must have a parser to know the values in those conf
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