Displaying 20 results from an estimated 3000 matches similar to: "Asterisk and CIsco 1760 SIP ?"
2003 Sep 29
2
cisco AS5300 : problem configuration
I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ...
I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly
2009 Oct 15
2
Asterisk with a Cisco AS5300 gateway
Hi
i test a new equipment on my backbone: a Cisco AS5300 with voice dsp
ressource
connected at a E1 Voice Link.
I want that all call incoming on the cisco 5300 are sent to Asterisk and
all Asterisk outgoing
call are sent to Cisco AS5300.
Actually, i configure the AS5300:
isdn switch-type primary-net5
!
voice service voip
sip
!
voice class codec 400
codec preference 1 g711alaw
codec
2006 Nov 18
2
AdvancedVoIP Billing ?
Hi
after 2 mounth of search, i don't have see a billing solution
for my small business..
i see only AdvancedVoIPBilling but i don't know if he can work's with
Asterisk.
I am search a billing software for the invoice of my custumers, no
Calling Card.
but i don't see a small and simple product for this.
thanks bye
2010 Aug 29
1
evil disconnect of call with cisco 1760
I have asterisk 1.6.2.11 talking to a cisco 1760 running
ipvoicek9-mz.124-25b.
whenever a call goes through the 1760's FXO or FXS (in or out) there is
a 915 second maximum call time due to asterisk hanging up the call
because of a "critical packet" being missed.
I read doc/sip-retransmit.txt and I don't see anything there that is
helpful to my situation - the asterisk box is
2012 Sep 24
3
"Out of Memory error
Hi
we have a Samba-Winbind 3.5.10 and i have a big quantity of errors
in logs :
[2012/09/24 16:01:29.248286, 1]
winbindd/winbindd_ads.c:728(lookup_usergroups_memberof)
lookup_usergroups_memberof ads_search
member=CN=TESTDD,OU=Villeurbanne,OU=ExEA,DC=ELIOTT,DC=fr: Out of memory
[2012/09/24 16:03:38.498533, 1]
libads/ldap_utils.c:323(ads_ranged_search_internal)
could not pull first
2005 Mar 19
3
Asterisk and Cisco AS53xx/54xx Access Server Platform
Hello,
I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in
turn is talking to an Asterisk server via SIP for call origination and
termination. Seems simple enough, and it works for the most part,
but:
1) Caller ID name data comes in on the PRI, but doesn't appear to get
handed off to the Asterisk server via SIP, at least not in any
format that Asterisk
2004 Nov 29
1
Cisco gateway help needed
HI,
I have been pulling my hair out trying to get a Cisco MC3810 to interface my
Asterisk box with a T1.
I am able to make outgoing calls but incoing calls never reach my Asterisk
box. The cisco give a fast busy when I try to call one of the DID's. When
playing around with the dial-peers I can get the cisco to pick up the call,
but then it forwards the call back to the ANI that is dialing.
2009 Oct 31
2
Asterisk, Realtime and specify MySQL Table Name ?
Hi
actually, i test a new Asterisk Server and i want add Mysql Realtime SIP.
I read on the wiki:
===================================================
Database Config
put the following in res_mysql.conf
[general]
dbhost = 127.0.0.1
dbname = asterisk
dbuser = myuser
dbpass = mypass
dbport = 3306
Values in sip.conf or iax.conf like in older versions of * are no longer
used.
Database Table
Lets
2010 Oct 01
2
Asterisk/Realtime and MySQL
Hi
i am not a expert on Asterisk and search a lot of small information :
I use Asterisk 1.6.1.4 with MySQL.
That's work and in my extension.conf, i have:
[as5300-incoming]
switch => Realtime
and in extconfig.conf
extensions => mysql,general,VOIP_Extensions
A lot of Extension are into the table VOIP_Extensions.
I am search to know if i can add a :
2009 Nov 14
2
Error Dialplan ?
Hi
I have a problems with a new Asterisk Server,
when i want call, i have:
[Nov 14 09:12:38] NOTICE[31992]: chan_sip.c:18160
handle_request_invite: Call from 'PHISIP000001' to extension
'00420225352184' rejected because extension not found.
but into my extensions.conf:
exten => _00420X.,1,Set(CDR(CodeTier)=CZE)
exten =>
2014 Nov 20
1
Asterisk problems
Hi
I have a problem with Asterisk 11.5.1.
When I pick up an incoming phone call sometimes I need to transfer to
someone else in the organization.
I then dial a number on my phone, and press Xfer.
Sometimes it works well, I mean, the number I dialed get the call and
can chat with correspondent.
Sometimes, the number I dialed get the communication and while he chats
with correspondent, gets a bip
2009 Oct 28
2
Asterisk/Cisco AS5300 => Two problems in incoming (extension not found)
Hi
Now, my Cisco AS5300 sent call to my asterisk, but two problems:
When i call the phone number, i have:
[Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite:
Call from '' to extension '0426000000' rejected because extension not found.
[Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite:
Call from '' to extension
2008 Jun 20
1
Voice only works from one way.
Hello, everyone.
Right now, we are trying launch our own PBX system based on Asterisk(Fedora)
with Cisco 2611. Cisco has 2 port FXO card installed on it.
For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx
fine. (I'll call it F). Using softphone, I can dial in extension 1001 on
asterisk, which should talk to cisco. After initial connection to Asterisk,
I have try to
2013 Oct 07
1
IAX and Variables
Hi
a new small question ;=)
We have two Asterisk, connected in IAX2.
On the first, in dialplan, we have:
exten => _XX.,1,Set(IAXVAR(ACCOUNTID)=${CDR(accountcode)})
we sent into the IAXVAR "ACCOUNTID" the accountcode.
On the second, in dialplan, we have:
exten => 18,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)})
That's work, the second server get the variable.
I
2013 Sep 27
1
Realtime Mysql
Hello,
I am looking to know if it is possible to modify the SQL query that is
on Realtime sip accounts.
I want multiple servers use the same sql table, so getting an extra
"server" field to indicate that the data is valid on the X server
is this possible?
thank you in advance
jerome
2009 Nov 20
1
I don't know how to authenticate
Hi
anyone know what is a the solution of this problems ? :
[Nov 20 10:15:40] WARNING[12049]: chan_iax2.c:9232 socket_process: I
don't know how to authenticate Voip-Classic to
[Nov 20 13:04:45] WARNING[12043]: chan_iax2.c:9232 socket_process: I
don't know how to authenticate Voip-Classic to 78.XX.XX.XX
we have two Asterisk 1.6.1.4, this error are on the first server, used
for the
2009 Mar 20
1
Winbind error ? idmap Fatal Error: UID range full!
Hi
anyone know this error:
Mar 20 12:01:06 gw winbindd[14756]: [2009/03/20 12:01:06, 0]
sam/idmap_tdb.c:db_allocate_id(106)
Mar 20 12:01:06 gw winbindd[14756]: idmap Fatal Error: UID range
full!! (max: 20000)
Mar 20 12:01:06 gw winbindd[14756]: [2009/03/20 12:01:06, 0]
sam/idmap_tdb.c:db_allocate_id(106)
Mar 20 12:01:06 gw winbindd[14756]: idmap Fatal Error: UID range
full!! (max: 20000)
2006 Apr 04
2
Distinctive Ring on SPA941
Does anyone know how to set the distinctive ring on the Linksys SPA941?
I want to be able to dial one extension and have the phone ring with a
certain tone and then dial another and have the phone ring with a
different tone. I have tried the following
-------------------------------------------------------------------
exten => 802,1,SIPAddHeader(call_info=Classic-4)
exten =>
2009 Nov 21
3
Connect two Asterisk Server in IAX ?
Hi
My first post get no answer :=<, i post new with new elements.
I have two Asterisk server, running on Asterisk 1.6:
SRV1 = 192.168.0.5 on Asterisk 1.6.1.4
SRV2 = 192.168.0.20 on Asterisk 1.6.1.8
I want create a link for exchange call.
on Srv1:
iax.conf:
[general]
bindport=4569
bindaddr=0.0.0.0
language=fr
bandwidth=low
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
2008 Mar 19
1
Getting config from SPA-941 or 942 phones
Hi,
Is it possible to get the XML config off of a Linksys SPA-941 or 942 phone?
I've tried http://[ip address]/admin/spacfg.xml however that file
doesn't appear to exist.
Thanks.