Displaying 20 results from an estimated 100 matches similar to: "SIP INVITE and caller id / proxy-authorization strange behaviour"
2005 Mar 07
0
chan_sip not 100% RFC3665 compliant - re-REGISTERs fail.
Greetings,
For the past 2 months I've been struggling with registration problems
with asterisk+external FXS/FXO gateways (www.addpac.com) that use
RFC3665 re-registration procedure.
This problem occured for devices with more than one FXS port with a set
non-empty password.
Those gateway attempt to re-register after the initial register timeout
period expires fully compliant with
2007 Jan 15
0
Addpac 2620 don't relay DTMF to PSTN
Hi Guys:
I'm using Asterisk with Addpac 2620 as gateway, internally I'm using
Grandstream BT200, unfortunately when I called to external phones (PSTN),
and I have to choose some extensions, the Phone don't dial the extensions, I
believe that DTMF relay in ADDPAC is not working well. I'm using RFC 2833
and ALaw for SIP Channel (Between ASterisk and ADDPAC). Someone have any
2008 Mar 06
1
OT: Upgrade Addpac AP200C
Hi guys,
I have made a upgrade to my addpac ap200c, however it does not upload
complete, now I can load addpac. Is there anyway that can I upload the old
firwmare? Any help is appreciated.
System Boot Loader, Version 2.2.5/DUAL(852)
Copyright (c) by AddPac Technology Co., Ltd. Since 1999.
System Bootstrap, Version 1.2
Decompressing the image:
#
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2004 Apr 14
0
ADDPAC 200 w/SIP
I am trying to connect an ADDPAC 200 to * using SIP protocol. Problem is the documentation for the ADDPAC is not very helpful (they don't mention SIP at all, seems to be a newer firmware).
I think I got the * part working but the ADDPAC setup is the problem, I will appreciate if somebody can post a working example or point me to documentation in english (found stuff in russian and korean so
2007 Oct 14
1
difference between FXO interfaces !
Hello everybody,
Which one is a better choice
1. Gateway device with FXO <-> SIP ( example Addpac
http://www.addpac.com/addpac_eng2/addpac_product_view_detail.php?class_id=19&item_id=59
)
2. Digium (Wildcard TDM400P)
3. Sangoma (A200 Analog FXO/FXS)
All i need is to put asterisk in place with 4-8 incomming lines (ordinary POTS ).
With IVR, Voice mail and International Call via SIP.
2004 Sep 23
0
Duplicated INVITE in SIP session?
Hello.
I'm trying to use Asterisk in combination with SER, to make the
routing proccess to my PSTN-Gateways. I made a simple test defining some
extension in my extension.conf, when i made a call my SER (SIP) Server
forward the call to Asterisk, this proccess is ok, but when the call is
answered i see an INVITE going out from Asterisk to my SER Server, this
invite is then passed to my
2009 May 15
1
Fax t38 capability
Dears I installed digium fax and followed the instruction at
http://downloads.digium.com/pub/telephony/fax/README,And as you can see
above that t38 is loaded
I am using a call file to send fax1.tif file as fax to the gateway named
add
The problem that Addpac send always Receive 488 Not acceptable here,and
lkindly find my debug attached
Please advice.
Thanks I Advance
shark*CLI>
2005 Mar 07
1
chan_sip not 100% RFC3665 compliant - re-REGISTERsfail.
asterisk-users-bounces@lists.digium.com wrote:
> Is there anyone else with the same problem?
Yes, we've seen the same problem. We have found a work
around, but I'm unable to to look into it today.
--
Andreas Sikkema Rits tele.com
Van Vollenhovenstraat 3 3016 BE Rotterdam
t: +31 (0)10 2245544 f: +31 (0)10 2245540
2005 Sep 07
1
presence settings and Eyebeam
What is the proper way of adding hints to multiple extensions?
In my case extensions are the same as the sip usernames, while as per
http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence
exten => 1234,hint,SIP/1234 works,
exten => _1XXXX,hint,SIP/${EXTEN} doesn't. Not sure if I can even use
${EXTEN} here...
Any hints?
Vahan
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2005 Jul 14
2
CVS HEAD voicemailbox full error
Anyone else has problems with CVS HEAD's from today with voicemail
hanging up silently without any debug/error messages when checked?
It also keeps insisting that the user's voice mailbox is full and can't
store more messages even if I clear/rebuild the
/var/spool/asterisk/voicemail stuff.
I've tried falling back to voicemail.conf entries from realtime
voicemail with the same
2005 Jul 06
3
OT: Congrats, Europe!
http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150&tid=147&tid=136
http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/
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2004 Jan 22
2
MGCP Problem.
Hi.
I'm new in Asterisk with MGCP. I set up a MGCP user agent and start asterisk
with the next configuration files.
'--------------- extensions.conf
----------------------------------------------------
[general]
static=yes
writeprotect=yes
[globals]
ap1 => mgcp/aaln/ap200@64.76.148.186
[macro-apl1]
exten => s,1,Dial(${ARG1},30,Ttmr)
;exten => s,2,Voicemail(u${MACRO_EXTEN})
2009 Oct 14
2
FXS to SIP gateway
Hello list !
I don't have the money to test out all the products and reading the
manuals is not always that enlightening...
Does someone here know a good gateway-product that lets analogue
telephones communicate with an Asterisk-server.
I have found the Grandstream GXW-400x to be able to add SIP-accounts to
analogue telephone devices that are connected to the FXS-ports. Moreover
this
2004 Dec 02
10
Conference
Good Morning,
I would like to know if is possible to do a conference with 9 client
with asterisk.
The client is connecting to sever through lan, we think don't use PSTN
or ISDN.
Thanks, Alberto
--
Alberto Carlana <alberto.carlana@virgilio.it>
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2012 Jan 13
1
LSI/3ware 9750-4i and multipath I/O
Hi,
I was wondering if anyone has successfully configured two lsi/3ware 9750-4i series controllers for multipathing under CentOS 5.7 x86_64?
I've tried some basic setups with both multibus and failover settings, and had repeatable filesystem corruption over a iscsi(tgtd) or nfs3 connection.
Any ideas?
Vahan
2004 Sep 26
2
Asterisk <-> WellGate 3502a : ulaw/alaw only?
Greetings,
I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought
several WellGate 3502A FXSes to play with till welltech guys fix the
3504a's registration bug.
So far everything is working as expected, except the fact only ulaw and
alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's
ports entries in the sip.conf, no voice is heard from both
2004 Sep 26
1
voicemail /w asterisk - voicemail() problems
I've setup the voicemail that auths against the mysql db. Now,
everything works ok, except voicemail() calls fail with
Sep 26 18:09:34 WARNING[157070336]: app_voicemail.c:1517
leave_voicemail: No entry in voicemail config file for ''
all my users are in 'sip' voicemail context, but adding context to it:
voicemail(@sip) doesn't help.. while if I put a vmbox # to it, it
2004 Sep 27
0
Speex/ILBC buggy with * 1.0 and X-Lite/Pro?
I'm playing with codecs at the moment and have found some notices errors
when x-lite/pro connects to asterisk with Speex or ILBC. Initially I was
getting garbled sound, but after changing magic number for both codecs
to 97 (as per
http://www.voip-info.org/wiki-Asterisk%20phone%20xten%20xlite and
http://bugs.digium.com/bug_view_page.php?bug_id=0000918) I was able to
get normal voice. BUT,
2004 Sep 30
1
sipfriends in MySQL question/request
Greetings,
Is there a way to tie a specific sip username to a IP address when
authenticating against mysql sipfriends table? (USE_MYSQL_FRIENDS=1
USE_SIP_MYSQL_FRIENDS=1 in channels/Makefile)
The reason is that I'm using Wellgate FXSes that have
second/third/fourth FXS ports bugged when I use a password, but work ok
when there is no password. Linking the username to a specific ip could
2004 Oct 01
1
BUG? no output from 'sip show users|inuse|active|subscriptions' when using MySQL auth
I'm authenticating against sipfriends in MySQL, and have just noticed
that none of the below commands return any output:
sip show users
sip show inuse
sip show active
sip show subscriptions
Is this a bug or something wrong on my side?
I'm using the stable 1.0 cvs
Vahan