search for: kfleming

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2012 Jul 18
1
Asterisk 1.8.13 / res_fax / res_fax_digium
We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13 The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf indicate v34 is supported, but when I enable it I get the message "res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored." Is v34 only supported with SpanDSP? Also, the res_fax.conf.sample does not indicate v34 as a valid
2011 Sep 13
1
High delay from Asterisk as PSTN simulator
I'm trying to use Asterisk as a PSTN simulator to run performance tests for echo cancellation algorithms. I'm using the following configuration: SIP <-----> Asterisk 1 <----> Asterisk 2 <----> Echo() Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan application. The problem is the high delay using this configuration: 20 ms only in Asterisk 2.
2012 Feb 16
2
Asterisk && RTCP
Hello list, I need to know about Asterisk's friendly nature with RTCP. I've phones which support RTCP and they connect to the outer world via multiple carriers. In one of my recent packet traces I've observed that the caller initiated a call with rtcp string in SDP while for the same call dialling our from Asterisk to the carrier has no RTCP string in SDP ! Can anyone please tell why
2012 Jan 12
1
Questions on hardware or software-based echo cancellation
Hi, I'm having some questions related to echo cancellation configuration on a Digium board enabled systems (B410P, TE420, TE420B, ....) for cases when a hardware ech canceller is present or not. I read in TEXXX manual that when setting echocancel=yes in chan_dahdi.conf on a VPMOCT64-equiped system, 128ms hardware echo cancellation was enabled. 1. I'm correct thinking that it is then
2011 Jun 08
6
issues.asterisk.org/jira not working
Bad day today. Why this new JIRA system not working. I have created issue and submit and i got blank page.. Please someone help me to create BUG!!!!!!!!!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110608/e99afa31/attachment.htm>
2011 Nov 16
1
Server-to-server BLF
Hi all, Do you have an idea on the best way on how to implement a system with multiple Asterisk servers with BLF working in such a way that a peer on one server can subscribe to another peer on the other server in a seamless manner? Has anyone set-up a system like this before? Thanks! Regards, Ronald -------------- next part -------------- An HTML attachment was scrubbed... URL:
2012 Jun 22
2
SIP over SSL TCP or SRTP?
Hello, Which one of these ensures that SIP packets are sent and received in a secure format so that users using public wifi don't allow MITM type of attacks or others can't read the plaintext SIP packet info. VPN is not an option. Looking for 2nd most secure to VPN. P.S. Are both options part of the configs of Asterisk or need modules to be selected and installed before doing the
2012 Jan 05
1
Where are the fax instructions?
Hello, Trying to set up res_fax_spandsp. Based on https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway I wrote this in my extensions.conf: exten => 306,1,NoOp(Fax transmission) same => n,Set(FAXOPT(gateway)=yes) same => n,Dial(DAHDI/3) ----->FXS port to fax machine same => n,Hangup() Call flow Im trying to pull out is as follows: Zoiper -->
2012 May 10
3
Digium IP Phones
Hello, Im looking to buy a digium phone D70 unit just for testing on lab; to really understand the phone and features. I cant find any website with opinions; any here? Are they really valuable to the price? (D70 quite expensive) Does the SDK for building apps is usable? Can you build powerfull apps? Examples? Many thanks -------------- next part -------------- An HTML attachment was
2012 Mar 02
2
Digium FXS specifications and limits Question
Howdy All, I'm considering Asterisk / Digium as a replacement to my existing phone switch. I need to continue to be able to push analog lines between multiple buildings in a campus environment. The Digium Analog 410 Card manual states it's not recommended to go beyond 1500 feet distance for an FXS card, and no line should leave the building or be bundled. The 2400 Series Manual does
2010 May 21
2
Using unix socket to connect with database
Hello, I am using asterisk realtime with a postgresql database on the same server. In res_pgsql.conf I have specified [general] dbhost=localhost dbport=5432 dbname=asteriskdb dbuser=psql dbsock=/tmp/.s.PGSQL.5432 Since both asterisk and db are on same server, I would like asterisk to connect to db using the local unix socket. However asterisk is not using the local unix socket to connect to
2010 Feb 24
3
Re-INVITE on BYE
Hi gurus, In need of a little help here. I?m trying to do the Asterisk media release by using canreinvite=yes. But I found weird behaviour when comes to BYE. Below are my current setup: Client A is registered to Opensips Client B is registered to Asterisk A ? Opensips ? Asterisk ? B On hangup below are the SIP flow which I?ve notice from the Asterisk server itself: 1. Opensips forward the BYE
2012 Feb 08
1
(last call for comments) Proposed changes to Asterisk release and support cycles
...ensure that everyone has the opportunity to raise any concerns they may have. In the absence of any reason to do otherwise, we'll put this change into effect at the end of this week (i.e. it will become official). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org
2011 Nov 15
4
Multiple SIP endpoint registrations
Hi guys, I want to ask if its possible to make calls using one SIP account, The problem is like this : I have an iPhone app and I want all my users to call the same extension which is virtual extension to my call center, so the iPhone app will be using the same SIP account for all users lets say for example: iPhone users uses 6000 at mydomain to call 9000 at my domain(which is the call center)
2010 Sep 15
2
incoming call FXO
Hi all, Recently I have instaled one Digium TDM410 on my Asterisk. After instaled , I can do outgoing calls but I cant receive calls. I receive the following messages: chan_dahdi.c: Got event 2 (Ring/Answered)...[Sep 14 11:24:44] NOTICE[2654] chan_dahdi.c: Got event 18 (Ring Begin)...[Sep 14 11:24:44] WARNING[2654] pbx.c: Channel 'DAHDI/4-1' sent into invalid extension 's' in
2012 Jul 19
1
Channel is rsrvd and does not turn off
Hi list. I have Asterisk installed on a Debian 1.8 6 64-bit. What happens is the following, some channels are not being hangup properly. They run the hangup in dialplan, but the output of the command "core show channels" shows several channels with status "rsrvd." Checking the server's memory, the "top" command shows multiple processes and stopped using the
2011 Mar 30
5
chan_dahdi unknown dependency problem
So, I've compiled and installed libpri-1.4.11.5, dahdi-linux-complete-2.4.1+2.4.1 and asterisk-1.6.2.17.1, but chan_dahdi is not getting built. If I do a "make menuselect" in asterisk I see it listed with XXX, meaning that dependencies are not met. XXX chan_dahdi Depends on: res_smdi(M), dahdi(E), tonezone(E), pri(E), ss7(E), openr2(E) res_smdi gets built fine, dahdi is
2010 Apr 22
3
How to do analog e&m on asterisk?
Hi, Can anybody with previous experience with it guide me on how to setup asterisk with analog e&m to connect it to an old style e&m system which uses 4 pair cables on RJ 45 jacks. All the analog cards I know of use RJ 11 jacks. And there is no choice of modernization of the customer equipment. Cable pin out are as follows: 1. M lead 2. E lead 3. Tip1 4. Ring 5. Tip 6. Ring1 7. SG 8.
2012 Jan 16
2
How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?
Hello, I can do simple, "yum install asterisk18-*" and it installs Asterisk and Dahdi-tools/Dahdi-Linux on my OpenVZ container. Everything runs well and smooth. However, if I want to compile dahdi-linux on the same openvz then I get the error, *"You do not appear to have the source for the 2.6.32-4-pve kernel installed".* * * 1- Based on above error and Google search I have
2012 Jan 03
3
NAT/IPTABLES workarounds
Hello List, I work in an environment where I have to request IPTABLES changes rather than doing them myself. Is there a way to utilize my SSH (port 22) to get a functional (and with good sound) Asterisk installation with multiple channels up without requesting the 5060(SIP) 5061 (TLS) and UDP/RTP (usually 10001-20000) IPTABLES allowances? Thanks Danny Nicholas