similar to: 1.6, t.38 and zoiper - t38_udptl or t38pt_udptl ?

Hi, I would like to improve my understanding of T.38. 1. What T38FAX_VERSION_0 or T38FAX_VERSION_1 in chan_sip.c means ? voip-info.org implies one has to change values in chan_sip.c to make it work. Shall I set T38FAX_VERSION_1 or leave T38FAX_VERSION_0 in global_t38_capability ? Source code says "This is default: NO MMR and JBIG trancoding, NO fill bit removal, transferredTCF TCF, UDP FEC,

Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can't get it to work. -David -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Julian J. M. Sent: Friday, March 31, 2006 1:44 AM To: Chris Earle; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: BRI

Hi there, I support a large number of enterprise users who contractually must connect to our support center via a 4G VOIP connection. I simply want to be able to auto detect all poor quality calls in realtme (as they are being made), play a message and drop the call - without user intervention. All decent call quality calls will be allowed through - to be handled by support staff. Its a

We're looking at implementing Asterisk in our department in the near future, we're looking at anywhere from 15-25 extensions. The machine we were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 1GB of ram. I've heard bad things about running Asterisk on SMP machines? Would we be running into any performance issues with this machine? Tim Jackson Network Engineer

Hi, I was wondering if anyone else has a similar setup and can suggest settings for the jitterbuffer: I have a client with an ADSL connection at site A & B with site A being dedicated to voice and having no Asterisk server, site B combining voice and data with traffic shaping and housing an Asterisk server. There seems to be packet loss / jitter on this connection and I wanted to know

Hello Is it correct that IAX2 uses more CPU, than SIP? Also, can it be true that IAX2 is much more sensitive against high CPU loads? Also, does Asterisk support and use multiprocessor architectures, such as Xeon? ? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/362 - Release Date: 12-06-2006

I am having some issues when trying to connect with perl to the asterisk manager and doing an "IAX2 show channels". If i do that on a server that is heavily loaded, i sometimes get some events instead of the channels i asked for. Any suggestions how i could fix that ? zoa.

Hello, How many licenses to buy?? : From what we understood from digium website, we must buy as many licenses as the number of maximum simultaneous calls using G729 Codec we wish to make. For example, If we want to be able to make a maximum of 10 simultaneous calls using G729 Codec, we must buy 10 licenses. Is it right? Thanks you

Hello all, I'm setting up a couple of test boxes and I'm running into a problem. What I need help with is determining whether I'm going something wrong or if I need to post a bug report. I have two asterisk 13.0-beta 3 machines set up with extensions connected to each as such: 3700 ----> AST-A <------> AST-B <---- 3800 & 3801 When I place a call from 3800 to

What I have is: * Asterisk 1.8.10.1~dfsg-1ubuntu1, * SPA112 ATA with analog fax in 1-st FXS port connected, * SIP trunk with provider supporting T.38. My network looks like this: * spa112 (192.168.33.200/24) and Asterisk (192.168.5.253/24) in neighbouring LANs, * Asterisk connects to the provider (80.75.130.136) via router (82.200.7.184). Router has full DNAT to Asterisk server. What happens?

Thanks. I think our problem ca be similar. Have you tried to call from analog phone #1 to another analog phone #2? It works. But when you try to call vice versa from #2 to #1 it does not work. When you restart asterisk it works again - but only one direction. -David ________________________________ From: asterisk-users-bounces@lists.digium.com

Hi, I'd like to test Asterisk performance under more concurrent sip calls. I use Sipp, but do get "Unexpected message for Call-ID ...", so I wonder if anyone is using sipp succesfully with Asterisk and is willing to share more info about his solution ... Any other convenient way to load test Asterisk ? Is sipp the right tool ? Thanks in advance, regards, Rob. sipp: The

Hi, I have asterisk 1.6.2.6 on a Debian Lenny system. When I try to send a fax in T.38 mode I receive this error ERROR[15035]: res_fax.c:795 set_fax_t38_caps: channel 'SIP/eutelia-sirio-out-00000000' is in an unsupported T.38 negotiation state, cannot continue. In my sip.config general section I have added this lines t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no If I comment this lines,

Hi List Asterisk 16.28.0 in use. PJSIP in use Two endpoints Both using IPv6 One Endpoint on UDP, the other via TLS. Both with: t38_udptl=yes ;fax_detect=yes ;fax_detect_timeout=30 rtp_ipv6=yes Both sides are T.38 capable and detect fax tone so no need for fax detection on asterisk. Voice calls between the two work fine. But on a Fax call, I see this situation: A <=> Asterisk

Hi all the people in the list! I'm new on this list, this is my first post. I configured asterisk 1.6 with freepbx 2.7 and dahdi to send faxes with t38modem conected to hylafax as a sip extension of asterisk. Everything is supposed to be configured fine, the faxes start sending, but at the middle of the transaction, it fails. The T.38 termination provider told me that they were receiving

All; I'm having a problem sending an outbound fax using Asterisk-1.8.15-cert3 and the spandsp fax module using a SIP trunk. I'm seeing hundreds of these: ERROR[14423]: udptl.c:294 encode_open_type: UDPTL (SIP/runcentral_outbound-00000074): Buffer overflow detected (59 + 134 > 175) Has anyone ever seen this before? I have the following configuration. udptl.conf:

I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating

Hi there. I have the similar problem ("Digium fax - sending fax call file vs manager originate") sending faxes with Asterisk 1.6.2.6 and Digium res_fax. Receiving is OK. I use Local channel in Call file and context [fax-out] in dialplan. My setup: Asterix<-SIP (T.38)-> Cisco(MERA MSIP v.1.0.2)<-> LocalTelco<->fax machine Debian GNU/Linux 5.0 ; Linux 2.6.26-2-686

I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and converted form SIP to PJSIP using the python script as a start and then mofiying from there.  I ran into an issue when testing that incoming calls from MagicJack would go silent after about 10 seconds.  This happened while in the automated attendant area.  This problem did not occur with Asterisk 13 LTS.  I reverted PJSIP

Hi Vardan I did same as you told and deleted the SIP information in Astdb and restarted asterisk. but the result was same. as you said there might be mistake in sip.conf so i am pasting both servers configuration here.. 1- nasir.server.com [abc] username=abc type=friend secret=mysecret nat=yes mailbox=12234568 incominglimit=2 outgoinglimit=2 host=dynamic dtmfmode=rfc2833 context=payasyougo