similar to: SIP/ Grandstream Issues

-This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow

I just checkout the cvs code for asterisk...... when I use my grandstream phone (that worked on the old code that was about 2 months old) I do not hear anything at all... I get this error: Sep 27 23:20:27 WARNING[1142127920]: File chan_sip.c, Line 444 (retrans_pkt): Maximum retries exceeded on call 0765c89e-9d67-3c0a-b9b9-2e7f3cd1d9ef@192.168.50.248 for seqno 58430 (Response) here is my

Hi all I am having some issues with a gs 100 phone. It is on the same network as my * server. There is no firewall. In extentions.conf exten => 5,1,Answer exten => 5,2,MusicOnHold(default) When I dial 5 from the sip phone -- Executing Answer("SIP/mlh-2e75", "") in new stack -- Executing MusicOnHold("SIP/mlh-2e75", "default") in new stack

Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The

Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: --

Hi everyone, I?m a graduate student in a Brazilian Intitute of Technology, and I?m doing some academic research regarding secure voice transmission over phone lines. One of our reserach goals is to implement a hardware vocoder, with low bit rates, and a preferably free algorithm, to be used in this secure voice system. Actually, there is a functional system using a proprietary AMBE

Here is a quick tally of the various things people asked for.. I'm going to go thru the list and weight the results based on my scale of 1-10. This is just a count of each "item", otherwords how many times that item came up. Some things I considered as bugs and lumped them as bug-fixes For the various requests for codec's I broke out which ones people where asking for.

Hi, My name is Jean Quirion and I am a DSP engineer. Currently I am working on a project where it is desired to implement a VoIP solution over a GSM GPRS link. I would like to use Speex as the vocoder for this application. This application would require the Speex encoder/decoder and possibly the pre-processor to run on a low power fixed-point DSP such as a TI C55x. Thus, I am interested in

Every time I send an e-mail to the * list, I receive this "AntiSpam UOL" E-mail. is anybody else experiencing the same? How can I get rid of it? Uriel -----Original Message----- From: AntiSpam UOL [mailto:andersoncbr.sspam@uol.com.br] Sent: Wednesday, September 24, 2003 11:51 PM To: uriel@adelphia.net Subject: RE:RE: [Asterisk-Users] SIP / GrandStream Configuration Ol?,

Hello, I'm interested in a low rate vocoder to run on SA-1110 or XScale processor (for PDAs voice application). Is there a suitable speex vocoder for that? Thanks in advance, Eyal. ________________________________________________ Get your own "800" number Voicemail, fax, email, and a lot more http://www.ureach.com/reg/tag --- >8 ---- List archives:

Jim, Thank you very much for your suggestions. I managed to get the C55x code working on the simulator. I would like to port Speex both on a C5502 EVM and a C5509A EVM. As such, if you can provide me with the details of your port on the C5509A, it would be greatly appreciated. Furthermore, I am looking for some technical documentation on the CELP algorithms. I would like to better

for the bits init I am using speex_bits_set_bit_buffer and I don't use the write to or read from because the data is already in the buffer I am reading from and I am writing to the final buffer so I don't need to move arrays around. what part is the vocoder part of the decode? Thanks for your help! -Mike >>> Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> 08/06/07

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Hi, I found this message concerning VAD and was wondering whether VAD has been ported to fixed-point in the latest version? Thanks, SingHui ---------- Forwarded message ---------- From: Jean-Marc Valin <Jean-Marc.Valin@usherbrooke.ca> Date: Jul 22, 2005 1:02 AM Subject: Re: [Speex-dev] Fixed-point To: gue baja <gue_baja@yahoo.com> Cc: speex-dev@xiph.org Hi Baja, Here's a quick

Just got GS 101 phone and plugged into the network. Got ip setup however, the following problems arise: 1. when dialing an extension, I cannot further send any key tone to Asterisk. 2. there is no sound coming from the other end. I have a sip.conf setup for GS: [General] disallow=all allow=ulaw allow=alaw [gs] canreinvite=no dtmfmode=info In the GS101 setting rtp port = 5004 sip port = 5060

I was wondering if the developers were using anything to "objectively" test the quality of the speex vocoder. For instance PSQM or one of the many derivatives. Mean Opinion Scoring seems an expensive route. Is there some open source software to use for this? <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To

Hi, Anyone know how to set up tftp server for grandstream. I gues it should be somethink like <tftpserver-dir> <mac-address> firmware.bin config.txt Is this correct ? And how should the config-file look like. ? I had search sipphone.com but did'nt find anything. /HHA _________________________________________________________________ Rethink your

Full_Name: Yves Rosseel Version: 2.2.1 OS: Submission from: (NULL) (157.193.116.152) The code below sketches a possible implementation of a function 'mlh.mlm' which I think would be a good complement to the 'anova.mlm' function in the stats package. It tests a single linear hypothesis of the form H_0: LBT'= 0 where B is the matrix of regression coefficients; L is a matrix

-----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: Thursday, June 14, 2007 11:17 PM To: Michael Shatz Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Blackfin inline assembler and VisualDSP++ toolchain Michael Shatz a ?crit : >>> Actually, you're the first I know using the VisualDSP++ toolchain >>> :-) >> >> I guess