Re: [asterisk-users] R2 with Alestra in Mexico...

Yes, i have the same problem with att a few months ago, the problem is the acount (abonado) code, att need 2 and the code of unicall send 1, maybe the problem is the same for you, please post the debug unicall code. In this code, you can see the dial number, but if you see, the last digit is 1

[asterisk-users] Please help with a telular mod. SX5e

Hi I have 5 telular mod SX5e https://www.telular.com/v2/html/products/product_display.asp?productID=94 is a great stuff, but i have a extrange problem with asterisk. Sometimes the sound is ugly or choppy. The telular alone work fine all the time For example if i made 5 calls from asterisk to gsm

Re: [asterisk-users] E1 for Voice and Data with MFC/R2

Hi a few months ago i use a digium card in a E1 with 10 voice channels an 10 in data, i use a gentoo distro with kernel 2.6, if you can tellme more info for example the channels, the type of card, kernel, etc maybe i can help you On 8/7/06, Moises Silva [EMAIL PROTECTED] wrote: I just found this

[Asterisk-Users] Please Help i have a error with unicall and ATT

I have a wired problem, i can recive call but i can't make any call. ATT say that is my problema because the call is operator mode (???) The log is MFC/R2 Chan 1: Call control(1) MFC/R2 Chan 1: Make call MFC/R2 Chan 1: Making a new call with CRN 32772 MFC/R2 Chan 1: 0001 - [1/ 1/Idle /Idle ]

[Asterisk-Users] Unicall Question

Hi I have a litle question, what is then version stable, in the web server i can see unicall version x.2.x and version x.3.x, and the time is same unicall-0.0.2e/ 11-Nov-2005 18:33 unicall-0.0.3pre8/ 11-Nov-2005 18:37Where i can find the change log or the diference from this

Re: [Asterisk-Users] ATA186 V2.15.ms upgrade

Hi you can download the file here http://kvin.lv/pub/Cisco/ata_03_01_00_sip_040211_1.zip bye On 2/15/06, Weiming Jiang [EMAIL PROTECTED] wrote: Hi, Iwanthave upgrade my ATA186v2.15ms toH.323/SIP ButI dont have a cisco acount yet can somebodyhelp me with the ata18x-v2-16-030401a-1.zipfile ?THX

Re: [Asterisk-Users] Fax with Asterisk and Sipura 2100

Change the RTP Packet Size: 0.010 to RTP Packet Size: 0.020 Asterisk only work with 2 frames. I can't send any fax with other values. On 1/5/06, Joash Herbrink [EMAIL PROTECTED] wrote: You could use a cisco ata 186.There aren't very cheap, but I have made them work on several of

Re: [Asterisk-Users] weird problem with sipura spa2000 and soundcardpa setup

Hi, check in the sipura in advanced mode the parameter of RTP Packet Size change it to 0.020 maybe with this you can fix the problem. On 12/26/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Yup all ata's can talk to each other just fine.I can call one for another,they all can make out going

Re: [Asterisk-Users] [EMAIL PROTECTED] Fax to Email problems

Hi First check i u have the program tiff2pdf,so change in the extension.conf in the line where say tiff2ps change for exten = in_fax,3,system(tiff2pdf -zf ${FAXFILE} -o ${FAXFILE}.pdf) this work for me. On 12/22/05, Jay Nemrow [EMAIL PROTECTED] wrote: I am using [EMAIL PROTECTED] (the newest

Re: [Asterisk-Users] Unicall E1 Error in Mexico

5000 #define DEFAULT_T3 15000 vuelves a compilar y a instalar y listo... On 12/20/05, Jorge Cisneros [EMAIL PROTECTED] wrote: Hi I have a weired problem. when i make a call with some numer the unicall show me a error. For example when i dial 30003300 in mexico city then log show MFC/R2

[Asterisk-Users] Unicall E1 Error in Mexico

Hi I have a weired problem. when i make a call with some numer the unicall show me a error. For example when i dial 30003300 in mexico city then log show MFC/R2 UniCall/3 R2 prot. err. [2/ 40/Group I /DNIS ] cause 32769 - T1 timed outDec 21 00:22:46 WARNING[17649]: MFC/R2 UniCall/3 8 off -

[Asterisk-Users] Unicall Problem with fax

Hi I can't recive or send any fax using a unical truck, i putfaxdetect=incoming in the unicall.conf but asterisk ignore it The log show Dec 21 06:21:10 VERBOSE[22400]: [chan_unicall.so] = (Unified call processing (UniCall))Dec 21 06:21:10 DEBUG[22400]: Parsing /etc/asterisk/unicall.confDec 21

Re: [Asterisk-Users] MFC/R2

Hi I have a few question about unicall What version of unicall is stable 0.0.2e or 0.0.3pre8 i am using 0.0.3pre8 but i have a litle problems 1.- I can't send or recived any fax. i put the faxdetect in the unicall.conf but your code ignore this. 2.- When the user dialsome numbers the ocurre

[Asterisk-Users] Ticking sound in wildcard tdm400p, Please Help

Hi My wildcards TDM 400P's with 8 FXO port(connexts to pstn ) has tickingnoise.I have followed http://www.voip-info.org/wiki-Asterisk+Hardware . and make sure wctdm is not shareing interrupt with any other devices.The sever hard disk is a scsi, so i can't run /sbin/hdparm -u1 /dev/hda1 to

[Asterisk-Users] Question about 3Com(r) 3101 Basic Phone

Hi, i have one question, the 3Com 3101 Basic Phone work with asterisk, if so i any a especial firmware o another thing. And wath other 3com ip phone product work with asterisk. I think is a good idea to create a list with the all voip device and the status with asterisk.

[Asterisk-Users] Question about 3Com(r) 3101 Basic Phone

Hi, i have one question, the 3Com 3101 Basic Phone work with asterisk, if so i any a especial firmware o another thing. And wath other 3com ip phone product work with asterisk. I think is a good idea to create a list with the all voip device and the status with asterisk. Thanks.

[Asterisk-Users] Wildcard TE110P in Mexico

Hi I have one question, somebody can tell me if the card TE110P work in mexico, and maybe can tell me the config. Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] DTMF outbound problem with ata 186

Hi This bug is really crazy, please help me In the follow scenary ATA-186 - SIP - Asterisk - SIP - ATA 186 : No DTMF gets through * in outbound mode, Sip conf [204] type=friend username=204 secret=somesecretpassword host=dynamic canreinvite=no ; The follow line don't work dtmfmode=rfc2833 nat=1

RE: [Asterisk-Users] ActXPhone Active X control link is dead - has anyone cached files ?

The link work fine, the link is http://www.ict.tuwien.ac.at/staff/darilion/ActXPhone/ -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Robert Rozman Enviado el: Sábado, 21 de Agosto de 2004 04:11 p.m. Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users]

[Asterisk-Users] RFC3389 support incomplete

Hi When i make a call using IAX2, the log of the remote asterisk say Nov 17 20:20:12 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:22 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport

[Asterisk-Users] IAX2 Problem

When i make a call, i have the follow error and loss the sound Nov 17 20:20:12 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:22 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete.

[Asterisk-Users] SIP QUESTION

Hi Is posible to make a call from site A to Site C, and my question is, the rtp data is from A to C or is from A to B to C Site ASite BSite C ata186FW-Asterisk-FW---ata186 Thanks

[Asterisk-Users] Test

Test ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] h323 Ringing sound

Hi When i make a call using oh323 is posible to make the ringing sound thanks

[Asterisk-Users] cisco 186 helpp!ª!!!!

toy buy my first cisco 186 but when i read this page http://www.djernes.org/~shawn/ata186.htm i cant find in my dev page some parameters just like " UseSIP " what i need to do to show this parameters Thanks

[Asterisk-Users] oh323 and GSM

When i make a call from asterisk oh323 channel to ohphone i have the follow warning WARNING[23573]: File codec_gsm.c, Line 165 (gsmtolin_framein): Invalid GSM data and the sound quality recived is bad but the quality of the sound send from asterisk is good any sugestion

[Asterisk-Users] Ringing tones oh323

When i make a call using oh323 channels, how i can send a ringing sounds to indicate to the users that the call is in progress thanks

[Asterisk-Users] Wildcard X100P question

Hi Why if the Wildcard X100P is really a modem with chipset motorola 62802-52, why you can use another modem as the same form. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users