Re: [asterisk-users] SPA400 and asterisk

-- Alberto Sagredo RD area Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Office phone : +34 91 120 5080 Direct phone : +34 91 120 50 39 Peoplecall Network : 700 757 139 Fax number : +34 91 661 9460 ___ --Bandwidth and Colocation provided

Re: [asterisk-users] WiFi SIP phones

/chris.html This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alberto

Re: [asterisk-users] test

ACK 2007/4/12, Razza [EMAIL PROTECTED]: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alberto

Re: [asterisk-users] spc.exe

Sipura Profile Compiler is only for ITSPs and agreements does not permit that Regards Andrew Joakimsen escribió: Does anyone have a copy of spc.exe they could send me? ___

Re: [asterisk-users] WRT54GP2 provisioning

___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alberto Sagredo I+D Area (Asterisk // Cisco

Re: [asterisk-users] Google talk and Asterisk 1.4

Check it! http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk Robert LaPoint escribió: Hello All Does anybody know where I can find information on configuring Asterisk 1.4 to work with Google talk.

Re: [asterisk-users] Google talk and Asterisk 1.4

already tried to follow this document but it did not work under 1.4, so I am just wondering if Google talk is even supported under asterisk 1.4 yet. Thanks Alberto -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Saturday, September 30

Re: [asterisk-users] señalizacion te110p, signaling te110p

Maybe you could try an asterisk forum in spanish in order to get better results using your native language. DiegoF escribió: hola a todos, tengo una duda, ye he resuelto algunas pero otras llegan, bueno como habia dicho quiero conectar una pbx a una te110p, la pbx me ofrece señalizacion r2

Re: [asterisk-users] Linksys SPA400

It has a proxy inside (asterisk), you could register to it as a regular sip proxy, so you could use it. Carlos Chavez escribió: Does anyone know if the Linksys SPA400 is compatible with Asterisk or is it only for the SPA9000 system? It is interesting because it is a 4 FXO ATA at a

Re: [asterisk-users] Linksys SPA400

Not True! You could register against it any spa product, and also asterisk. Cory Andrews escribió: It's only designed for use with the SPA-9000 (LVS-9000) product ecosystem. Cory Andrews -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez

[asterisk-users] Attended Transfer Asterisk 1.2.11

-- Alberto Sagredo I+D Area (Asterisk // Cisco-Linksys) Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39 Fax./Fax.: +34 91 661 9460 ___ --Bandwidth and Colocation

Re: [asterisk-users] Choppy MOH (Cisco gateway)

VAD maybe was caussing this. Regards Zeeshan Zakaria escribió: Actually the problem was somewhere in the Cisco equipment, as the service provider has confirmed. Some option in their device to conserve bandwidth by compressing voice data was causing this choppyness. As they've turned this

Re: [asterisk-users] asterisk logging per day

://lists.digium.com/mailman/listinfo/asterisk-users -- Alberto Sagredo I+D Area (Asterisk // Cisco-Linksys) Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39 Fax./Fax.: +34 91 661 9460

Re: [asterisk-users] Whcih phones are better for mass deployment

I prefer Linksys ones. Spa 9xx series, are great, and provisioning from Sipura/Linksys is much better than PA1628 (Unencrypted). Supports https,tftp and http. With Encryption. Vonage use it. Regards Thomas Kenyon escribió: Michael Graves wrote: Polycom Aastra are both great in this

Re: [asterisk-users] QUINTUM TENOR ASM200 Configuration

If you want to answer directly to him, try Reply to all, and delete [EMAIL PROTECTED] email address. It is not so had to do. FRANCISCO PEREZ-LANDAETA escribió: Hi, this message is for Steve. Sorry for replying to the digest. It wasn't my intention. I would appreciate if you can guide as to

Re: [asterisk-users] Call Processing Slow 11 seconds

Yes you could script a dialplan putting ... and S0 (zero) at the end. An example : (xxS0) It will dial 6 digits directly when you enter the 6th. You could learn how to adapt your Linksys dialplan looking this wiki. http://voip.wikispaces.com/ [EMAIL PROTECTED] escribió: Yes that

Re: [asterisk-users] How to Install H323

I think remember there is a readme on /docs that talks about chan_h323.Check it ! Anyway you could try too at voip.info dot org. Regards Wasif escribió: Hello, Could anyone tell me how to install/configure H323 with Asterisk 1.2.11 . Thanks Wazb

Re: [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls

I have 30 SPA 941 phones with Asterisk 1.2.7.1 and 1.2.9 and it works fine. Are you canreinvite=yes ?. I have not been notice any problem related to transferring calls (blind and attended) Regards Dan Serban escribió: I have a system running Asterisk 1.2.10 (Debian packaging) and about 50

Re: [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls

I use canreinvite=yes in my config files, and it does work, so maybe its a spa 941 misconfiguration. I think if nat=no sometime it has problems if you are behind NAT, but under same network it must not fail. Which firmware are you running on spas? Dan Serban escribió: Alberto Sagredo wrote

Re: [asterisk-users] Linksys SPA-3000 Administration Guide

This Guide is offered as i know only to ITSP and large distributors not to end-users. You could find a User Guide for SPA 3102 at Linksys Website. Regards Marcos Rubino escribió: Anybody have a recent copy of the Admin Guide (not the user guide) for the SPA3000/3102? The only one I was able

Re: [asterisk-users] Choppy MOH (Cisco gateway)

://lists.digium.com/mailman/listinfo/asterisk-users -- Alberto Sagredo I+D Area (Asterisk // Cisco-Linksys) Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39 Fax./Fax.: +34 91 661 9460

Re: [asterisk-users] Encrypting the Conversation

Maybe in Asterisk 1.4 SecureRTP application would do that. Regards Henry J. Cobb escribió: Hi, Is it possible to encrypt the conversation between two parties on SIP,IAX or ZAP channels? Sure, setup a VPN. You can get a Linksys VPN router for less than $100 and run whatever protocol you like

Re: [Asterisk-Users] Asterisk h323

Im using several Asterisk Box with chanh323 from asterisk, and it works fine. Sometime it gets deadlocks , but on 1.2.9.1 and 1.2.7 i have estability. A fail (crash) last month with about 600 calls per day. Regards Alberto Sagredo hakem voip escribió: You can do this by installing a h323

Re: [Asterisk-Users] SPA-941 Disable call waiting or Disable Call waiting via asterisk

and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alberto Sagredo I+D Area (Asterisk // Cisco-Linksys) Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel

[Asterisk-Users] Compiling SVN Trunk

I have the same problem on some modules. For example app_math.so [app_math.so]Jun 9 18:16:45 WARNING[19001]: loader.c:728 __load_resource: missing mod_data for app_math.so Any help?. I have been looking , but nothing reasonable found. Thanks -- Alberto Sagredo

Re: [Asterisk-Users] Compiling SVN Trunk

Umm. Maybe i have left some asterisk 1.2.9.1 modules.. and it has not been replaced. By i made a make install after i compiled it, so it would be replaced?. I will check it. Thanks Joshua Colp escribió: Alberto Sagredo wrote: I have the same problem on some modules. For example

Re: [Asterisk-Users] Compiling SVN Trunk

Thanks again. Sorry. Kevin P. Fleming escribió: - Alberto Sagredo [EMAIL PROTECTED] wrote: By i made a make install after i compiled it, so it would be replaced?. It doesn't get replaced, because the new version of Asterisk doesn't have that module any longer (it's been moved

Re: [Asterisk-Users] Wanted: CISCO 186 ATAs

Why use it? It has been replaced by other Sipura/Linksys Stuff. Do u use SIP or H323...? James Ching escribió: Greetings, I'm looking to purchase 10 Cisco 186 ATAs. Please send me quantities available, payment methods and out the door pricing (shipping + tax + unit costs).

Re: [Asterisk-Users] statistics

Check these ones. http://www.micpc.com/qloganalyzer Queuemetrics http://www.ag-projects.com/CDRTool.html Take a look on voip-info.org for more options Regards issam escribió: Hello How can we do statistics with asterisk thanks

Re: [Asterisk-Users] British English voice files are ready for download

If you need g729 and g723 format, let me know and i could convert it to you. Tim Panton escribió: On 19 May 2006, at 17:05, Mark Phillips wrote: Hi folks, With thanks to Alison Keenan (another Alison!) for the voice, Chris Bagnal for converting from 44k wav to sln and finally Terje Elde for

Re: [Asterisk-Users] change dchannel number

In the TimeSlotting procedure in E1 frames, you have TS0 for Framing and Syncronization, and TS16 for Signaling. Remember 32 channels (30+2). Jose Luis Garcia escribió: Hi: I'm using a te110p E1 card in Spain. There is way to change the dchan channel number? Here in Spain there is a voice

Re: [Asterisk-Users] ATXFER

How many times do u need to repeat it?. You could change this info via web list manager. I think you need to read how to do that before sending 20 emails with same subject. [EMAIL PROTECTED] escribió: Please change the email address of [EMAIL PROTECTED] to [EMAIL PROTECTED] Thanks

[Asterisk-Users] Having Rinback tone generation issues with 1.2.7.1

Today i move our central server to 1.2.7.1 , and im having some issues with SPA Phones and RinbackTone. Without r option, it also happens. Is having anyone this issue? I think it has not been changed anything sustancially to happen this to me. It is happening between extensiones

Re: [Asterisk-Users] H323 to SIP

You could make a H323 to SIP transport. Before to do that, you need to have installed and working both chan protocolos on Asterisk. aFarhad Ibragimov escribió: Hi all I have installed station which support only H323 protocol. I want to install SIP telephone. Is it possible to call SIP

Re: [Asterisk-Users] H323 to SIP

have practice to work with Asterisk but I see that is a great soft. If you have any idea or some config files can you help me -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:34 PM To: Asterisk Users Mailing

Re: [Asterisk-Users] Auto Logout from queue

Via dialplan maybe? exten = xxx,1,Dial(SIP/101_Queue,20,tr) exten =xxx,2,RemoveQueueMember(Comercial_Queue,SIP/101_Queue,1) Kerry Garrison escribió: Yes, that is the functionality I am looking for, just not sure how exactly to pull that off. _ From: [EMAIL PROTECTED]

Re: [Asterisk-Users] How can I get a recording from a CD to my asterisk digital assistant

You will need them in one of asterisk supported formats. wav, slin,gsm, g729, g723... Davi-Ann escribió: I got someone to record the messages we want for our auto-attendant menu on a CD. All I have to do not is to upload the files into the asterisk box, however the format is not recognized

Re: [Asterisk-Users] Quick question

You could try chan_oh323.so and chan_h323.so. I think also ooh323 supports inband DTMFs. Regards Alberto Sagredo Tomislav Parčina escribió: Is there any h323 channel driver that supports DTMF inband signalization? Thank you for your answer! -- Tomislav

Re: [Asterisk-Users] SPA-941/942 Bulk provisioning

-- Alberto Sagredo Departamento Técnico Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39 Fax./Fax.: +34 91 661 9460 ___ --Bandwidth and Colocation provided

[Asterisk-Users] TO have ringing tone instead MOH

I need to avoid MOH on my asterisk box, so i need to have a ringing tone when attendant transfer is made, or a call is on hold.. Is there any way to do that. I did not see a simple way to do that. Regards ___ --Bandwidth and Colocation provided by

Re: [Asterisk-Users] SMS in Spain (it seems Protocol 2)

about deadlines, etc... -- Alberto Sagredo Departamento Técnico Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 Fax./Fax.: +34 91 661 9460 ___ --Bandwidth

Re: [Asterisk-Users] H323 behind a Firewall

If you open h323 port and rtp ports, it should work. Il Neofita escribió: There is a proble to put an H323 Asterisk server behind an iptables firewall? ___ --Bandwidth and

Re: [Asterisk-Users] Ability to put call on hold via manager?

You could park it to parking extensiones. Does it help you? Steve Totaro escribió: Does anyone know if there is built in ability to put call on hold via the manager interface? Thanks, Steve Totaro http://www.asteriskhelpdesk.com ___

Re: [Asterisk-Users] Development news :: T38 passthrough support

Really interesting Olle We are expecting :) Miguel escribió: Olle E Johansson wrote: Asterisk won't be an T.38 endpoint, but will handle T.38 calls properly, regardless if the T.38 was offered in the original call setup, or if the caller suddenly sends a fax in the middle of a call (a

Re: [Asterisk-Users] Can log into the mailbox from Soft-phone , but not from Hardware Phone

I suppose you are using 1.2.4 asterisk version Maybe is not sending dtmf tones as rfc2833 and inband mode is not being detected by your asterisk box. Im a wrong? Could you try to configure dtmf tones on your softphone? John Joseph escribió: Hi I am using asterisk 1.4 on RHEL4 I am

[Asterisk-Users] Authenticated SIP NOtify with 1.2.4?

I have been working with authenticated notifys for auto resync my autoprovisined devices. But it seems to stop the state machine, and when Endpoint answers 401 Unauthorized, the Sip Notify command from cli, does not answer with a Authenticated Notify? Have i misconfigured something?

[Asterisk-Users] Limiting Sip Calls ?

Is there any way not using group count, to limit calls received by every endpoint SIP?.. Outgointlimit and Incominglimit seems to be deprecated on 1.2.x branch. Is there another command to do that? Regards Alberto ___ --Bandwidth and Colocation

Re: [Asterisk-Users] OH323 Peer

An easy way to do that, if you do not neet to register on a gkp, its doing a dial OH323/ipgateway:port Did you try this? Abdul Lateef escribió: Hi all, I have H.323 Gateway, and i want to make a peer to route calls to this GW. But i don't know is oh323.conf supporting to add peer type entry

Re: [Asterisk-Users] R2 implementation problem

I hope this link will help you. http://zarzamora.com.mx/asterisk/17 Regards Manuel Marin Garcia escribió: I have a TE110P connected to a Telmex E1 circuit with R2 signaling. Asterisk version= 1.0.10 Zaptel= 1.0.1 Spandsp=0.0.3pre6 Unicall= 0.0.3pre8 *zaptel.conf span=1,1,0,cas,hdb3

[Asterisk-Users] About Meetme and CDRcustom

Hi all. Does anyone know how to cdr the meetme conference number that the person who enter called?. I did not find the variable and, last_data, seems not to give me the correct info. Regards -- Alberto Sagredo Departamento Técnico Peoplecall Email : [EMAIL PROTECTED] Blog: http

[Asterisk-Users] About Extensions

Im trying to detect before entering in Meetme , which dtmf has been entered. I did a Background(file) and go to a context where i define a exten = _X.,1,Meetme() I have detected that with (1.2.1) when 1 is entered and conference 1 must be created, extensions say it is not possible and

Re: [Asterisk-Users] Asterisk always uses 127.0.0.1 address

Maybe you have not configured correcly your sip.conf externip=your_external_ip try this RumaTech escribió: Hi, all Can someone tell me where to tell asterisk no to use 127.0.0.1 IP (localhost)? When I am registering with VoIP providers, they get my info as [EMAIL PROTECTED] (This is SIP

Re: [Asterisk-Users] Easy to Access Telephone Directory AGI

Really interesting. Thanks Hannes!! Hannes Vogel wrote: I've written myself a easy to use telephone directory which I use at home and thought it may be of interrest to others. The purpose of this agi script is to provide an online telephone directory that can be easily accessed using the

Re: [Asterisk-Users] How to Unload app_rxfax.so

Yes, you could do that making some changes on modules.conf noload = app_rxfax.so Regards Alberto Nitesh Divecha wrote: Hello All, Dunno what happen but Asterisk is refusing to start... Went over the log and found out that app_rxfax.so is failing to load. Jan 7 11:57:28 VERBOSE[4320]

Re: [Asterisk-Users] Asterisk Call Forwarding

://lists.digium.com/mailman/listinfo/asterisk-users -- Alberto Sagredo Departamento Tcnico Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voip-novatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 755 048 Fax./Fax.: +34 91 661 9460

[Asterisk-Users] Fast AGi Variables

Has anyone an example to pass variables to a fagi script? I have succesfull made some examples with traditional AGIs, but i could not find a way to do with FastAGI. Regards -- Alberto Sagredo Departamento Técnico Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voip-novatos.es Tel

[Asterisk-Users] Too high volume on Music on Hold

Hi all. I have an asterisk box on gentoo , and when i try to play MOH, it get too much volume. At a point that it could damage my ear system :) If i normalize the music, decreasing the volume, it normalizes again and play at a volume that i could not use. What could it be wrong?. In other

Re: [Asterisk-Users] asterisk + H323 + 723

Hi, I had the same troubles too. It does not recognise correctly g723 with oh323. With h323 i have dtmf rfc2833 issues but g723 and 729 are transported correctly via H323 capabilities. So, let make a try with h323 included in asterisk branch, not the oh323 Kanishka Somaratne wrote: Hi I