similar to: SIP REGISTER

I write the email again, the third time!!, cause the other two ones, I have had problems while sending them. I hope this time it works. Here is the email again: Hi (and sorry) everybody I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! 1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have

Hi eveybody again! I don't want to be annoying, but if nobody can help me with this, I'll have to desist of working with SIP.I have some questions about SIP, as I wrote in another mail. I have a SIP Gateway and I have two phones (an analog one and a DECT one) conected to it.Also, I have two Dlink dg102s with four phones conected to them. The main problems are two. Calls between the

Please I need help, I don't know why,almost every time I dial on my dect phones, the dialtone doesn't go off and * doesn't recognise anything!!!! I'm using two dlink voip gateways, MGCP: DL102s. Any ideas? thanks in advance michelle matis ----- Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com Ya.com ADSL Home 24 h, Módem + Alta ¡Gratis!

I've tried to configure samba to lock files bewteen windows and linux but i couldn't i've read a lot of messages here, but trere is no one that have something about the file smb.conf. i have this in my global secction but i doesn't work [global] workgroup =3D GMC create mask =3D 0777 os level =3D 16 directory mask =3D 0777 hosts allow =3D

Can anyone recommend some phone sets that are adsi compliant and work well with asterisk?

<P>I write the email again, cause the first one I have had problems while sending it. Here is the email again:</P> <P>Hi everybody,</P> <P>I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! </P> <P>1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have two

<P>Hi everybody</P> <P>I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! </P> <P>1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have two dlink dg102s (MGCP) and I'd like to can call them too. The problem is that when I use g723 I can call MGCP but I can't

I'm using * as a Call Agent for two DL102S but I have some problems, like the tones not being sending from the phone to the *. I have not changed the configuration of the DL, except the IP and the Notify Entity (*). Must I change another thing in * or in the device? Thanks very much michelle <br><br> <b>Tu cuenta de correo gratuita <a

HI, can somebody tell me how and where must I put the SIP register line? I think is in [general] section of the sip.conf and that I have to put: register => user:password@host:port/localextension but, user and password of the SIP gateway? Because I'm trying this and doesn't work... thanks a lot in advanced michelle ----- Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam

Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine;

Hi, I've got an Asterisk box with grandstream and xlite clients on it. No here's the thing: - I grey out all the codecs on the Xlite except for GSM - I call the Grandstream from the Xlite, the Xlite uses the GSM codec and the Grandstream uses ulaw, with Asterisk doing the conversion, everything fine - I call the Xlite from the Grandstrea, the Xlite ends up using the ulaw codec as

Asterisk version 1.2.14 I use snom190 and xliteV3 as sip phones. asterisk send the rtp stream never to the xlite softphone. Any hits for me? *CLI> rtp debug RTP Debugging Enabled -- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack -- Called snom -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 answered

Hi, I'm wondering if anyone has experienced an issue with the XLite softphone and asterisk accepting dtmf? I can listen to my voicemail perfectly from my hardphone. However when I dial the voicemail number from my XLite softphone and enter the password at the voicemail prompt, an error appears vm-incorrect and I get an "Unable to read password" message on the asterisk console. Has

28th February I am working with Asterisk 1.2.15. I have configured sip.conf for two soft phones (I am using Xlite).I have installed the Bluez stack and so far, i manage to make a phone call from a soft phone to a GSM network. However, i have an audio problem. The soft phone can be heart by the GSM costumer but the voice in Xlite is not transmitted to the GSM. In asterisk all i got is the

Dear All, I have downloaded the xlite version 2.0 for windows and I made the following conf in the xlite itself as the document suggested in order to make it work with Asterisk but still it doesn't work as a matter of fact when I tried to make a tcp dump I can see no packets going between the windows client and the Asterisk server at all, here is the my conf on the xlite itself: in the

My first Asterisk install: Debian sarge with the 2.6 kernel, and two X-Lite soft-phones. I followed the online how-to documents and was calling between the two soft-phones and calling the demo system with no problems and had full audio. I then went on to configure the TDM400P's two FXS modules. I got into that a ways and was having some success, but no dial-tone when I was off the

Hello I thought I had things set OK to have Asterisk play FR files for prompts and MOH, but for some reason, it still can't find them: ============ ll /var/lib/asterisk/sounds/ drwxr-xr-x 2 asterisk asterisk 4096 2011-01-21 16:18 custom/ drwxr-xr-x 10 root root 61440 2011-03-17 14:21 fr/ Note: fr/ contains core + extra + moh as downloaded from here:

I have a bunch of road warriors who I've set up with Xlite clients. Unfortunately the sound quality has been intermittent at best. Sometimes it's great other times completely unusable. When it's bad one usually hears harsh static when the other party speaks or their voice gets "clipped" to static if they speak too loudly. Many of these users have migrated to Skype ? much

Beginner to Asterisk, but not beginner to VoIP FreePBX front end running on a dell 1550 and XLite running on a different Woindows XP box Both boxes connected via switch on same subnet. No NAT involved On FreePBX I created a new extension 1001 with a SIP password of 1001 On Xlite, username is 1001, password is 1001, authorization user name is 1001, and domain is IP of Free PBX XLite tries to

Hi guys i signed up to iconnect a few hours ago to try do some cool stuff. but im having a few problems. Im running asterisk and xp/xlite softphone.. both xp box and asterisk box are on public ips. The problem is that when i ring anyone in the world it'll ring they'll pickup and i can hear them 100% perfectly/clearly.. but they cant hear me.. occasionaly they can hear something like a