search for: natel

Alex thank you for your response. In this case we are USING INBAND, though I have tried both. Nothing works. Yes ser is configured with mediaproxy. Thank you, -JK JK, In-band or RFC2833 DTMF signaling? Also, unless you have SER configured with a media proxy, the actual "call" is not running through SER. It's a signaling proxy only. -- Alex Balashov Evariste Systems Web :

Howdy, folks. I'm having a problem finding a way to auto-record calls (both incoming and outgoing). I know how to make it so either party can initiate recording, but I want it done as soon as both ends are connected (or prior to that if that's what it takes). It's probably right in front of me and I'm just missing it. Any help would be much appreciated. Thanks, Jay

Using a Polycom 550 and 650 phones on my Asterisk server for testing. I can't figure out why the volume is so low. How can I adjust the volume control on Asterisk? It's at max on the handset phones. Thanks! Hin

...itional information that would aid in answering these questions. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david at safedatausa.com ------------------------------ Message: 3 Date: Fri, 20 Jul 2007 11:46:58 -0500 From: Doug <Doug at NaTel.net> Subject: Re: [asterisk-users] Any plans for proper faxing support To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message...

Hey guys, In your experience what is the best way to go for a production asterisk box in your offices? With desktop prices so cheap you might think that you should just buy them off the shelf, but is that really a reliable machine? Anything you can tell me that would assist me in deciding the best way to obtain and maintain these boxes would be very helpful. I have even looked into building

Hi, Has someone figured out how to change the main autoattendant message easily? Right now, you call *77 and record the message. Then you have to get into the Unix/Linux command line to get that message over to where it will be used. Is there a simpler way? Thanks for your help.

I have an odd issue, where a polycom 601 stops ringing, or more properly, maybe, stops *being* rung, when a call comes in. Other phones/extensions, continue to work fine, they being run at the same time. My dial plan works fine (?) seems it will ring properly, right after a reboot. It works fine for outgoing calls at all times. Hints? joe a.

From: Doug <Doug at NaTel.net> >"Net Neutrality" is great in principle. But ISP's need to >somehow control those few percentage of users who suck down >a huge majority of the bandwidth. It's dollars and cents. There is a rational solution for the traffic management issue. It just needs to...

...mation from listproc@samba.org. And the search engine seems to have been discontinued. So please reply to me as well as to the list, if you like. -- Dr. Andreas Mueller Beratung und Entwicklung Bubental 53, 8852 Altendorf afm@othello.ch Tel: 055/4621483 Fax: 0554621485 Natel: 079/3556714

Does anyone know the maximum number of digits for an international phone number? Doing some searching, it looks like 16 numbers including the "011" is the maximum number, because 17 is just not found: OK: 1234567890123456 http://www.google.com/search?q=011XXXXXXXXXXXXX Not OK: 12345678901234567

Has anyone figured this out yet? Lots of places say to add the following to sip.conf of an Asterisk 1.2 system (current production machine/Asterisk as root): tos=0xB8 (Hex B8 = Decimal 184 = Binary 10111000) or if you are running Asterisk v1.4 or newer: tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. tos_video=af41 ;

Hi, I am running samba 2.2.7 as a PDC with a couple of Win2k workstations participating in the domain. Once in a while of the workstations, I can't access shares on the other, getting a message "no logon servers available". Once I restart smbd and nmbd via /etc/init.d/smb stop start, the problem goes away immediately. I tried to search in samba logs, but there were no error

I have a client who is using a Shoretel PBX. This PBX apparently does not send DTMF information OOB, but instead sends this inband via the B-channel. This is traversing an Asterisk box via a PRI. The user calls the IVR (1-800-CALL-DHL), receives audio, but is not able to present DTMF to engage the IVR. With some light research it appears that the DSP is not activating until the call is

Hello, Please forgive me if I'm repeating this post. I have searched and looked for similar problem with a solution but have not see a similar one. My outgoing SIP and other channels work fine but the incoming/inbound SIP call goes straight to Broadvoice voicemail. I see that Broadvoice is registered when I look at the SIP registry. I have turned on SIP Debug and it is below. Anyone know

...you very much -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081129/9853fc5f/attachment-0001.htm ------------------------------ Message: 11 Date: Fri, 28 Nov 2008 23:43:22 -0600 From: Doug <Doug at NaTel.net> Subject: [asterisk-users] libspandsp.so.0: cannot open shared object file: No such file or directory To: asterisk-users at lists.digium.com Message-ID: <984185085-267834602 at mail1.natel.net> Content-Type: text/plain; charset="us-ascii"; format=flowed libspandsp.so...

Our small company is replacing Cisco CallManager with Asterisk (because we are tired of sending them money) and I am documenting the process as I go on my blog. I am trying to make the notes as easy as possible in hopes that I can ease someone else's pain. Here is the link: http://qvlweb.blogspot.com/2009/03/asterisk-pbx-system-install-01-what-i.html Please feel free to comment on the

Hi, Does someone know what the problem is when the call goes through but one or both parties can't hear the other? What are the common causes? Solutions?

Hi, In modifying the template for voicemail emails, I can only get ${VM_CIDNUM} to appear in an email. Does anyone know how to get the others to appear? Thanks for your help!

Hi, Last night I could dial *8 and pickup a call that was ringing to another phone. This morning, I searched on the Web for a solution to mixed up digits when dialing on a Polycom Soundpoint 501. I found that if you go to the SIP page on the phone's >Web interface and change the "Digitmap Impossible Match" setting from "0" to "2" that fixes the mixed up/eaten

Yesterday the phones were working fine. Now they won't register. The Trunks section is greyed out. Does anyone have an idea about what is wrong?