similar to: How to use one single IP as origination

Displaying 20 results from an estimated 1200 matches similar to: "How to use one single IP as origination"

2010 Oct 26
5
Mobile Phones and Asterisk
Hi, Is the dev_state can also be used to track a mobile phone's status via SIP? I tried it on several phones(nokia, samsung) but it returns NOANSWER but i can hear a beep beep beep sound indicating that it is currently busy. regards, RYAN ICASIANO
2011 Feb 15
6
Fax Woes
Hi all, I'm trying to get outgoing fax working from an SPA2102 to a PSTN fax machine via a T.38 enabled trunk.? I've got t38pt_udptl = yes faxdetect=no in my sip.conf file.? The ATA has all of the T.38 options turned on, echo cancellation is off, as well as silence suppression off.? The only configured codec is u711.? When the user tries to send a fax, it gets to the point where it
2010 May 22
4
US "Truth in caller id act"... and it's impact on services
For the 3rd consecutive term, the US Senate has introduced the "Truth in caller ID Act of 2009". It was passed by the Senate (finally) in January, and has moved to the House for a vote. A lot of states have ambiguous or overly restrictive language on how caller ID may be manipulated. For instance, if you have a PBX, and a call comes in from the PSTN, which you then loop back out
2010 Sep 08
2
Max TDM calls per asterisk box
Hi Everyone, Can you tell me how many concurrent TDM (Dahdi) calls that a single asterisk box can handle. Configuration is as follow : Quad core Xeon 3 GHZ, 4Gb RAM, asterisk 1.6.2.9 Also do you know a good tool to stress out asterisk? Kind regards -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* --------------
2011 Apr 20
2
Call files or AMI originate for mass outbound call
Hello Guys, In the case of a multiserver environment for outbound automatic calls, can you share you experience and preference between call files and ami originate ? thanks -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -------------- next part -------------- An HTML attachment was scrubbed... URL:
2010 May 19
2
Asterisk Cluster
Hello Everyone, I must deploy an asterisk system that can support at least 500 pstn outbound calls. It's a challenge as it's the first time i'm gonna build such a large system. I want to have your advice on hardware, software and so on . What i have in my plan is a cluster of servers with quad PRI cards. I will appreciate your advice. Thank you all . --
2010 Jul 16
1
g729 codec loading
Hello Everyone, I've successfully registered my g729a licenses. When i try to load the module from asterisk Cli i got the following error *Error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after reloc: Permission denied* * loader.c:795 load_resource: Module 'codec_g729a.so' could not be
2010 Nov 08
4
Integrating With Asterisk
Hi, I'm trying to send Voice mails from my existing Windows application to an Asterisk system. I'm new to Asterisk and to VOIP. Could you please guide me with this? Regards, Shyamala -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101108/3238c1d5/attachment.htm
2010 Jun 06
1
Assign dhadi channel to several groups
Hello guys, I was wondering if it's possible to assign a dahdi channel to two diferent groups. Thanks Adolphe Cher-aime From my Iphone
2011 Feb 11
6
On-Hold Music
Hi gang, In 500 words or less (if possible), please explain what is a legal music-on-hold file? My boss hates the stuff provided with the distribution and I figure that I'm asking for trouble if I take my Les Mis tracks and run them through Audacity and SOX to make new files. Thanks in advance Danny Nicholas -------------- next part -------------- An HTML attachment was
2012 Nov 03
2
dahdi 2.6.1+2.6.1 compile fails
I am trying to compile a dahdi module from checkout: svn co http://svn.asterisk.org/svn/dahdi/linux-complete/tags/2.6.1+2.6.1 with ubuntu 3.5.0-17-generic and gcc 4.7.2 Error on compile is: oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_conf_bridge.c:3870:47:\ error: 'NULL' undeclared (first use in this function) This is identical to the error reported in this patch fix:
2010 Nov 09
1
SMS Gateway
Hi list, Anyone has some guidance in how can I project a SMS gateway with Asterisk. I mean, some good web link,pdf or something like that? Thanks in advanced!!Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormirandaru -------------- next part -------------- An HTML attachment was scrubbed... URL:
2010 Mar 10
1
Diaplan reload command not working
I am a complete newbie, completed editing the extensions.conf file, having problem reloading my diaplan via asterisk console, tried to reload it with diaplan reload command, but it says command does not exist. Please help _________________________________________________________________ Hotmail: Powerful Free email with security by Microsoft.
2010 Nov 08
3
Get the Uniqueid of Action Originate in the AMI
Hi to all. I'm begin a use the AMI and i have the need to get the uniqueid from the call i have generate using the Action Originate. Anyone can help me? When I generate these commands: action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr The only response I get when the call is answered, is this: Response: Success Message: Originate successfully queued Thanks a
2010 Jul 16
6
Video IVR Asterisk ?
Hi Is it possible to receive video calls using Asterisk and then process them as an IVR ? One of our clients wants to set-up a video IVR system in the US and we are evaluation possible options. Also, what is the bandwidth of receiving a video call in US ? What protocols and codecs are supported and does it work on DID numbers ? Can I rent a hosted solution for this ? Thanks in anticipation of
2008 Feb 08
10
Rsync 2.6.9 does not skip any files based on modification time
Hi I am trying to rsync some ghost images from a windows client running Windows XP to my Linux server. The problem is that rsync sends the complete files again even if nothing changed on the client side. The only way to avoid this is to use the "-c"-option but this takes nearly as long as uploading the files would. The server is running rsync-2.6.9, /etc/rsyncd.conf looks as follows.
2011 Apr 13
11
Realtime SIP & peer status
Hello, I'm using SIP realtime with MySQL DB. Is it possible to get the status of the SIP peer (free / calling) from this realtime DB ? If not, is there another way to obtain the call state of a SIP peer ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2011 Mar 09
5
One Way Audio
I am having trouble with no return audio on inbound calls. I have been working on this for 18 hours and even built a fresh server and moved everything over and I am getting the same results. I need someone that can help get this resolved tonight. I can provide access to all machines involved. Please email me at tim.compnetwork at gmail.com if you can help. -------------- next part --------------
2006 Apr 04
6
Loading module chan_zap.so failed! PLZ help me!
Hi, I' ve just connected a carte X100M to my asterisk server running zaptel-1.2.5, libpri-1.2.2 and asterisk-1.2.6 on SUSE 10.0. When I make modprobe wcfxo and modprobe zaptel I haven't any error, I have also chan_zap.so module existing in /usr/lib/asterisk/modules. But, when i run ztcfg, it shows me this: Zaptel Configuration ====================== Channel map: 0 channels configured.
2006 Jan 30
3
How many digium cards per server ?
Hello, How many digium cards is supported per asterisk server ? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com