similar to: Providers for PSTN Access

I am looking for a supplier of Digium hardware in Canada. Any suggetions? Thanks, Adi

Hi, What is out there in terms of SIP enabled handsfree speakerphones? Looking for something that works well and also fits a low budget. I am used to using a Cisco 7940. It is a great phone but a bit expensive. Thought about the Polycom SoundPoint 300 until I realized that it does not include speakerphone functionality. Thanks, Adi

My PSTN line doesn't allways hang up properly after it goes to voicemail. The problem occurs when a caller hangs up during the initial greeting. Even though the hangup accured, voicemail continues to record, usually a fast busy and/or a teleco generated "please hangup now" message. After the voicemail.conf 'maxmessage=180' expires the line simply stays offhook. The hardware

I admit that this might be some very basic question... How do I obtain Asterisk 1.0.3 from CVS? Does '-r v1-0' get me 1.0 or 1.0.3? Thanks, Adi

Hi, Where can I find information on H.323 for Asterisk and/or integration with Cisco CallManager in particular? <http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration> I have oh323 working on Asterisk. Since the CallManger I am working with is running 3.3.3 I cannot use SIP... Thanks, Adi

Hi, How well to the Cisco 7905G or Cisco 7912G phone work with Asterisk? Cisco claims both phones do SIP. I was strongly considering Polycom phones. However, it appears to be quite difficult to obtain support or firmware for Polycom phones. On the other hand, I find Cisco is very well supported. Thanks, Adi

Hi all, I have a Cisco ATA186 connected to an Asterisk Server (SIP) The dialplan uses 1XX for local extensions and XXXXXXX for external numbers, where the first digit is always different than 1. In this moment, when I dial 123 for example, ATA waits till timeout before dialing that number. The same for the longer one. How can I do to make it dial imediately when 3 digits starting with 1 are

BroadVoice sells a wireless SIP phone for $149. Does this phone as sold by BroadVoice work with Asterisk or is it a locked down device like the Vonages ATA186? Adi

Hi, How can I setup Asterisk to place calls if the same dial pattern can be routed through several PRI gateways. I have one way that I tried: exten => _9737XXXX,1,Dial(SIP/${EXTEN:1}@172.17.99.5) exten => _9737XXXX,2,Dial(SIP/${EXTEN:1}@172.17.99.6) exten => _9737XXXX,3,Dial(SIP/${EXTEN:1}@172.17.99.7) exten => _9737XXXX,4,Congestion exten => _9737XXXX,102,Busy

The situation is that I want to have certain extensions that can only call nationwide and others that can make international calls (with a different provider). I suppose, you could also see this as a scenario that could involve multiple internal extensions that get matched to specific outbound trunks. I have tried to find appropriate commands in the Asterisk documentation, but could find nothing

I am wondering how Asterisk selects codecs between devices. For example, in my sip.conf I have: disallow=all allow=ulaw allow=alaw allow=g729 Does the order matter? Does it mean it will try each codec in succession and use the first that both endpoints support? Thanks, Adi

I am experimenting with oh323 channels and h.323 gateways and a Cisco CallManager. I am not using a gatekeeper at this time. Is it possible to place calls coming into Asterisk from specific peers into specific contexts? In iax.conf eaxh peer has a context in which I can specify the context an inbound call will be placed in. I don't see anything like this in the oh323.conf file or the oh323

It appears that no matter what I place into the <encode></encode> section of my ices2.xml, the stream is re-encoded. If I remove the <encode></encode> section alltogether my ogg files are streamed as is but the status page no longer shows the correct information. So what is the proper way to configure a stream that just 'passes through' the ogg files at whatever

I am looking at some lower cost phone to use with Asterisk. What is the ArtDio IPF-2000 or the Sipura SPA-841 like? Also, I see voipsupply.com has an ArtDio IPF-1000 listed, is this a new or an old model? I cannot find any information on it. Adi

I had my asterisk configuration working very well with broadvoice, but it stopped working this afternoon. I plugged the Cisco 7960 phone I used for my origional signup (just a few days before they offered generic BYOD) and it works fine. I did notice it seems to do all of its comunication through proxy.broadvoice.com (I used tcpdump). I have never contacted broadvoice about using asterisk

Hi, I connected my Wildcard X100P to the PSTN and created a context in extensions.conf which rings a number of SIP phones on inbound calls from the PSTN. When I compare the actual PSTN rings with Asterisk recognition of the incoming call, Asterisk rings my SIP phones on the third ring of the incoming call. Here is some log info: On the first ring: -- Starting simple switch on

<http://voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf%20sorting> This page on voip-info.org describes how it is possible to affect the sort order of patterns in extensions.conf. What is doesn't explain is how asterisk really does sort patterns. How does this happen? Adi

Hi there, I have a RoR app but for one of the funcationality (photo uploads) I am using PHP. I need to know how I can share/access session information from RoR in PHP since the PHP also updates the database and the entry information is contained in the RoR sessions. Please advice. Thanks in advance. Adil --~--~---------~--~----~------------~-------~--~----~ You received this message because

Hi, Did anyone here tried to use rsync between a unix machine and a NetApps Vfiler volume? I have strange case where it does create the folders on the vfiler volume but does not copy the files. Command is something like this: rsync -va dropzone pushacc@prod:/launchpad where, * dropzone is the the folder which keeps source files and folders in unix. This folder is owned by a unix account

Hi, I am Adil Arun Dangui, a 3rd year B.Tech student at IIT Hyderabad with a CGPA of 9.17. I am interested in Compilers, Compiler Optimizations and excited about LLVM and Polly. I have some familiarity of LLVM structure as we studied it in the Compiler Engineering course at IITH. Here