similar to: extensions.conf sorting

I am looking for a supplier of Digium hardware in Canada. Any suggetions? 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, 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, I've been looking at the various SIP VoIP service providers and their plans. I understand that Asterisk can be configured as a SIP client to access, for example, a BroadVoice account to access the PSTN and discount LD. I see that a lot of the features provided by SIP VoIP service providers are really not needed since Asterisk will provide them locally. I have no plans on dropping my

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

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 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

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

Is there a way to match the last 7 digits of an extension? So that 1008014445454 8014445454 4445454 Would all match? I have looked at extension matching and I can't figure out how to do this:-) Thanks in advance! Chris -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I've just upgraded our CRM and it has an Asterisk Integration Module that I would like to test out. The CRM is running on one of our hosted servers in the cloud. The Asterisk server is running in my office. I am running Asterisk 1.4.21.2~dfsg-1ubuntu3. Reading the page http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf got me a little concerned

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

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

We are using Bellsouth 8867 phones on our TDM400B FXS lines (asterisk-1.0.3). It has a "Voicemail" light, which appears to be MWI (according to the manual it works with voicemail from the telco that sends a FSK signal). The dialtone stutters when a line has voicemail, so I know that I have the mailbox setting right in zapata.conf, but the light doesn't go on. I am also getting

Hey Everyone; I inherited an Asterisk box where the dialplan is a real mess. ( I would actually be embarrassed to post some of the stuff!) So, here is what I need to do - and again, I am looking for fishing nets and places to cast them - if I don't figure it out, I will never freakin' learn! I have several users configured (101, 102, 105, 155, 211, etc). They are all in different

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

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

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

The subject says it all. A couple of my sons have very annoying friends that tend to call ALOT. I usually don't like to answer the phone but these kids keep calling back with in 2 minutes of calling. I'm sure someone else has this problem and maybe using * to do a callerID match and block? Even add logic that if they called so many times in an hour? Or in my case, make it a