Re: [Asterisk-Users] RE: Cisco 7960G phone crashes during SIP upgrade

I can confirm that the Cisco instruction for installing/upgrading to the 7.3 SIP image do not work. When I originally installed SIP on some brand new phones Cisco TAC indicated that a phone has to run 6.3 before it can be upgraded to 7.3. Loading the 6.3 SIP image has been a success. I've had no p

Re: [Asterisk-Users] Cisco 7960 Message Light on multiple phones

> I believe the current implementation for vm notification is to use > a sip 'notify' message to turn on the mwi, and the sip protocol > implementation within * does not support sending 'notify' messages > to multiple phones. (E.g., how would * even know how many phones > you are trying to ring via

Re: [Asterisk-Users] Cisco 7905 automagically sends to VM (again...)

I put this into the configuration file for the 7905G. Solved all my forwarding to voicemail issues. # Some other defaults ForwardToVMDelay:4294967295 Adi On Tue, 25 Jan 2005, Alen Salamun wrote: > Hello All! > > First thank you I solved the 7905 ForwardToVM thing if user didn't > pick u

Re: [Asterisk-Users] Cisco 7940/7960

> I also have loaded POS3-07-3-00 and hitting any numbers does nothing. I > am using the default dialplan.xml file and a really basic SIPxxx.cnf > file. This is the same on a couple of phones I am trying. Any ideas? I am running SIP image 6.3 on Cisco 7940. The same here, I have to pickup t

[Asterisk-Users] SIP USB Phone?

There are a number of Skype USB phones available. Are there any when connected to a Windows PC can access Asterisk? Adi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE

Re: [Asterisk-Users] Autio cut off at beginning of call

My wife brought to my attention just yesterday that this is happening on all my inbound PSTN calls. I am using a ZAP interface, not IAX. Adi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk

Re: [Asterisk-Users] IAX outgoing redundancy

Hi, I have my IAX peers setup in iax.conf and use a qualify statement to see whether the peer is up or not. I then use a macro for each peer. This way I can take appropriate action depending on the DIALSTATUS variable. exten => _9011.,1,Macro(gw-voipjet,${EXTEN:1}); VoipJet.com WORLD exte

Re: [Asterisk-Users] Cisco ATA186 and Asterisk dialplan

> I have a Cisco ATA186 connected to an Asterisk Server (SIP) > The dialplan uses 1XX for local extensions and XXX 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 sa

[Asterisk-Users] IAX peering between two Asterisk servers, how?

How do I setup IAX peering between two Asterisk servers? I found a few examples for the IAX client side that conects to a service provider. But what does the service provider end look like. I would also like to use md5 authentication. Adi ___ Asterisk-Us

Re: [Asterisk-Users] Multiple gateways for same dial pattern

Hi, I just tried this using a couple of iax peers and it works quite well. But I did have to alter my dialplan. In my iax.conf I added 'qualify=5000' for the nufone and voipjet peers. [macro-gw-voipjet] ; ; This is the VoipJet IAX peer ; ; Use: Macro(gw-voipjet,${EXTEN})) ; Requires 'qualify=' st

RE: [Asterisk-Users] Multiple gateways for same dial pattern

> I have a similar problem. I asked the same question in a message to the > list a few days ago titles "IAX outgoing redundancy". It would seem > app_dial would need to have some code added to it to have two different > kind of timeouts, one an answer timeout (which is the current timeout in > the

RE: [Asterisk-Users] Multiple gateways for same dial pattern

ipJet/NuFone/LookieLoo and if I set them in order of my > preference, I've never had the primary fail so I've never witnessed this 60 > second delay. But am interested in what solution you discover. > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[E

[Asterisk-Users] Multiple gateways for same dial pattern

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 => _9737,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) exten => _9737,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) exten => _9737,3,Dial(SIP

Re: RES: [Asterisk-Users] chan_oh323 & gatekeeper

That's correct, just send FXO calls to the Asterisk box. Calls from H.323 sources will go into the context specified in the oh323.conf file. Adi On Wed, 5 Jan 2005, Humberto Aicardi wrote: > You're right it works, but how about receiving calls, how can you register > so the FXO gateways knows wh

Re: [Asterisk-Users] chan_oh323 & gatekeeper

> Until now I have used only SIP & IAX2 with success and understand > them pretty well. The point is that someone has asked me to configure an * > box for them, the problem is that they want to use H.323. I have already > compiled and tested the chan_oh323 with asterisk and works. The problem

Re: [Asterisk-Users] oh323 context for peers

> Then in extensions.conf or better in a file like oh323peers.conf > included in extensions.conf switch to "contexts per peer" via gotoifs: This will work ok for my purposes, where I work with CallManager and gateways. Thanks, Adi ___ Asterisk-Users mai

[Asterisk-Users] oh323 context for peers

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 inbo

[Asterisk-Users] extensions.conf sorting

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 ___

Re: [Asterisk-Users] BroadVoice WiSIP with Asterisk

Any idea where it can be purchased. Adi On Sun, 2 Jan 2005, Mike Dent wrote: > I've heard good reports of the Senao Wi-Fi SIP phone. > Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asteris

RE: [Asterisk-Users] BroadVoice WiSIP with Asterisk

After some reading I see that DECT requires dedicated base station hardware. I am looking for something to work with established 802.11 infrastructure. I've used a Cisco 7920 on Cisco CallManager. It works quite well. Unfortunately its price is way out of line for my purposes. Adi On Sun, 2 Jan 2

Re: [Asterisk-Users] Codec Selection in Asterisk

Yes, this makes a lot of sense and gives me the information I am looking for! Thanks, Adi > The order does matter. > > You want to offer ulaw, alaw and g729. I want to offer g726, g729 and > gsm. > > In the order you have them listed in your stanza you offer them to me. > In the order I have them

[Asterisk-Users] Codec Selection in Asterisk

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 __

[Asterisk-Users] ArtDio IPF-2000 or Sipura SPA-841

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 ___ Ast

Re: [Asterisk-Users] Softphone in German

> Hello, > X-Lite in German: > http://www.globalipphones.com/xlite Thank you! That is exactly what I was looking for. Thanks, Adi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To

[Asterisk-Users] Softphone in German

I am looking for a German language softphone. Is there such a thing? Adi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digiu

Re: [Asterisk-Users] BroadVoice WiSIP with Asterisk

> > 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? > > > You'd probably have to ask them that. Just so you know, you can buy that > phone elsewhere. It is made by Pulver Innovati

[Asterisk-Users] BroadVoice WiSIP with Asterisk

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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/ma

Re: [Asterisk-Users] Voicemail and Zapatel

> Not knowing where you are or your local telco, a lot use battery removal for > disconnect and ls ignores that and ks sees and disconnects on that signal. I am in Northern Ontario and Bell Canada is the local telco. Adi ___ Asterisk-Users mailing list

Re: [Asterisk-Users] Voicemail and Zapatel

On Thu, 30 Dec 2004, Lyle Giese wrote: > Is your X100P set for loop start or Kewl Start? I am betting loop start, > try changing to ks instead. > > Lyle This is what I have in /etc/asterisk/zapata.conf so it should be Kewl Start. [channels] ; X100P signalling=fxs_ks echocancel=yes ; Yo

[Asterisk-Users] Zapatel ringing multiple SIP devices

My incoming PSTN line is configured to ring multiple extensions and eventually fall trough to voicemail if the call goes unanswered. If a SIP phone gets picked up just before voicemail should kick in, the call quite often goes to the phone but voicemail happens as well, the greeting plays and the w

[Asterisk-Users] Voicemail and Zapatel

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 voicem

Re: [Asterisk-Users] Shorten the recognition time of rings on Wildcard X100P

Hi Michael, > > 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

[Asterisk-Users] Shorten the recognition time of rings on Wildcard X100P

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 incom

[Asterisk-Users] Asterisk Cisco CallManager Integration

Hi, Where can I find information on H.323 for Asterisk and/or integration with Cisco CallManager in particular? I have oh323 working on Asterisk. Since the CallManger I am working with is running 3.3.3 I cannot

[Asterisk-Users] Skinny not working?

Hi, I am using Asterisk CVS-v1-0-12/08/04-23:34:49 and I am getting these messages in my log: Dec 15 12:05:57 WARNING[11931]: Unable to get our IP address, Skinny disabled Shouldn't bindaddr=0.0.0.0 automatically bind to all available interfaces? Thanks, Adi ___

[Asterisk-Users] Digium Hardware in Canada

I am looking for a supplier of Digium hardware in Canada. Any suggetions? Thanks, Adi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium

[Asterisk-Users] Asterisk and Cisco 7905G or Cisco 7912G

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.

Re: [Asterisk-Users] Asterisk and Cisco 7905G or Cisco 7912G

> I can't speak for the 7912G, but I have several 7905G phones and these > work perfectly with Asterisk. This is great! The 7905G is what I have in mind for a plain basic phone and the 7940 where a speaker phone is needed. > The firmware is easy to obtain if you have a Cisco support agreement - >

[Asterisk-Users] Asterisk from CVS

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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To U

[Asterisk-Users] Providers for PSTN Access

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 ar

[Asterisk-Users] Handsfree Speakerphone

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 spe