similar to: rollover simulation

Hi, Some odd behaviour here. A friend and I were talking tonight, and it seems we have both seen the same problem. We are both using aastra phones (I am using 9113is). We have a connection to and from providers via SIP and IAX. When I place a call on the local hold of the phone, and then pick them back up I can hear them, but they can not hear me. However, if I park the call,

I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this channel.The incoming route for the channel is Zaptel Channel 0. No DID or CID settings applied. My IP

Hi Folks, Looking for feedback on the cordless phones with the Aastra 480i CT handset. Is voice quality comparable to standard consumer residential 2.4GHz cordless phones in the US$30 - $50 price range, or better/worse? How about handset and speakerphone quality for the main phone? Seems like there have been various big issues with firmware in past, but is it pretty stable now? Thanks,

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Have a question about implementing Call Rollover with my current extensions.conf configuration. [macro-stdexten] exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Voicemail2(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce exten => s,3,Goto(default,s,1) ; If they press #, return to start exten =>

I am having an issue with doing FTP auto provisioning of Polycom 501's when they are behind a NAT. If I put the phone on the same subnet as the provision server it loads the configs and changes fine but as soon as I put in behind a NAT it comes up with cannot contact boot server. I have tried behind and replicated this behind a PIX 501, a Linksys SOHO router and a Motorolla SOHO router.

Thanks Curt, that's "too cool for school", any idea on when this is coming to the MS SBS platform? I use SBS for myself at home and would love that level of functionality included. Does Asterisk therefore handoff voicemail storage etc to Exchange for this level of integration? Cheers, Dean ________________________________ From: Curt Shaffer [mailto:cshaffer at

Hello, I've got 2 analogue lines (from SBC) coming into a TDM22B. SBC have put rollover from the first to the second line. The rollover works fine when handsets are connected directly to the lines (ie when Asterisk is not involved), but when the lines are connected to Asterisk, the rollover fails: the caller just hears the line ringing, and the person on the first (busy) line hears call

Just emailing the list to see if anyone out there has used Vitelity? If so what has been your experience with service, support etc? Thanks Curt

What is everyone out there doing in an all IP termination environment to change trunks when quality drops to a certain provider automatically? Thanks Curt

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

I put iptables on my asterisk box and an odd thing occurs. I allow 5060 and 10000-20000. As soon as I start iptables and make a call it literally takes 60-90 seconds before the call even starts to ring. As soon as I shut iptables off, the call goes through immediately again. Its quite odd. The call does eventually go through and talks fine but it takes sooo long to connect. Anyone have some

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, I have a simple question about incoming phone line rollovers. How are these usually done? Is this done at the phone company usually, or is this something that Asterisk or channel bank is capable of? I just need someone to give me a brief explanation how it usually works, and if someone was implementing an Asterisk system, how they would go

Does anyone out there have a recommendation for tools that will monitor the quality of VoIP systems? I am looking for jitter and MOS monitoring. I have a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms but I am looking for a little more detail. I would not be against writing something in Perl for Nagios to do but I don't really know where to start on measuring jitter

I have read different posts over the months wondering who Vonage uses for their VoIP technologies. I stumbled across this article (although it's from 2002, I think) that suggests strongly that they use Cisco. There is no telling what they might use in conjunction with this but this should clear some of the conjecture.

I have been working with a couple companies who are interested in integrating Cisco VoIP (mostly call manager express) but utilizing Asterisk for AA, VM, failover trunks etc. I have found some materials and guidance out there but I think a list and/or wiki for general asterisk integration with other vendors would be great and feel that it is enough off topic that it deserves its own space. Just