search for: asteisk

Does anyone Know if tha avaya 4621 IP phone work wiht asteisk? if it work it has featuras working Thanks Ignacio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051014/d3a65784/attachment.htm

hi My stations are behinds a firewall, the system is windows 2000 & 98, i use sjphone asterisk is on the internet gateway where is the firewall Shorewall the system is linux debian (sid) kernel 2.4.20 j do whaton http://www.automated.it/guidetoasterisk.htm (grateful Andy) to write my sip.conf but i can't call an external sip user. (an external user can call me) i try without asterisk with

Hello, I have a friend that is asking if he can use his Ericsson 3413 H.323 IP phone with Asterisk. I can't seem to find any reference to this phone on the Wiki... -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST

Got my latest Linux magazine (www.linux-magazine.com) and fetured is asterisk in home network. I've also been in contact with Novel/SUSE about their asterisk pakages. *Reinhard Max the maintainer. He has hinted at new packages for SUSE 10. The current ones work well (in production) however he is unsure about the new zaptel intergration.... but I'm keeping my fingers crossed! * -- A.G.

...sk. here are my settings, Codecs ------- Default codec - g.711u/g.711a Packet size - 20ms Negotiation - Interoperable Type - 160 DTMF ---- Inband - negotiate Outband - negotiate Payload Type - 101 when a call comes to the SNOM or when making an outdial, following warning messages are coming on asteisk, WARNING[1209214400]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[1209214400]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[1209214400]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames ... WARNI...

...ver. I place calls with DIAX. The H323 gateways only support G711A De DIAX only supports GSM When I perform an inbound call: H323 -> asterisk -> DIAX :: sound is ok. When I perform an outbound call: DIAX -> Asterisk -> h323 :: sound is terrible and CPU load is 80% When I perform an asteisk internal call with DIAX: DIAX -> asterisk IVR :: sound is good and cpu OK. Does anyone else have this problem ? Know how to solve it ? regards, Arne.

i have problem to setup application meetme. i'm using asteisk-1.0.3 and sjphone as client. ++++++++++++++++++++++++++++++++++++ monchemin --------------------------------- Do you Yahoo!? Dress up your holiday email, Hollywood style. Learn more. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/piper...

Hi, I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1 link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both the Digium and Samgoma types, as I have successfully hooked up to many PBX's and such, but I just cant seem to get this one to work. None of the 30 channels 'come up'. What signailling, crc checking, should I b...

http://lisas.de/~patrick/temp/rtp-optimierung.png Everything is working fine in my Setup, but I want Extern1 to talk to Extern2 directly whitout going over Asterisk as the uplink is slow. When I set for Extern1/2 canreinvite=yes it works, but "Intern-2-Extern" doesn't work because Asteisk gives out the private IP-Adresses of Int1/2 I defined localnet=10.0.0.0/255.0.0.0 (Private LAN) but this doesn't help. Ideas, how to handle "Extern-2-Extern" (RTP bypass Asterisk)? Do I have to adjust "nat" somwhere?

Dear list i am using Asterisk 1.2.5 with A@H . here is my problem. if i dial a number (consider 79) i have to wait around 20 seconds before my Asteisk box response. now i want to decrease this waiting time . any idea how to do that ? thanks Salaque

...round 80% idle. >> >>Is there any tuning I can do??? >> >>Besides that, Asterisk normally goes down once or twice per day... >> >>Thank you >> >>Dov >> > C F wrote: > >From what you say it sounds that the problem is not with asteisk, but >the way it's configured. Asterisk should *never* go down that often. >Asterisk as a normal PBX should run without a restart for as long as >there is power to the box, in the case of a call center if I would >hear of a restart once a week I would accept it, but still would...

hello is there a gui that would allow me to configure everything from phones, to extentions, to voice mail to basicly everything that asterisk can do? I did go to www.voip-info.org and none of the guis I saw there do the trick and the ones that come close aren't downloadable just wanted to see status on this thanks hank ---------------------------------------- My Inbox is protected by

Hi all, Our asterisk PBX, randomly restarts all the channels of the E1 connection. It sends this message "There is no D-Channel, using channel 16 anyway".Then the asteisk recive (or it thinks it recives) yellow alarms at all the B-channels, after that it restart all the channels. When restarting the B-channels it cut all the conversations that is handling at that moment. Does anyone have an idea for what it is happening? We run a asterisk 1.2.1 on a HP Proliant ML31...

...Just to confirm back to the list, the chattr +i and turning off prelinking seems to have solved the issue with the executable file being corrupted / touched. Been running fine for a week now. We have a very simple monitoring app running which we wrote ourselves which monitors some metrics on the Asteisk, but nothing near as comprehensive as you apparently use... Anyway, thanks for the reply! Kind regards Stefan

Hi, Just to clear things out.. Can asterisk transmit faxes over IP ? If not, are there any works being done towards implementing t.38 on asteisk ? Also dialing in from a mediatrix fxs sip gateway to voicemail, asterisk does not see the digits entered after mailbox prompt. I have dtmftone settings correct - inband (also tried others to make sure), however asterisk shows 'username not entered'. Any clues how to tackle this ? Chenking...

...rk I would like a prefix ( 0 ) for the classic calls and another prefix ( 1 ) for voip calls. The problem is that pbx can talk with asterisk only with S0 synchro (like a terminal) and succeeded not to make call with prefix in this mode. I also try to consider asterisk like a provider, but pbx and asteisk don't snchronyse themselves. (need pbx protocol like qsig I think). I use a old isdn card (Dynalink IS64PH Adapter) with isdn4linux and modem asterisk chanel. Is it possible to do that with my old isdn card ? else, which card type I need ? isdn active ? E1 ? which protocol need to be implemen...

All - I've a new system, that since it's been in production, has seen a few issues, that look like they should be fixed by upgrading asterisk @ home to the latest version. I was curious if anybody out there can tell me their experiences with this, and what to expect. Thanks, Rolf Brusletto

...since I want to avoid a NAT, having SER on my Internet gateway makes most sense, which led me to Milkfish on a Linksys WRT54G. Having never worked with this before I wasn't sure how it would be configured in the end, so I just "went for it". I assumed that OpenSER would sit between Asteisk and my ISTP and handle both ends of the resgistrtion. I was able to register two Xlite phones to my WRT54G and have them talk. Not so lucky w/ Asterisk. I was able to create an account on OpenSER for my DiD and have Asterisk register on the router, but how would the ITSP end know where I am? I...

Hi, I've been doing some testing on moving from 1.2 to 1.4 and one issue I've encountered is re-transmits whenever an INVITE is cancelled. I have a stateless SIP proxy in fron of my asterisk servers (all it does is direct requests to one asteisk server or another) and the re-transmits do not occur on 1.2.17 which is the current verion I have in use on my production servers. The retransmits do not occur on a 200 Ok Response. When the INVITE is cancelled the CANCEL request is acted on correctly and the cll is cancelled and the only problem...

Hi, all Can someone tell me where to tell asterisk no to use 127.0.0.1 IP (localhost)? When I am registering with VoIP providers, they get my info as s@127.0.0.1. (This is SIP registration). Also, in SIP logs, when calling I am getting things like this: Executing SetCallerID("SIP/phone2-22c3", ""CID Name" <CIDNUMBER>") > in new stack > -- Executing