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Hi Friends, i have installed and configure asterisk-1.8.0. When i have tried asterisk start get below errors and not able to start asterisk. *FD 32767 exceeds the maximum size of ast_fdset!* Thanks in advance. -- Best Regards, Rajnikant Vanza -------------- next part -------------- An HTML attachment was scrubbed... URL:

i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ -------------- next part --------------

We have the possibly rather unique setup where we have cell phones posing as SIP devices. The SIP registration for those unfortunately doesn't go away just because the phone is off, since the registration is done by our cell-phone<=>SIP gateway, and that gateway has no way of knowing whether the phone is on or off. This is usually ok, but it gets problematic if the cell phone is a

Imagine that you have this code: exten => _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})})) If ${QueueName} happens to be unset, this will cause a warning: [Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187 queue_function_queuewaitingcount: QUEUE_WAITING_COUNT requires an argument: queuename The obvious solution: exten => _X!,n,ExecIf($["${QueueName}" !=

Is there a proper and accepted way to go about setting an E164 compliant caller ID (ANI) ? Currently, we're using just the Set(CALLERID(num)=XX) where XX is some E164 compliant number like 3539146632431 or some such. Is there another way we should be doing that or is that proper? N.

I have a problem with asterisk-1.2.13, where it retries SIP INVITEs too quickly. It happens when qualify is on, and the server it tries to reach is only 1ms away according to qualify. The time between the first SIP INVITE and the 7th (last) is then only 64ms, and that can be too short for the peer to react. I reported this bug in much more detail in bugs.digium.com, but the bug is gone now

Hi, I feel I've read a thread about this previously but I couldn't find it. Is there way for an Asterisk server to check if a sip phone is forwarded without bothering phone's user ? I was thinking of some Alert-Info option that would let the phone reply with a 302 Moved Temporarily or 182 Queued message and not let the phone ring or display anything on its screen. So that, you could

Hi! What are the typical ways to work around the 64 groups limit? thanks klaus

I have a setup with a number of customer Asterisks with T.38 enabled. This works quite well for each customer sending faxes between branch offices. They all have a SIP trunk to a central Asterisk, which connects them to the PSTN through various providers on dedicated lines. I cannot enable reinvite on those SIP trunks, because that would allow calls from the customer's phones to get

I have two Asterisks connected using SIP. One is acting as a SIP "server", the other as a SIP "client". This almost works; but calls from 50607795 are rejected with this error: check_auth: username mismatch, have <50607796>, digest has <50607795> On the "client" I have these accounts configured in sip.conf: register => 50607795:test at

Hello guys, I need to be able to throw cdrs on more than one servers at a time. Please let me know how this can be done. Thanks

Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Doug.

Hi, Is it possible to assign a plus sign on the callerid(num) ? currently this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing "bs523450017" instead of +6523450017. i tried putting it inside double quotes CALLERID(num)="+6523450017" telco says the same thing. is this possible? thank you Regards, nhadie

...transport is >> usually more expensive and high-margin product. >> >> Do you have a routed IP interface on your side? If so, what equipment is >> it on? It's not the switch, as the switch is Layer 2. >> >> >> On Fri, 13 Feb 2009 10:09:03 +0100, Benny Amorsen >> <benny+usenet at amorsen.dk> >> wrote: >>> Vikas <topgun9 at gmail.com> writes: >>> >>>> The ISP said that they ran a fiber optic wire to a media box at our >>>> office and from there there is a RJ45 to the switch. They bring no n...

All, has anyone any experience in using Wifi smartphones as SIP clients? Does this work properly? What models/brands are optimal for this (in terms of ease of use, battery life etc)? Thx!! B.

Anybody care to muse on Wi-SIP vs. SIP-DECT? My limited research indicates that none of the WiSip phones will ever be able to match the performance of DECT phones. Maybe I'm wrong but a Wi-SIP phone seems like a DIESEL sports car. There is nothing wrong with the technology, but it seems like a shoe-horned fit into the requirements of a wireless endpoint. DECT uses a wireless radio layer

My provider has a Coppercom switch. I have included the authentication information they gave me. How would I structure this in Asterisk to the registration and the entry in sip.conf? User Name - 8159093010 Password - XXXXX No Pin Proxy - sip.essex1.com (10.1.3.2) Outbound Proxy - proxy.essex1.com (63.164.210.14) Change setting to use "outbound Proxy" ---------- Mike Hammett

Hello! Most are probably bored seeing another letter about this, but I've put in a fair amount work on a spec for rewriting the CDR system in Asterisk, and I have some questions: First, please look at what I've written so far: svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs and look at the file "CDRfix2.rfc.txt" in the RFCs dir. The spec SIGNIFICANTLY alters the way

As I did two years ago, "I'm posting a new thread with the "Top Posting" subject" rather than hijacking the "Paging for Praying" thread. Two questions: 1. Steve K: What do you mean by "/coat"? 2. How do we change rule #5? --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax -------------- next part

I'm trying to understand why vendors keep making 100Mbps integrated 1-port switches in their hardware SIP phones. Even the recently-announced D40 and D50 Digium phones are limited to 100Mbps. Only the more expensive models (like the D70) can run at 1000Mbps. However, you can't expect a firm with hundreds of extensions to buy the most expensive model... And gigabit speed is important when