similar to: SIP retry time too low

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

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

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

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 --------------

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

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}" !=

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

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.

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.

A regular Dial(somepeer) to a SIP peer which doesn't reply at all seems to not time out, or at least have a very long time out. We have a set up where we can dial two different peers, a primary and a backup peer. If the first one dies completely, so that no error messages (no ICMP unreachables or anything) are returned, Asterisk does not continue in the dial plan but just gets stuck on that

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

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

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

What the minimum Server Specifications do I need to run 200 concurrent channels at the time with .WAV recording (MixMonitor)? It will be connected via VOIP sip account. Codec will be ulaw. Which UK dedicated server provider do you recommend and how much bandwidth do I need? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have two version 1.6 Asterisks running. One is a small hobbyist thing just at home, and the other is handling calls for several customers. On both, I have added the line exten => h,1,Set(CDR(hangupcause)=${HANGUPCAUSE}) to all relevant contexts. On my little hobbyist box this works perfectly; all calls have their hangupcauses recorded with cdr_adaptive_odbc and cdr_custom. On the

I see a lot of these on an otherwise idle Asterisk 1.6.0.15: Extension Changed 773[Hints] new state Ringing for Notify User 792-00041327d17e-1. Then a little while later it changes to InUse or Idle, completely randomly. It happens for many different combinations of phones and watchers. There are no calls being made, so I can think of no reason why this happens. Obviously it wreaks havoc with

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.

I have a challenge that is ending up quite interesting. I need to identify which SIP phone "touched a call last", that is, which phone did the last transfer or dialed the original call if no transfers were done. It is easy in the case of a regular, non-transfered call. Just put something in callerid= in sip.conf, and that will show up in ${CALLERID}. The same with an attended transfer,

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