similar to: Problem with 302 "Moved Temporarily" Do not disturb

Hi All I was working in an env that had some Snom 200 with release 'r' firmware i beleive, when thy put the phone in DnD, mode then released it from DnD mode, the phone does not re-register with * Does anybody else observe the behavior ?? Is is a config issue

Greetings all! I bought some Polycom 501s and got them installed, configured and running. I'm using bootrom 2.6.2 and sip 1.5.2. My problem is that when I press the Do Not Disturb button, the phone stops responding and reboots by itself eventually. Has anyone seen this problem before? Jesus Mogollon -------------- next part -------------- An HTML attachment was scrubbed... URL:

I looked on the voip-info wiki and found sparse and conflicting information on how to do this with Asterisk... My incoming lines are all on Zaptel. Is there a simple why to implement a '*363" (do not disturb) toggle via the dialplan? It would be nice to be able to pick up an extension, dial *363, and have all calls sent to voicemail without ringing the extensions. Doing it again would

That could explain why it wouldn't work on any of my sip extensions I tried it on this morning when I first read about it and thought cool the things you learn. Is there anyway to make it work on Sip extensions? Cheers, Dean -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Aaron J. Angel Sent: Wednesday, 23

Hi, Is there ANY way for me to see the status of the Polycom DND buttons in the Asterisk hints? I`m using the BLF buttons to see the status of other people`s lines, and DND should logically be somehow reflected (I don`t care as much about Polycom showing the BLF button as DND, but I do care about Asterisk hints showing it in the CLI). Right now, a Polycom phone on DND shows up as being

Hello, I 'm looking for a way to pass the '302 moved temporarily' received from the SIP device back to the SIP provider. Here is the setup: Some SIP phones are connected to an Asterisk System version 1.8. External connection to the public network is also done via SIP to a VoIP provider. Phone A has a CFW all calls to a phone number in public network (Mobile Phone) incoming call to

Hello, Some of my users have phones lacking a DND button. I need to provide an extension they can dial that will put them in DND, i.e. tell the server not to send them any calls until they get off the DND. I've researched it for almost 3 days now and tried a range of configurations. I'm hoping somebody here has an answer. Currently, I have this in extensions.conf [app-dnd-on]

Dear Asterisk-Users: Question: ======== How do I get asterisk to pass DNID/RDNIS information between asterisk machines using iax2, in a Dial(IAX2...) command ? Setup: ===== I have two asterisk boxes, MASTER and SLAVE. MASTER is running 1.2.0 and SLAVE is running 1.2.1. The main box handles incoming calls on a multiple lines (both via hardware connection to our internal PBX and calls

Hello All, I have managed to get my cisco and asterisk able to talk to one another I think. But cannot make a call from a phone behind call manager to the asterisk server. I have followed the cisco asterisk integration on the wiki. I have also setup a number 3000 for dialing for current local time and date on asterisk. I can call from a sip phone behind asterisk, no problems. The problem

; Below assumes you are using the same number for Voicemail boxes as extensions ; if ${RDNIS} is blank then GotoIf will go to extension 2, otherwise it will go to extension 102 exten => 8500,1,GoToIf($[X${RDNIS} = X]?2:102) exten => 8500,2,VoiceMailMain(s${CALLERIDNUM}) exten => 8500,3,Hangup exten => 8500,102,VoiceMail(u${RDNIS}) exten => 8500,103,Hangup ; you should now be able

Hi, I'm facing a strange problem. My setup is: Alice cellphone <--GSM--><--ISDN--> Asterisk <-- ISDN --><--GSM--> Bob cellphone When Alice calls Asterisk which forwards the incoming call to Bob, sometimes Bob sees Alice's number, sometimes he sees a default CallerID (which happens to match the dialed number and the ANI). For various reasons, Bob really needs to

I haven't been able to find much support for the RDNIS or DNID variables online. I am trying to prove a concept of call routing before we move towards development of a production system. I need to have calls routed coming into a call center based on DNIS. What type of syntax is needed in the extensions.conf file and how can I test it with a softphone (ie: can I emulate the DNIS from xlite)?

Does anyone know the mechanism by which companies like YouMail (and MNO's using their own voicemail system) are able to redirect ALL calls from a ALL subscribers to *just one* voicemail DID, yet determine WHICH subscriber did the redirection? I had always assumed this was simply done using RDNIS. In other words, the original calling party's CallerID is passed with the redirected

I'm trying to use RDNIS with asterisk, and I don't appear to be receiving any information (the value is blank). The upstream who provides the PRI says they are passing all the info through, I don't see this value coming across. I've tried it with a Verizon call forward, as well as a Nextel with the same results for both. I'm trying to use this for Voicemail. I'm using

Hi, a small question.. I'm using NTP to synch our phones with an ntp server, but it seems the Cisco 7912G (with SIP image) does not handle daylight savings time very well? Am I overlooking something or is this a known feature? I'm using GMT+1 and minutes are correct but it doesn't respect DST. SIP software seems to be: v1.02.00(040406A). Cheers, Kristof.

Perhaps I've got the whole concept wrong, but here goes: Using 1.4, when someone from the outside dials my direct line (123456), I want it to call my extension at work (SIP/456), my extension in my home office (vpn connection to corporate lan, SIP/678) and my mobile (654321). So my dialplan is thus: exten => 123456,1,Dial(SIP/456&SIP/678&Zap/G3c/07803654321,30) exten =>

Has anyone had success patching chan_sip.c so that Asterisk will recognize the tel: header for RDNIS information? exten = get_in_brackets(tmp); if (!strncasecmp(exten, "sip:", 4)) { exten += 4; } else if (!strncasecmp(exten, "sips:", 5)) { exten += 5; } else { ast_log(LOG_WARNING, "Huh? Not an

I have a couple of numbers that are diverted to a number that is conected to an isdn30 card, running asterisk 1.4. eg. 123456 => 22334455 654321 => 22334455 What I would like to know is the number of the orginal number dialled (123456 or 654321). I thought that RDNIS was the answer, but it is always coming up blank. When I did a debug on the pri span, I saw the following message

I'm preparing to roll out Asterisk for the voicemail portion of my VOIP network. This week I downloaded a fresh version from CVS of Asterisk and installed the following MySQL 4.1.7 RPMs directly from Mysql.org For some reason after I enable MySQL for CDR and Voicemail in the cdr_mysql.conf and voicemail.conf I don't get any MySQL functionality at all. It almost seems as though MySQL

Other than having stripping out IPs this is what I am receiving for my voip calls. Now I normally use ${CALLERID(rdnis) for RDNIS, this works fine calls that come in on my PRI. BUT at least from this VOIP source the To field which is my RDNIS information for these calls, doesn't actually fill into ${CALLERID(rdnis). But as you can see I'm getting the information. My question is, Does