similar to: Caller ID transforms

Hi All, We're using Asterisk 1.4, and Cisco phones exclusively (mostly the 7961G, but a few 7911Gs and one 7912G for the time being-all running the SIP firmware image, plus a few analog extensions until the next capital funding cycle). Each user has a phone at his or her desk, but there are also a growing number of "common area" phones (hallway, kitchen, conference rooms, data

Hi, I have been looking for an example of accomplishing this, but I've been unable to locate something similar to what I'm trying to do. Here's the scenario: Users caller ID is set to their internal extension (200-250). This is set in sip.conf for each user. Each user has a local DID as well (hosted through Vitelity, for example (555)111-2222). The problem is that this extension was

Hi, I'm using asterisk 1.4.22-4 in Trixbox with snom 360 phones. When calls come in on our POTS lines, the caller id shows up like "555-555-1234 at 192.168.1.10" where 555-555-1234 is the correct phone number and 192.168.1.10 is my pbx server IP. This format does not work for redialing on outbound calls. While there may be an outbound dialing change that could be made, it

Set(CALLERID(number)=XXXXXXXXXX) works here. Also check with your VoIP provider what format they want for the number. (I believe) most accept a 10-digit number, but I seem to remember reading about the odd provider that wanted a leading "1". On Thu, Jun 18, 2015 at 11:47 AM, D'Arcy J.M. Cain <darcy at vex.net> wrote: > On Thu, 18 Jun 2015 13:45:10 EDT > kenner at

My first post here, so a brief intro: I'm somewhat new to Asterisk, but have been working with SIP in depth for about 3 years. I studied Asterisk for about a year and have been experimenting with it hands-on for the past month or so. I've done 6 Asterisk installs in wildly different roles/applications, some of them test systems, others in semi-production, so I know a little bit about

Hello, I'm trying to figure out how to change the redial, thus far if I hit redial it will redial the last called I made that was answered, not the last call I made that was not answer. I'm using Asterisk 1.8 Thanks, Motty

Hi, What is the best way to implement Automatic Redial on No Answer ? Looking at http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-12.txtI can see how Automatic Redial on Busy could (should) be done. How would you do it on No Answer ? Is there any event you should SUBSCRIBE to so that you're notified that you're callee is available ? What if you ask to be notified

I have an issue with the Snom 360's (any firmware) and asterisk call files. When you setup a call using a call file from Asterisk and the call is connected, Asterisk will start to redial the call after about 5 minutes when the conversation is already ongoing. (Annoying and it can only be avoided by disabling call waiting) I tried to reproduce the problem with a GrandStream phone and a

In principle, transform passes are not responsible to inspect @yyy is not aliased to @XXX. I guess "opt -basicaa -{any-transform-passes}" would help you. You may also try "opt -{any-analysis-passes} -aa-eval}" Note, opt is the tool not for users but for developers. He does not invoke unspecified passes automatically. ...Takumi 2012/8/11 Jun Koi <junkoi2004 at

Hi all, I have an external application commanding asterisk by AMI and AsyncAGI. I also have a dialplan like this: ; AsyncAGI extensions exten => _8.,1,Noop(entering in AGI loop at 8 ${EXTEN}); exten => _8.,n,AGI(agi:async); exten => _8.,n,Hangup(); ; Meetme extensions exten => _1.,1,Noop(Conference ${EXTEN} ${CONTEXT}); exten =>

Chris Mason (Lists) wrote: > Cory Andrews wrote: >> >> >> >> >> IP430, will sit between the IP301 and IP501, IP430 will have dual >> Ethernet, PoE, and full duplex speakerphone. List price (MSRP) $239 >> street price should fall likely between IP301 and IP501. >> > That looks great, the 301 is almost useless due to the lack of speaker

Hi, I apologize if this is a duplicate question; I didn't see it previously on the list. I'm using llvm 1.6 and my setup is as follows. LLVM is installed at /usr/software/llvm and the cfrontend is installed at /usr/software/cfrontend. If I cd to /usr/software/llvm/lib/Transforms/Hello and compile the Hello transform I have no problems. However, if I copy the Hello directory to my

Asterisk SVN-trunk-r7353M (will be moving to 1.2.2 this weekend) E1 connected to Sangoma A102 SIP phones (Cisco 7960) I've been making a call from my mobile to the office, when, suddenly the conversation is terminated and replaced by a "fax-type" sound. This has happened to me several times over the past year, so it's not the version of asterisk (we've had cvs trunk and

Has anyone ever tried to roll out a *66 auto-callback-redial feature on asterisk? I'm sure that implementing this for outbound Zap calls would be a nightmare, but what about something easier, like internal extensions? On my old Panasonic key system, it used to be such that, if the called extensions were busy, you could press 6 while hearing the busy signal, it would beep twice and hangup.

B.H. Hello, all. We have a dialer software that runs outgoing telephony campaigns. We have been using it successfully with PRI cards, now we're evaluating it's use also with a SIP trunk. Most of the things run perfectly good without a need to change anything except for dial string, but there's some strange problem with asterisk interpreting SIP result codes. Our software is written

Hi list, I am having some difficulty implementing a certain dialplan where the following happens. If the first Dial() is not answered, I want to play a small greeting then ask the caller to either hold the line (try calling again) or press 1 to leave voicemail. exten => s,1,Dial(${BLAH},10,Tt) ; Dial 10 sec exten => s,2,Answer exten => s,3,Playback(greeting)

I have an asterisk box and SIP / IAX2 phones. To call out, users have to add 0 (zero) before a real telephone number. That means, that if they want to call someone that has a number 123456, they have to call 0-123456. Simple, right? This has a serious drawback though - when someone calls us from the number 123456, we see the callerID 123456, and we're unable to use the callback/redial

Hi, I know it sounds weird, and this is part of the reason I have not reported that sooner. As I upgraded from 1.6.2.x to 1.8.x several months ago I am experiencing this problem. If a call is initiated from a DAHDI extension after no DAHDI extensions were used for some time, arbitrary DTMF digits are skipped and the call fails. If the call is redialed it goes through. Normally just one (1)

For those that are interested, here is my report back to Sayson on the 480i ---------- Forwarded message ---------- Date: Sat, 31 Jul 2004 22:03:31 -0400 (EDT) From: xxx@bgcfreedom.com To: xxx@sayson.com Subject: 480i User Feedback With Asterisk Seshu, I am using a 480i, and I am impressed with the phone on a whole, but obviously the firmware is lacking. Details follow. Hold button works, but

I was wondering if anyone knew of a way to create a busy-redial feature in the * dialplan? For example, you try to call 12125551212 but the number is busy, so you hang up and dial *XX12125551212 and hangup again, then * would continue to retry calling the number until either it rings or a timeout is reached, if it rings * then calls back the exten that made the *XX call and bridges the two