similar to: Packet2Packet Bridging Questions

Hi, I have Asterisk set up on Fedora with a single SIP trunk, with a few handsets configured. The Asterisk box has both public and private addressing, so "canreinvite=no" is set on both the SIP trunk and handset configurations so I can get around the nasty NAT issues. One odd behaviour I am seeing is certain destinations are resulting in different SIP codes being sent back to Asterisk,

....I THINK. When dialing 1+10 digits, I occasionally get a telco message "You must first dial a 1....". When I look at the console, the number is being sent to the ZAP channel properly. We're talking about a couple of POTS lines on a TDM400P. I'm thinking that it may be starting the dial too early after coming off-hook because I can just redial and have it work (or not)

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of David Burr > Sent: Sunday, January 11, 2004 4:31 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive > > > We have a new contest starting today! > > The first three

> -----Original Message----- > From: Travis Johnson [mailto:tlj@ida.net] > Sent: Monday, September 08, 2003 1:05 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Cisco 7940/7960 ethernet ports > [...] > We are having a problem with Cisco 7940 and 7960 phones when > the PC is plugged into the 2nd ethernet port on the phone. It > will drop the

I have put canreinvite=no for all my internal SIP-clients in sip.conf because I want Asterisk to be in the middle of the RTP-stream so he can provide MusiconHold and so... Now, what the Asterisk CLI tells me when I make a call from my one internal SIP-phone to another internal SIP-phone is : Verbosity is at least 25 == Spawn extension (intern, 51, 1) exited non-zero on

I am working on a call-back solution where the initiating call should never be answered. I was doing this simply through the dial plan, sending a progress tone, and then dumping the channel, and firing off a DeadAGI which created a call file to make the callback. Now I've tried extending this so that an AGI is fired first to check for things - like no inbound ANI - and play a

I'm looking for an application that can monitor a channel for voice input and then proceed on. The closest thing I've found is BackgroundDetect, which expects DTMF. Here's what I'm doing: -Call file generated which calls someone and connects them to an extension. -Extension plays stuff, etc. etc. etc (not important) With digital or VoIP termination, this works fine, because *

Im trying to figure out the base way to check the callerID being sent to my Asterisk box and use it if it is a valid NANP number, but replace it with a static NANP number if it is not. (Why? I have a few carriers that require this, and a few international users - if it happens to take one of the carriers that require it, I want it to set a static number that is valid). I'm playing

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Gary Franczyk > Sent: Thursday, January 15, 2004 10:37 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] Basic Asterisk capabilities question > > > Whaaa?? So, to allow 24+ lines of dial in access, how

Hi all, Which is the practical (from your experience) limit of the number of X100P cards installed in a single Asterisk box? Asterisk can work reliable with 6 X100P cards in the same box? Anyone know when the 4 ports FXO Digium card will be available on the market? Many thanks, Dan P.S. Please do not aswer with RTFG ...tried before without success...:-))

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Jeremy McNamara > Sent: Thursday, December 11, 2003 2:19 AM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Re: * with RADIUS > [...] > > Explain why you think you really need RADIUS Accounting? >

I'm a bit displeased at the way this happened. I received an email from VoicePulse. Here's some excerpts: ------------------ >We're sending you this important update so you can take advantage of improvements we've >been making to your VoicePulse Connect! service. >We've been working hard on improving the audio quality and reliability of your Connect! >service,

I'm sorry, i send mail in asterisk-bug, but asterisk-users is better for my problem Hello, During a call with canreinvite = no, at the beginning of the call I lose 2 seconds of audio. is obvious when I call autoattendant. schema: SipPhone --> Centrex (asterisk 1.4.24.1) --> Voip1 (Asterisk 1.4.24.1) --> Operator SIP capture of voip1: - Executing [0825387205 at

Dear concern, I want to configure packet2packet bridging in asterisk. How could I do this any of the tutorial or instructions will help ? I found the setting the canreinvite=yes will do the stuff but it is not working I am using asterisk 12.3 version I am very new to asterisk please help me in doing the same. Thanks in advance. -- Regards Sameer Rathod 8109413462 -- Regards Sameer

Assuming the price of an ADSI screen phone (say, Aastra 390) was the same as an IP screen phone (say, Cisco 7960) and someone was setting up an * server for their 20 employees (each of whom would have either an ADSI or IP phone on their desk), would there be advantages to using the ADSI phones over the IP phones, or vice-versa? For discussion, let's assume that the hardware needed to

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Test A: Outside line calling in

If anyone is looking, I just ran accross an ebay auction for X100P Cards at what I thought was a very reasonable price. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3046843672&category=48483&rd=1 --------------------------------- Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software -------------- next part -------------- An HTML attachment was

Hi everybody Hope I picked the right mailing list. If not, please tell me. We've got a problem with call forwardings. It's exactly the same problem as described in bug 12708, which is resolved by now. Situation: Caller -> asterisk -> call forward to mobile (packet2packet bridge) Quote from original bug reporter: 'One issue that we have noticed repeatedly is that there is a

Hey guys Is it possible to record with mixmonitor straight into mp3. I am trying to reduce disk space and want my calls to be recorded in mp3 Instead of wav. Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: Sameer Rathod <sameer at hostnsoft.com> </div><div>Date:30/06/2014 9:23 PM (GMT+02:00) </div><div>To:

Hello, I have ISA card LineJack. I could not find any information if this card can work as fxo with Asterisk. If it can work, can somebody point me how to install it on my Asterisk box. Or maybe there is some documentation about it how to install LineJack. I will be very thankful for any help. Regards Bartosz Jozwiak ------------------------------------------------- This mail sent through IMP: