similar to: IAX Calls - One Way Audio

I'm going to try and lay out all the relevant information I have here in this one post. I can provide more info if necessary. ISSUE 1: Office A routinely looses connection to Office B. When typing IAX2 Show Peers, it will show as Unreachable. I issue IAX2 Reload and it will work again for 1-3 days (haven't narrowed the time down yet). My theory is that the DSL at Office2 is changing

Have asterisk@home 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring

Hi, I have a customer who experiences, once in a while, one-way audio... That is... they can hear the person they called, but the person can not hear them. The customer is connected via IAX2 to our softswitch. On the customer's end I have the following config in iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all

Hi, I've tried to upgrade my asterisk to 1.2.3 again after disastrous bug incident yesterday but when I called and the phone was picked up, there was simply busy tone... Weird, is this another bug in asterisk 1.2.3? Currently, I rollback again to asterisk 1.0.10...:( Is there any configuration change issue in 1.2.3 cause I've just used my configuration that worked in asterisk1.2.2 ?

Hello, I have just set up an asterisk box (Debian unstable) and I would like to test it with a H.323 application (gnomemeeting). When I call the demo voice menu, I can't hear any sound. asterisk says that the soundfile is played: -- Executing BackGround("H323/ip$212.9.189.172:30005/29597", "demo-congrats") in new stack -- Playing 'demo-congrats' (language

Hi everyone, I am trying to connect 2 asterisk servers via IAX, but it just fails to do so.. I'm using SIP to connect the IP phones on the LAN at the 2 different physical locations where each server resides and I'm able to communicate on my LAN via SIP without any issues. The problem is that I'm unable to make Asterisk servers talk with each other via IAX.. Here is my issue.

In my IAX config file I have: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) delayreject=yes disallow=all allow=ulaw allow=gsm jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 notransfer=yes allanrobertson- 209.23.224.97 (D) 255.255.255.255

Hello list! I?m having an issue when reloading Asterisk, I?ve had this problem in Asterisk 1.6.1.6 so I upgrade to 1.6.2.1 version, but I still have the same error. For example, I send a "reload" in Asterisk CLI and this is the output: isb152*CLI> reload == Parsing '/etc/asterisk/extconfig.conf': == Found == Parsing '/etc/asterisk/manager.conf': == Found

Hi- I'm using IAX between two boxes, where one box is located in US and the second in Europe. I'm trying to achieve the best voice quality and mainly reliability between these boxes and looking for hints and experience of others. Facts: - Asterisk 1.0.7 - RTT varies from 130-170 ms, depends on time and actual Internet throughput Questions: - What is the sugested codec for such setup?

I have a customer who wants their receptionist to input the users' long distance PINs for the because they use each others pins. I am having trouble coming up with a way to do this because of creating a channel between the user and receptionist, dropping the channel and its variables and creating a new one for the actual long distance call. Any advice is really needed. 1. User Dials Long

I need to debug a call quality issue with remote users on the other end of a satellite link. The symptoms are: we here on the Internet side can hear them just fine. On their end, things work sorta OK most times, but they often suffer from severe dropouts and digital warbling, both of which I attribute to them missing packets. Often times they can't make out a word we are saying while we can

Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's

I have a Polycom IP600 serving as a receptionist phone. We developed a call manager via c/gtk that runs on a touchpad. It allows them to transfer calls, transfer to voicemail, page, etc. The problem is this: When paging another phone from the touchpad, I have to open a channel to the receptionist phone. This rings the receptionist phone. When she picks up, it then pages the desired person. This is

Hello all, What my app does is accepts a call in on a Dial-In Number (DID) via IAX, and then prompts the caller for the top secret password (123) and then authenticates the user and prompts them to dial in the number they'd like to call. Once they press pound after dialing in the number it will read it back to them, if they press pound it will attempt to connect via the second IAX provider,

Hi, Quick question. With an old phone system a receptionist receiving a call has 1 button to push to transfer calls to a specific extension, with Asterisk, a receptionist would actually put the caller on hold, pick up another line, call the extension, ask if the person is available, hang up pick up the caller again and transfer. To me it's seems a long way to simply do a receptionist

hi We take calls inbound via SIP from our Cisco PSTN gateways, and pass it to customers using IAX (they run their own asterisk servers). We've noticed that asterisk is transcoding the call into a different codec, if the customer prefers a codec different to that which our cisco gw prefers. As such, the quality of the call can degrade. We'd rather asterisk just passed through the RTP

Hi! I've got a number of extensions (about 50) on a working Asterisk setup. For each user, I have two extensions configured (for example 11021 for a Cisco 79XX phone and 11022 for X-Lite), and a ring group that ties the two extensions together (for example, 1102). Reason being that if the user is away from his/her desk or working offsite, they can answer the soft phone on the PC. From

Hey all... Scenario (INTERNAL) 1 Call comes in to receptionist and gets transferred to someone 2 No one picks up that transfer 3 Call goes back to receptionist Now when the call goes back to the receptionist, how can I change either the ringer, the callerID or both? * TIA

I am looking at placing a system in an office with a central receptionist, and phones for each individual employee thereafter. Could I use a Snom 220 with additional keypads to view if the lines are in use by the other employees? Fred is in sales... A call comes into the receptionist and they transfer the call to Fred. The receptionist can tell Fred is still on the phone by viewing the assigned

Hi, We are looking towards a 8+ CO line setup (20 extensions) in our office but we do not want an IVR(auto-attendant) feature. All incoming will be answered by a receptionist. I have read the multi-line configuration for cisco 7960 thread in this list but that way I believe we could only display 6 incoming lines. What will happen to the rest? Does the expansion module for the cisco 7960 work