similar to: IAX2 channels denoted as '(None)'

hi, i want to call PC2PC between to IAX client without authentication i want to allow every user to use PC2PC no any password required. Please let me know what i have need to do in IAX.conf or any other file to allow any user to call Pc2Pc. My IAX.conf [guest] type=user context=default callerid="Guest IAX User" My extensions.conf [default] exten=>_.,1,Dial(IAX2/${EXTEN})

hi, i want to call PC2PC between to IAX client without authentication i want to allow every user to use PC2PC no any password required. Please let me know what i have need to do in IAX.conf or any other file to allow any user to call Pc2Pc. My IAX.conf [guest] type=user context=default callerid="Guest IAX User" My extensions.conf [default]

Dear Sanjay, Sorry sanjay i miss to explain completely. My PC2PC mean is Dialer2Dialer i want to allow call between Dialer with out any registry and authentication through IAX. so i need to setup Asterisk accept calls from any user and users can call to each other without any password and registration. please help how can i configure Asterisk using IAX in this regards. thanks, Asif Message: 9

Hi, I'm trying to mount with smbmount a WinNT4.0 share resource from a Linux running Samba 2.2.2.=20 WinNT box is installed with code page=3D850. The resource name includes grave accented characters, in particular: =E0. I'm trying to smbmount with: $ smbmount '//NThostname/of=E0rs' /mnt/ofors -o ip=3Dxxx.xxx.xxx.xxx,username=3Dofors,password=3D******,workgroup=3Dwwwwww where:

Hi all, Some time ago I posted an issue regarding the hangup of active calls from the CLI and someone told me that "soft hangup" should work. Well, in fact it does work, but only if the channel is known, i.e. it doesn't work for zombie channels. For example, I have this scenario (CLI output of command "iax2 show channels") IP-AM-PBX*CLI> iax2 show channels Channel

Hello all, I have seen some chatter again about DTMF. I see most of the talk about DTMF around not being able to get an external IVR to recognize digits, not a big issue for me at this time but sill interesting. My issue though, is with talk off on a zap channel. It seems to be getting worse or maybe my patience is thinning. All my calls go out and come in pstn through an FXO as I do not

I am having some problems getting TelIax service to work with *. Outbound calls work just fine. When I try an inbound call the phone rings and there is no audio. Upon further investigation "iax2 show channels" indicates that the codec is "unknown" The provider confirmed that they are set for ulaw and so am I. Does anyone have an idea what could be causing the codecs to

Hi! I'm trying an asterisk-FWD connection. I'm using X-Lite OR SIPPS as the IP phone. I configured the X-Lite and SIPPS to use GSM codec. Whe I call FWD, I get this info on the channels when the call has not been stablished yet: sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.246.69.223 613 1770bf3430d 00102/00000

Dear All, During CentOS Boot-Up user can See all of the standard messages within this front-end by clicking on the 'Show Details' link and hide the standard messages within this front-end by clicking on the 'Hide Details' link. Previously when i click the "Show Details" Menu it will display the system boot-up standard messages.I do not know what

Hi there I have a question regarding IAX jitter. I have 3 users on a LAN dialing into a Meetme conference on an Asterisk box which is also hosted on the LAN. I have set jitterbuffer = no and tos = lowdelay. Now, for 2 of the users the audio is fine, but for the 3rd user there is intermittent break up in the audio when they are receiving. I have had a look at "iax2 show channels" and

Hi! I'm running a RedHat 7.2 server with Samba 2.2.3a - but recently I have a problem that mapped drives on Windows 2000 workstations get disconnected and have problems reconnecting (error message is something like "local device name already in use" when I try to browse the drive in Windows Explorer). Any idea what causes this problem ? Regards, /Brian

I seem to be having a problem that I have narrowed down to a disagreement on codec negotiation or codec setup of some kind in an IAX peering arrangement. Here's a non-ASCII art version of the setup: DID origination provider via SIP/gsm to Call routing asterisk server via IAX/gsm to Client asterisk server via SIP/ulaw to Polycom 501 UA The problem that occurs

Hi all, Below is what I did to run Asterisk in pass-thru mode: sip.conf: [general] disallow=all allow=ulaw canreinvite=yes For each channel, canreinvite=yes is enabled. No dial command has 't' option. However, it seems that Asterisk still stay in the media path and bridge the 2 end points. Am I missing something??? sip*CLI> show channels Channel (Context Extension

Has anyone seen this before? This channel is consistently present on both of my asterisk servers. Sometimes they disappear for a few seconds and then come back. It always has the same Call ID. voip1*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.102 (None) df92fb1b-8a 00101/03059 00000ms 0000ms UNKN

We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway provider supports both, so that's not a problem, and if I force him (the PSTN gateway) to allow G729 only, the outgoing call will take place with G729. The problem is that I want to have my PSTN provider configured to allow ULAW as a first priority, then G729. I did it like that: [mypstngate] type=friend

I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from

I've been trying to resolve some quality issues and I was hoping someone might be able to provide some insight. To give you an idea the calls are coming in via a SIP DID and sent out via an IAX2 connection. Latency to both the SIP equipment and IAX equipment are around 80ms with 0 packet loss accoridng to ping tests. The server is located in a data centre so bandwidth is not an issue. Most

I have the MusicOnHold feature working great when called from ATA-186 extensions. It's pretty cool. However, when I call from a BudgeTone-100 phone, no music is heard -- instead it continues the ringing feedback and acts like the call is unanswered. At the same time, I can call from (multiple) ATA-186 extensions and hear music as long as I like. How can I debug this? As far as I can tell,

Hi all, I have tried to figure out how to do this one but I think I have just confused myself more. My firewall is a 2 interface setup, the same box is my router to my uplink. I''m not using nat at all and have a public IP range behind this machine. net = eth0 loc = eth1 Most of my rules are mainly the basic HTTP(ACCEPT) net loc:111.111.111.112 SMTP(ACCEPT) net

Hi, I'm slowly getting to grips with *. My next quest is to get IAX2/FWD calls working. I've setup a fwd account and added IAX capability to it via the website. I got the email saying it had been done with some welcome text and sample phone numbers to try, such as 10001 for the answer phone. I followed the setup example on the fwd site for configuring * to work with fwd's IAX.