similar to: Problem with CID matching

I notice most things say to open ports 10000-20000 for UDP for SIP, however from time to time this range isn't where Asterisk is opening the ports: We're at xxx.xxx.xxx.xxx port 8542 Answering with capability 0x2(GSM) Answering with capability 0x4(ULAW) Answering with capability 0x8(ALAW) This call has no audio, presumably because port 8542 is firewalled in the iptables on the server.

I'd like to integrate a few peoples cell phones into our asterisk system as we're walking around a data center some days and carrying another cordless phone just doesn't seem to make sense. Forwarding is easy, however once forwarded I want to be able to flash and transfer back to another party / voice mail / etc. How can this be done? As far as I know there's no way to generate

We were just testing forwarding one of our numbers to a VoIP CLEC and ran into an issue that we've seen before but never figured out the cause of. It seems if you call us and immediately hang up - or if the call is forwarded by Verizon (which stills rings about ? of one time) the Zhone detects the ring and answers the call.. The Zhone however DOESN"T detect that it's no longer

We want to test capacity of our MeetMe room. The thing that is distinct about this is that the incoming line is being delivered IAX2 to our server across the net - so Telephone -> VoIP Gateway -> MeetMe. We want to test both the VoIP Gateway and the MeetMe room performance. You can reach our MeetMe room directly at 1-301-561-9229 If you want to test with us we're thinking maybe 9pm

I'm guessing you aren't on a digital line (i.e. POTS). If that's the case you'll have problems sometimes that the dial tone isn't there as fast as it dials. I forget the pause character (p? It seems like that was pulse not pause). I'd break this into 2 steps, answer and announce something - make sure you are answering fine. Step 2 - Dial via ZAP and hopefully then you

Hello, I have a really irritating issue that I haven't had time to investigate much - I hope someone has encountered it and can tell me a solution. I didn't see anything in searching archives / sites.. When my Nextel i90c phone gets a page (2 way text message via the internet option) it has an irritating tone to get me to hear it. However this tone seems to mute asterisk (reproducible).

I see in NexentaStor''s announcement of Community Edition 3.0.3 they mention some backported patches in this release. Aside from their management features / UI what is the core OS difference if we move to Nexenta from OpenSolaris b134? These DeDup bugs are my main frustration - if a staff member does a rm * in a directory with dedup you can take down the whole storage server - all with

Try : killall -s HUP /usr/sbin/smbd is causes that smbd rereads its config. At 11:20 10.09.2002 +0200, Kurt Pfeifle wrote: >Hi, > >I have a question regarding the visible list of printers in the network >neighbourhood of my Samba server, and how to force it to become updated. >Maybe one of my settings is wrong? Maybe it is a bug? > >My problem (short):

Greetings - I have not been a long time follower of this list, but I have scanned through the last year or so of archives, after not finding much from google searches. I am hoping someone here can inform me if what I want to do is feasible, and give me some general guidance to follow so that I can continue my research and complete this task. I admin a RH3 system that is primarily a Samba

Hi, Recently I am encountering an authentication error when making a phone call between Asterisks. That call is intended as follows. (1) SIP_phone2 to Asterisk#2 (2) Asterisk#2 to Asterisk#3 (3) Asterisk#3 to SIP_Phone3 At (2), that is transferring a call such as -- Executing Dial("SIP/211-6da6", "iax/k0.dyndns.org/302") in new stack -- Calling using options

Hi all, I am trying to connect to a softphone application using an Iax channel on Asterisk 1.4.30. I can do outbound calls, from softphone to asterisk, but not inbound from asterisk to softphone. I get the following Debug: ---------------------------------------------------------------------- ---------------------------------------------------------------------- Tx-Frame Retry[000] -- OSeqno:

Can any one help? Thanks. we have two * servers (Version 1.2.1) and one 1.09 server. Calls between these two 1.2.1 servers have odd behavior. But call from 1.09 to 1.21 server working fine in either situations. See below pls: Local server iax.conf [tosyd] username=mel type=peer secret=xxxx host=198.168.2.66 remote server iax.conf [mel] type=user secret=xxxx host=198.168.2.67

Hello, I have some problems in placing calls through IAX... It does not work :) in the asterisk console I can't see nothing about dialplan enter or so, IAX debbugging seems to be unuseful... this is my configuration: [612] type=friend secret=123456 notransfer=yes disallow=all allow=gsm allow=ulaw allow=alaw context=from-internal host=dynamic requirecalltoken=no I enabled IAX debugging, but

Hi, I'm testing an upgrade from 1.4.18 to 1.4.37 in a VM prior to putting it into production. Ive done this by installing 1.4.18 onto the VM, putting my config files in place and then installing 1.4.37 over the top (which is what I'd have to do on production). I've found a few issues in the config files, but nothing I couldn't handle until... I hit inbound IAX issues. My

Hey all, Any suggestions on how to estimate how much data rsync would synchronize on average in a given installation? Assume that a full rsync has already run and the only data being updated is just the daily diffs. Thanks, noam Noam Birnbaum http://maccentricsolutions.com/ 877.luv.macs x89 ð Apple Certified Technical Coordinator ð Apple Certified Help Desk Specialist --------------

which ir is the * channel? I have decleared 1 peer one in veracruz: [aullox_gdl] type=peer username=aullox_gdl host=200.67.99.127 and aat gdl I have: exten => 200,Dial(IAX/aullox_gdl@200.64.35.58/200@aullox) 200.64.35.58 is my ip 200 is tehe xtension and aull is the context and at veracruz i have: [aullox] exten => 200,1,Wait,2 exten => 200,2,Playback(transfer,skip) ;

Hi people, i have problem with IAX2 between two asterisk PBX. When i try call some number i get "INVAL" packet, but when i try call same number via OpenVPN (is between this two asterisk) call is working fine.So i debug communications and here is my opinion ... Schema of connection: Asterisk1 -> ADSL router with NAT -> INTERNET -> Asterisk2 A)Calling directly via public

Hi, I have two asterisk servers and I want to make these servers call each other as they were internal. I have succeeded in one way. Server B can call Server A without problem, but Server A cannot call Server B. Here's the iax configuration of servers Server A: ================== [ipek] auth=rsa context=from-internal host=XXX.XXX.XXX.XXX inkeys=ipek outkey=odtu peercontext=from-internal

============= SJphone Log ============ Outgoing SIP session Respondent: (sip:8612@192.168.2.2) Remote client: Started: May 26 16:33 Accepted: no Ended: May 26 16:34 End reason: Call rejected: 503 Service Unavailable =============== Asterisk Debug ================ Executing Dial("SIP/2201-a83e", "IAX2/<FWDNUMBER>:@iax2.fwdnet.net/612|60|r") in new stack --

I have something going on that I don't fully understand after a weekend of looking for answers. I have an iax account with Tesco that works flawlessly with the Zoiper client - but is giving me trouble with inbound calls in Asterisk 1.6. After some playing I have ended up with an iax.conf file that looks like this: [general] calltokenoptional = 77.75.0.0/255.255.248.0 maxcallnumbers = 16382