similar to: Asterisk with Draytek 2600V

well, now that I have moved to dovecot, I'm not getting any spam... in fact, I'm not getting e-mail at all. default_mail_env =mbox:~/IMap2/:INBOX=/var/spool/mail/%u when I first tried to access my inbox, I was told I had permission problems: Aug 14 17:47:22 harvee imap(esj): open(/var/spool/mail/esj.lock) failed: Permission denied Aug 14 17:47:22 harvee imap(esj): file_lock_dotlock()

I have recently installed a test Asterisk server but am having problems getting this to work with a Vigor 2600V. I have no issues using X-Lite to connect to Asterisk but if I configure the 2600V, it registers correctly but I don't get any sound at all although calls do seem to connect. The 2600V works fine with FWD and other services so I suspect I am missing a setting but as a user have

figures. Converted over to maildir today. Procmail is a piece of cake once I managed to pride out of its cold dead fingers the essential tiny bit of documentation I needed. But I'm having a problem with my nested directories. I have a directory called incoming sort my incoming mail. The conversion utility created a bunch of files called incoming.<stuff> and 1 file is visible

I'm in conversation with Draytek's pre-sales dept.............. Here's the most recent reply: <Hello, We really don't know of anyone who has run an Asterisk server on a Vigor2900. There are doubtless people around, but it's relatively rare. Most people don't run SIP servers. Regards,> All I want to know is, if I buy one of these routers, will it break my setup or

Red Hat 8.0 stock dovecot 99.10 for a variety of reasons, I decided to cut over to dovecot this morning. I extracted all of my mailboxes from mbx purgatory back to mbox purgatory, set up dovecot and proceed to get authentication failures. (am using simple password based authentication either direct or through pam) I turned authentication verbosity on and got: Aug 14 11:51:38 harvee dovecot:

I'm trying to debug a NAT issue and I can't make up my mind if the problem is with my Vigor 2800 or Asterisk 1.6.2. I know the Draytek is alleged to suffer from nat 'issues' but I did not have the issue with 1.6.1 - so I'm wondering if something has changed? The Draytek offers 'NAT & Routed' on a single device - so my Asterisk sits on a Public IP, and I have a

One of our clients has a Draytek Vigor 2920- their natted Snom phones behind it are registered to an Asterisk 1.4 server on an external public IP. I've set QOS, bandwidth management and turned off the SIP ALG via telnet but I'm still having some problems with some of the phones losing registration if Asterisk is restarted. I can see the phones sending SIP REGISTER messages, but they

Dear all, I have a very strange problem : * external calls work perfectly, * internal calls between some phones too, * but internal call between two similar phones don't work !!! (Snom 710) When we have sound, there are no errors in asterisk. When we do not have sound, there is the following error : * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module

Snom default configuration is SRTP enabled. You should disable the SRTP from the phone web GUI configuration Sincerely, Sam Basan From: Mitul Limbani [mailto:mitul at enterux.in] Sent: Thursday, November 12, 2015 5:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] No sound with internal

I finally got the power supply for my 7960 and am having problems getting it working. What should be in sip.conf and the SIP(macaddr).cnf file? This is what I have in SIP0002FD3BA8F7.cnf # SIP Configuration Generic File # Line 1 appearance line1_name: Asterisk Test # Line 1 Registration Authentication line1_authname: "phone1" # Line 1 Registration Password line1_password:

Hi, I have been trying to get my asterisk box to send SMS's to my Panasonic dect phone via a Linksys pap2. I believe I have the message centers setup correctly between * and the phone. The pap2 is configured to only use G711a. The Asterisk version is 1.0.7. In my /etc/asterisk/extensions.conf I have [smsphone] exten = 199,1,Goto(smsmorx,${CALLERIDNUM},1) [smsmorx] exten =

Hello, if you have a Draytek SIP phone, please check if the phone doesn't flood your server with SIP REGISTER messages. Draytek phones are broken and keep sending REGISTER messages after receiving 200 OK (even if expires value is long enough). Several such phones are flooding iptel.org public servers these days. If you have direct contact to Draytek developers, please send it to me.

Hi, I installed Asterisk an all works fine exept for Grandstream. When I call with a softphone (ex X-ten) to a Grandstream (BudgetTone-100), I can make a conversation. = ok When I call to a softphone with a Grandstream I can pich up the call with the softphone but the Grandstream keeps ringing like on the other site you didn't pick up the phone.(even if you do so) It's the same when I

Hi! I want to tell asterisk to simply pass-through any codecs that my phones support. I have to use codecs that are not popular and implemented by a third-party, asterisk has nothing to do with them. I've made a test with g722 (that asterisk doesn't support), i've set all my two snom 300 phones to support only g722 and asterisk declined the sip invitation. That is bad for me. Is it

I have been trying to read everything I can find on Sipura 2000 and Asterisk. I am trying to make the Sipura-2000 act as two analog lines off my asterisk. I have followed (what I believe) the example on http://www.voxilla.com/Article39.phtml and I still can't get my Sipura to register with my Asterisk server. I can re-config my Sipura to talk to fwd, or voice-pulse connect and it works

Hi, I've have followed through the help docs in trying to get an initial setup going with two phones and the asterisk server. Firstly, all I'm trying to do is get the two phones actually talking to one another VIA asterisk.. I've added this to sip.conf: [phone1] type=friend host=dynamic defaultip=192.168.1.106 ;username=blah ;secret=blah dtmfmode=rfc2833 ; Choices are inband,

Dear, When I start asterisk -vvvvvvgrc and I ask 'sip show peers', I don't get the ip adress in the 'Host" field. Name = phone1 and phone2 Host=unspecified mask 255.255.255.255 port = 0 status = unmonitored I can ping the two phone's and get a reply (also from the laptop) phone ip adres 192.168.10.12 and 192.168.10.13 (server 192.168.10.11and laptop 192.168.10.14)

Hello all, Here my architecture : Proxy1-asterisk1-proxy2-phone1 If a call arrived from proxy1 to phone1 AND phone1 always forward to proxy, asterisk1 say: -- Now forwarding SIP/phone1-0000001d to 'Local/969990349 at proxy2' (thanks to SIP/proxy2-0000001e) Why it use Local ? I just need to use as a normal call, not a local Thank you Francois -------------- next

I'm testing two options from dial command and can not make them to work. L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. The following special variables are optional for limit calls: (pasted from app_dial.c) w: Allow the called user to start recording

Hi, Actually on my setup all outgoing calls are going trhu a SIP unique account A have a second SIP account with another operator and I would like my setup to use alternatively each of the two accoutns Call 1=> Dial SIP/phone1 Call 2=> Dial SIP/phone2 Call 3=> Dial SIP/phone1 <...> If you have an sample please let me know