similar to: Can't call my extension

Dear Dovid, thx for the input. for timer in sip.conf, I used default setting. This is some of the result for "sip show settings" RTP Keepalive: 0 (Disabled) RTP Timeout: 0 (Disabled) RTP Hold Timeout: 0 (Disabled) Session Timers: Accept Session Refresher: uas Session Expires: 1800 secs Session Min-SE: 90 secs Timer

Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? Thank you! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY

Hi All. got this problem that IP phones could not re-register to my server. even if device is power cycled it still would not register. the solution i found was to change the SIP port settings on the phone and it will register. but after registration expires and its time to re-register the same thing will happen, so i have to update the port settings again just to make it work which is

Hi list, I've been googling this issue and found some good resources however I am still running into problems with the following combo ... Here's my story; - Asterisk 13.4 with FreePBX 12. - Migrating from Asterisk 11 / FreePBX 2.11 - Mix of Cisco 79xx handsets, mostly 7940G's. My problems started with (the very common) issue of the 7940 not replying to 401

I?ve gotten to the bottom of this; Seems that the pjsip.endpoint_custom.conf isn?t getting included properly, or my syntax is wrong. If I put force_rport=no into pjsip.endpoint.conf and reload only Asterisk, everything works perfectly. Unfortunately, I?m using FreePBX, so it owns this file and my changes won?t persist a FreePBX reload. Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F

I want to capture all SIP messages. I have about 30 hosts in about 6 colos. My first thought was dumpcap, but the output file name format bugs me. What do you use for long term SIP capture? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

For someone that's network-aware, but hasn't sat down and plowed through umpteen SIP-related RFC's and memorized the standards, is there a good primer on troubleshooting SIP issues? I'm seeing a lot of NOTIFY/603 messages on my network between Asterisk and my Sipura 942's, for instance... Not sure what these are... perhaps the qualify keepalives? In which case, I guess

I am quite surprised about the degree of surprise in the group. A few days ago, somebody called a school and issued a threat, through my network. The call came from China, but of course it was US caller. The DA wants to know where call came from. The caller ID is "Restricted" and the chinese carrier is playing games. If I had a way to store the media IP, I would be able to pinpoint the

I have some troublesome numbers that I would like to capture the SIP dialogue when I am calling them. When I am about to dial the number, is there any way to turn on SIP debugging in the dial plan before I make the call? (and turn it off after the call is completed?) -------------- next part -------------- An HTML attachment was scrubbed... URL:

**OFF TOPIC** but perhaps of interest to some on this list. I apologize in advance to those who may be offended. The byline: ******************************** "ChatGPT's odds of getting code questions correct are worse than a coin flip But its suggestions are so annoyingly plausible" ************************************* from here:

I have an account with FWD and I have configured my SIP.conf with [fwd] type=friend secret=password username=901835 host=fwd.pulver.com But when I am trying to dial out my own DID , I dont see any call landing in asterisk. In extension.conf (vicidial) file I have exten => 2062036895 ,1,Ringing() exten => 2062036895 ,2,Wait(1) exten => 2062036895 ,3,Answer() exten => 2062036895

I am trying to configure mysql to use a different datadir than default in order to move this to a larger volume. I have copied all mysql data from /var/lib/mysql to my new volume and ran both chown -R mysql:mysql * and chmod -R 660 * in order to setup correct ownership and rights for the data. It was working for a few days until today upon going into mysql and typing show databases, I receive

Hi, I am relatively new to Asterisk and am looking for help in configuring an IP based phone. This phone is not on the same subnet as the PBX. I read that there could be an issue with NAT so I am bypassing this by connecting temporarily with an Internet IP. So far, I have configured the server extension with the phone MAC address along with configuring the phone unit with IP, Gateway and SIP

I''m using a legacy table, where the unique id is not ''id'' I have a Model class like the following: class Article < ActiveRecord::Base set_primary_key "ARTICLE_ID" end however, using a link_to like the following (modified scaffolding), the link has no id value: <% for article in @articles %> <tr> <% for column in

Hi there I am wondering if anybody can help me illuminate a problem I am having with my asterisk installation. I am using: - IP phone (Siemens gigaset S685IP) behind a modem/router that has ports udp 5060 and 10000:10100 forwarded to the static ip of the IP phone (192.168.0.3). This has to go to: - modem that operates in half bridge mode (no nat) to a linux firewall (does natting ip is

Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x

ngrep is a great network packet capture. will it be included in epel? -- Peng Yong

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

While trying to debug a goofy XML loading issue in IE, I''ve found that Mongrel (latest) returns Content-Type: 0 with every request on a particular (CentOS 4) server, yet not on my local (OS X) box. These both access identical Rails apps. This seems like a clue, but thought I''d ask here if for some reason this is expected behavior. Both running Ruby 1.8.4. Both return

Good idea, El - thanks. The link is https://docs.google.com/document/d/1QwuaWZk6tYlWQXJ3WLczxC8Cda6zVERk/edit?usp=sharing&ouid=103065135255080058813&rtpof=true&sd=true This is helpful. From the article, which is typical of Lexis+ output, I want to extract the following fields and append to a Calc/ Excel spreadsheet. Given the volume of articles I have to work through, if this