similar to: Notice message meaning for C7960?

When a Snom 200 (v2.1l) calls a C7960 (v4.4), both using g711u as default, the conversation is extremely noisy from the Snom to the Cisco, but clear in the reverse direction. Using a sniffer, I see packets from the Snom to the Cisco of 87 bytes and Cisco to Snom of 214 bytes. Asterisk is CVS from Saturday. The communications between the two was working fine on Saturday, however something has

I have multiple voiceage g729 licenses installed on a RH9 box, and have a remote C7960 configured to use it (low bandwidth). In calls like: Remote C7960 -> g729 -> asterisk -> g711 -> C7960 the audio is oftentimes rather choppy. Changing the remote 7960 to use g711 seems to eliminate/reduce the choppyness. Any ideas on what might be behind this?

Using the following defnitions with a C7960: exten => 3001,1,Dial(SIP/3001,15,r) exten => 3001,2,Voicemail2(u3001) exten => 3001,102,Voicemail2(b3001) exten => 3001,103,Hangup If someone is on this phone (real conversation) and another call comes in, the second call goes through the 15 second timeout and is dropped into the 2-priority as "unavailable" (not the 102 busy as

pstn -> sip gw -> * -> C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates both the sip gw

Someone recently posted the following list as functions built into * *0# sends flash *8# remote call pickup (pickup phone in your group) *67# disable caller id *70# no call waiting *78# do not disturb on *79# do not disturb off *72# enable call forwarding *73# disable call forwarding *82# enable callerid I'm running a CVS from a couple of weeks ago with multiple C7960's, snom 200,

Just received a new Cisco 7960 (not refurb, but brand new) and it won't tftp the initial config file (OS79XX.TXT) from an FC3 box. The 7960 does get an appropriate dhcp response including the tftp address. Using a sniffer, I see the tftp request being sent from the 7960 to the FC3 box, but the FC3 box responds with error code 4 (Illegal TFTP opertion) and an error message of "Request not

Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? I want to stop on the basis of sip.conf account only. bcoz I can't apply iptables rules on server it's remote server of server provider and we used it for making voip call for customers. for the time been i have close all sip accounts. but can't stop for more then

My server is being attached all day and fail2ban is not stopping the attack. I updated stamstamp to match fail2ban requirements. [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 handle_request_register: Registration from '"7002" <sip:7002 at x.x.x.x>' failed for '38.108.40.94' - No matching peer found [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830

Hello All, my asterisk server is constantly under attack [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '"4941" <sip:4941 at public_ip>' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '"4941"

Hi the list, I am using package.skeleton on a file that has a very classic name, but package.skeleton detect it as a invalid name : --- 8< ------- package.skeleton(name="packClassicBis",code_files="./packClassic/R/progClassic.R") Cr?ation des r?pertoires... Cr?ation de DESCRIPTION... Cr?ation de 'Read-and-delete-me'... Recopiage des fichiers de code... Cr?ation

Hi every one, Hello i am doing project on evaluating the sip proxy performances like asterisk, openims and opensips using the traffic generator SIPp. I am using 2 computers of same configuration as SIPp clients one as uac and other as uas... and one laptop for asterisk server...... UAC:192.168.1.99------------------------>Asterisk

Greetings and salutations to all... I'm having a bit of a problem getting a SIP phone (Xten) to call an H323 Cisco ATA-186. Both devices can call into the * and get the demo, voicemail, etc... I'm pretty sure my problem is in my configs as it feels like a stupid error and to prove this to myself I set tcpdump on the * box to capture all UDP traffic going to and from the ATA-186. If I

Hello after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1 none of my realtime SIP peers (saved in MySQL DB) register anymore. [Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from '<sip:testacc77 at 178.19.90.240>' failed for '78.119.140.190:5076' - Wrong password [Aug 15 22:04:13] NOTICE[30098]:

I have a new wellgate 3802 unit. I have not gotten it to register with asterisk 1.2.6. My proxy setting is the correct IP in the 3802. My security config is 1001/1001 and 1002/1002 on the wellgate (simple at this time). My sip.conf has: [wellgate3802L1] type=friend dtmfmode=inband username=1001 secret=1001 host=dynamic canreinvite=yes nat=no context=wellgate [wellgate3802L2] type=friend

Hello, I''m trying to use sqlite3 and I''ve followed the steps here : http://wiki.rubyonrails.org/rails/pages/HowtoUseSQLite However I get the error above (uninitialized constant API) when trying to run a migration. I''m under Windows XP, using cygwin, the sqlite3.dll file is on the lib directory of my rails app (filemon, an utility that reports file access attempts

Has anyone played around with QoS or TOS relative to * and sip phones? I was just doing a little real-time research and noticed our C7960's mark IP packets with "low delay" and "high throughput" (presumably due to tos_media: 5 in the SIPDefault config file), and rtp packets flowing "from" asterisk back to the sip phone are not marked at all. Is there a * config

data/Makefile.am | 8 ++--- player/swfdec-player.ui | 22 +++++++------- player/swfdec-window.c | 2 - po/POTFILES.in | 1 po/POTFILES.skip | 1 po/fr.po | 72 ++++++++++++++++++++++++++++++++---------------- thumbnailer/Makefile.am | 20 +++++-------- 7 files changed, 74 insertions(+), 52 deletions(-) New commits: commit

I have a Polycom Soundpoint IP 500 that I have been using with Asterisk for a few weeks. It has been working OK, no major problems other than a freeze up every now and then, until today. The power apparently went out last night and for some reason the phone appears to be working but I keep getting the following errors repeating over and over in my Asterisk log file (IP's X'ed out): Aug

I would appreciate any tips/clues/advice very much. I have started using ruby on rails and when the MS SQL Server database is on a different box than my dev box (where the RoR app is located) I can connect fine. However when I try to connect to a MS SQL Server database on the same box as my RoR app I get an error. I have tried to use the IP address of the dev box as the value of

We have IAX2 peer between two asterisk and I am getting following error following IAX2 WARNING. IAX calling is functional [May 10 15:23:34] WARNING[2056]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 11, ts=3030332, seqno=211) [May 10 15:23:44] WARNING[2047]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded