similar to: SIP behind NAT: NAT'ted end has to talk first?

Does any one regularly reboot GS101? It sometimes lost registration with * and needs to be reboot. What is the best way to do it by cron? David Kwok

Just got GS 101 phone and plugged into the network. Got ip setup however, the following problems arise: 1. when dialing an extension, I cannot further send any key tone to Asterisk. 2. there is no sound coming from the other end. I have a sip.conf setup for GS: [General] disallow=all allow=ulaw allow=alaw [gs] canreinvite=no dtmfmode=info In the GS101 setting rtp port = 5004 sip port = 5060

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,

I wonder if someone out there could loan me a peek at their sip.conf? I have conflicting advice, for instance, about whether or not to use "nat=1" and also whether or not the ATA should be registering with the instance of asterisk it is going to be using to dial out. Thanks in advance. B.

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?

Just for the record and to possibly help with others who get BudgeTone phones. My asterisk box is behind NAT, and I use Vonage, NuFone, and iconnecthere for my "POTS backhaul." On the front end I have an ATA186, a Digium TDM20, and now a BudgeTone 102. The BudgeTone definitely has issues wrt the RTP stream and NATting, although unfortunately I haven't yet been able to dig

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

Hi, I have recently loaded Openssh 3.4p1 on an Tru64 Unix 5.1A system. I followed the installation instructions described in INSTALL, essentially using all default settings, and it went throught without any obvious errors. I can then use the root account to initiate outbound and inbound ssh calls, and can log on without any problems. The trouble is that when I try to use ssh to log in (from a

Hi All, Outbound Iconnecthere calls work without any problem but Inbound calls are very intermittent. It seemed to work for a week or so but over the past week 99% of inbound calls are dropped to ICH voicemail. Would moving the Asterisk box to a public IP resolve the problem or is it just an ICH/Asterisk problem? I am registering against natrelay.deltathree.com. asterisk -vvvc shows an

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

(Apologies to those Toronto Asterisk Users' Group folks who have seen this message... I figured I'd have more success with a wider audience) I'm trying to boot a Cisco 7960 from an ISC DHCPD server (3.0.3 on FreeBSD 4.11), so far unsuccessful, and getting some odd behaviour on the wire. I wonder if anyone has done this before and therefore can validate whether or not the traffic I am

I've been to the WIKI and I've searched the archives. Is anyone on the list successfully using iconnecthere behind NAT? I was, for over a year, and then I changed my "plan" with them. Now all my calls get intercepted immediately, "We're sorry, but your account is temporarily unavailable." Incoming calls work just fine. I contacted their so-called

Two asterisk boxes 150 miles apart, both cvs-head as of this morning (and since Sept 27th), connected via iax2 with low-utilized ds3 internet, C7960 calls exten on remote system (also C7960), and call goes to VM. No other calls in either system (eg, no load). Both boxes have iax config'ed as: trunk=yes allow=ilbc jitterbuffer=yes Recorded VM messages are very distorted. Changing only

Have a strange problem... When a C7960 calls the Polycom ip600, the ip600's first line button blinks, the ip600 display shows the proper callerid, but the phone does not ring at all. If I call the same ip600 from a bt102, the ip600 rings properly. If I call the same ip600 from another C7960, the ip600 rings properly. All phones and asterisk are on the same lan within a few feet. The

Hi, I know this is slightly off topic but I figured the knowlege here is probably the best on the subject.. I want to setup remote offices with 4 to 6 SIP phones (SNOM 200) using ADSL and the internet to connect to the Asterisk box.. These phone will be behind an ADSL router using NAT... I don't want to setup another Asterisk system in each office so IAX is not an option.. I could use

We have started to upgrade or Unix servers from Samba 2.0.7 to Samba 2.2.5. I did one machine as test and all worked fine. Today I tried adding two more today and both come up with the "Incorrect password or unknown username" box in Windows Explorer. All the Unix servers are running Solaris 7. I did the make on an NFS file system once and then ran 'make install' on each

Some of you have been following our progress on http://farfon.convergence.com.pk as we blundered our way through the development of a low-cost ethernet IP phone that does IAX and augments the client options currently available for the kick-assterisk server. With help from the denizens of #asterisk and kind words of advice from Mr. Spencer and the rest of the gang ... we're proud to have

Hi folks, As part of our build, the Tock project uses remap-path-prefix [1] to create a reproducible build. This means that the paths inside of built artifacts are not full source paths. When we later attempt to produce a listings file, the source mapping fails. The result is many copies of this recently merged warning [2]: llvm-objdump: warning:

Dear Linux Gurus I'm having problems with KVM and networking. My guest cannot use NAT through the host's connection. This is what I've done: I installed a new version of Centos 6.5 on the hardware. Starting with a Net-Install, I selected the Virtual Hosting, and later added "Desktop". I ran "yum update" with some reboots until nothing needed updating. The

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