similar to: Snom high SIP ping time

Hi, I'm trying my new Snom 360 phone (6.2 firmware) and I'm seeing that it doesn't register with the Asterisk 1.2.9.1 server after a reboot. I need to click "Re-register" in the web interface. I set: - Support broken Registrar: On - RTP Encryption: Off Any help? -- Domenico Viggiani

I have a client that has 5 Snom 320s. 4 work great, one does not. I upgrade the firmware to the latest (6.5.2) and the problem goes away, but then comes back a couple days later. There is a slight packet loss on the phone (about 1%), though there is no packet loss on any of the other phones. I determine the packet loss by the Linux command "ping -f -c 10000 192.168.2.10". Outgoing

Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. ----- Mike Hammett Intelligent Computing

A client of mine has a Snom 320. Usually when he comes in each morning, it is asking him for a password. A power cycle brings it back to normal operation. How do I troubleshoot this further? --Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070221/0e4d3ab8/attachment.htm

----- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070905/c62f4465/attachment.htm

Hi List, Here is the config ext 2601 is a GS BT-101 phone ext 2062 is a SNOM 200 latest public firmware on both asterisk is Asterisk CVS-11/14/03-22:55:45 Make a call from 2601 -> 2602 life good, call works have 2602 place call on hold. The music on 2601 IS NOT my music on hold. It seems its a MOH server SNOM has. take call off of hold on 2602 and 2601 still trys to play parts

The USB core was completely rewritten in 2.6, and as such the functions that ztdummy depends on do not exist in 2.6. I get the feeling that these changes are too much to easily fix ztdummy, so I don't expect to see it working on 2.6 any time soon (if ever) I made some small changes to zaprtc to work on 2.6 and I have MoH and Meetme functions working fine in my lab. For production I would

I need to setup a small mail server on a local network. It only needs SMTP ability as it's just so Asterisk can send out emails. The machine has sendmail installed. My primary mail server seems to be rejecting the messages. Some research says something isn't configured properly. What do I have to do so the outside world accepts emails from my Asterisk box? It is behind a NAT.

My provider has a Coppercom switch. I have included the authentication information they gave me. How would I structure this in Asterisk to the registration and the entry in sip.conf? User Name - 8159093010 Password - XXXXX No Pin Proxy - sip.essex1.com (10.1.3.2) Outbound Proxy - proxy.essex1.com (63.164.210.14) Change setting to use "outbound Proxy" ---------- Mike Hammett

My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I

I thought I read somewhere that you no longer have to do a special make command for the 2.6 kernel. Is this true, or should I still make linux26? I'm having problems getting anything zaptel to load properly. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Folks, I have posted a couple of message to the list and do see them, even after waitin for long time (2 days). Can someone please point me to the rules for posting to this list? I think it had to do with the subject that I was looking for. I was looking for IAX terminiation service that can handle high volumes. Thanks John. --------------------------------- Yahoo!

I have multiple upstreams in my office. The primary upstream is having some issues with latency\jitter. I want to move the VoIP traffic to another interface. I have the router set to send all traffic destined for "local" networks out the respective interfaces. Traffic destined to the Internet goes out one of the upstreams. I can do this on a per-IP basis and have successfully done

One of my providers has a different SIP account for each number. I have all of my users in one outbound context (caller ID passes fine). How do I ensure that the callers get routed down their correct SIP account with my provider for e911 purposes without each having their own context? ----- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part

In all the information on Asterisk it takes about TDMoE to link asterisk servers together. Is this IAX??? How would I use TDMoE. Maybe my first question should be, What is it??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/cd74bddb/attachment.htm

I have two Asterisk servers. One is my primary server that I link to all of my providers and the other is at an office building with multiple tenants. If I tell Asterisk to dial an entry in the iax.conf that is for one customer off that second box, why does it use a different account for a different customer? It still ends up at the correct box, but it is hard to troubleshoot issues when

I currently have a single server with a few SIP and IAX upstreams for origination and termination with IAX clients. I am adding a second server that will have a much higher capacity and will be handling a larger call volume. However, this second server is not going to be geographically near the first. It will largely share the same upstreams. I would like for this to be an integrated system

Most of the cases can easily be solved by setting the handset mic gain to 2 (out of 1..8). The gain is usually much to high - optimal for whispering voices. If the other side talks loud the echo of the cable will be amplified too much. CS ________________________________ Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Im Auftrag von Mike Hammett

I was looking at the ILECs' web sites to determine how their users access voicemail. I looked at AT&T, Verizon, Qwest, and Embarq. They supported one or a combination of the following for calling from your phone: *98 #55 Toll free number Your number A varying phone number, based on your number's location. Calling from anywhere else, they supported: Hitting star when

Hi, I've been trying to bring our Asterisk server to the latest version. I've grabbed the latest CVS and upon trying to compile zaptel, I get the following errors: gcc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o gendigits.o gendigits.c gcc -o gendigits gendigits.o -lm ./gendigits gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB