similar to: Registration failed problems/Polycom 500/maybe nat problem?

Hi people, My asterisk wont register with any sip providers, I have tried three different but they all end up with: Sep 26 17:36:36 NOTICE[114696]: chan_sip.c:4035 sip_reg_timeout: Registration for 'whatever@provider.tld' timed out, trying again There is no firewall and my server has a public IP. Could this be a Asterisk problem? -Fredrik vK

I am so happy about learning how to read in multiple Excel files, that I have to try and make another improvement. I know what I have been doing is clumsy, but it works. Hopefully, someone can suggest a more elegant solution. As a novice, I have been using MS-Word and mail merge to write my code. I start with about 2 pages of code, and end up with 2,220 merged pages that I copy and paste into R.

Our users are experiencing some unacceptable delay when trying to have a conversation. The delay is so noticeable that they keep stepping on each others words and resort to calling the customers via cell phone. Here is the setup SDSL Connection (PPPoA) Speedtouch 610 SDSL Modem 3Com 2224PWR Plus Switch (phones on separate VLAN) 8 Cisco 796 Phones All connecting to a remote Asterisk Server. We

Hi Tinc group, I have recently got an adsl connection at home, which only provides a single static ip address. I want to set up a tinc vpn from home to my workplace. I am trying to decide what hardware I need to do the above. At the moment, I am aware that I could use an Alcatel Speedtouch USB modem directly connected to a Linux vpn box at my home. ( Wall socket---adsl cable---Speedtouch---usb

>>eth1 & eth2 are both connected to the EN5861 router. I''ve done that >>because I couldn''t work out any other way to use the static IP address >>that out ISP have given us. >> > >Aha, this is a config question. You can just add as many ip addresses to each physical card as you like. I forget the exact syntax, but check man pages for

>> No I am using the EN5861, just that we have two ADSL lines and only >> one >> EN5861 and rather than buying another I''d like to use linux to do the >> same job as the en5861, if you know what I mean. > >You can, but since you have two ADSL lines, you''ll need two ADSL devices of some kind, whether it''s a pair of 5861s, or a 5861 and an

Hello, I'm using Asterisk 1.6.2.0 and I like to use escape characters with SendText, because I can just delete the message from my phone (Thomson Speedtouch ST2030) display by sending a return-char (\n). But \n is not escaped: I tried already: exten => 222, n, SendText(\n) exten => 222, n, SendText("\n") exten => 222, n, SendText('\n') exten => 222, n,

Hi, I'm looking for opinions on the "best value" router to use for home offices. It should work for a scenario in which there are 3 computers and 2 SIP phones, handling QoS so that the phones always have higher priority traffic than the PCs. (and not rely on the phones to do the QoS because some PCs may not be connected to the phones). QoS could be based on destination and source

Does anybody has experience with the SIP phone of Alcatel the ST280. I can't make a call with this phone. Everytime I make a call I get the error Jun 12 19:38:38 WARNING[1133718080]: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call 5624@81.124.191.47 for seqno 1 (Response)

Refering to pppoe i have next problem : I asked my isp if i can buy a class of real ip`s to be routed by them. They said elegantly it can`t be done . I want opinions . I am using an ADSL connection through a Speedtouch 510 configured in bridge. About Bgp : i asked someone if i can peer 2 different locations on 2 different ip`s using private asn number and he said yes , and what i don`t

We have newly installed ADSL/speedTouch 510 modem. After the installation, we are getting problems with out Win2K machines. On out NT workstation everything is OK, but on the Win2K we are getting following error, "The semaphore timeout period has expired" whne trying to connect to the domain (GARAGET), My Network Places Entire network MS Windows Network Garaget (our

I''m getting the following error/warning at some point in my config script, and I''m not sure which class it is referring to. htb: class 10007 isn''t work conserving ?! I [think I] understand that htb is a non-work-conserving qdisc, and I [think I] have configured things so that every htb qdisc I instantiate limits the bandwidth, so I don''t understand why this

Ok here's the deal. I set up Icecast and Oddcast, got them to communicate properly, but cannot access the live audio remotely. On other computers in the LAN it works perfetly. Currently I operate wirelessly behind a D-Link 514 router (a mite outdated by now, yes?) and have opened all necessary ports, turned my firewalls off, set this computer as a DMZ, pretty much everything I can think of -

Hi All, Our internal users need to connect to the bank to do internet banking. The bank requires that the client computers connect to ports 15000-15010. In Shorewall how do I do a masq for anyone trying to connect to these ports? Thanks Raymond

Hi i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling client so i can make VOIP calls thru that phone. Aslo that Phone easly able to register with Asterisk Pbx to recive inter-office calls. i try to search from web & also from Nokia site but they only mention this features as "VOIP call from wifi" they mentioed only this much info. they not mentioed info about

Dear list, Shorewall is running here with 2 ISP''s: ISP1: corporate ADSL-line with fixed set of IP''s ISP2: fast consumer-grade cable-connection with higher bandwidth All our main traffic (web, e-mail) is routed trough ISP1. Only for special purposes (frequent large ftp-transfers) ISP2 is used, configured trough tcrules. ISP2 is not so reliable as ISP1 (duh) and they sometimes

I have an asterisk 1.2.9.1 machine with zaptel 1.2.6 running. On the TDM400P, I have 1 FXS port and 3 FXO ports. dmesg reveals: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.6 Echo Canceller: KB1 PCI: Found IRQ 10 for device 01:01.0 PCI: Sharing IRQ 10 with 01:05.0 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1:

hello all, now, i want to do configuration to make sip client have extension on my asterisk.but i have a problem with registration of sip client. *CLI> May 31 13:58:01 WARNING[4927]: chan_sip.c:886 retrans_pkt: Maximum retries exceeded on call 4b5cbb235a46d6ee0bcd278c1e294105@192.168.8.125 for seqno 115 (Critical Request) May 31 13:58:15 NOTICE[4927]: chan_sip.c:4585 sip_reg_timeout: --

Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is

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