search for: sipservic

Hi, Thanks for your response on the kernel switching.I was away and could not reply immediately. Right now, I am facing a differentissue. I have to set up DNS server using BIND on Centos 4.3. When Itype the hostname on Centos, it shows: sipserver.vodcalocal.com But the cli prompt has root at sipserver~ meaning only the sipserver part of the hostname is displayed. whyis this so? What is the

Hi I am a newbie to * and I am having a problem which appears strange as I did not find any mention of it anywhere in my search. Simply speaking, I have an external SIP proxy server which I am trying to configure for incoming and outgoing calls from my asterisk installation. So here is my configuration in sip.conf [general] register =>

Hello, I?ve been using Asterisk 1.2.18 for a while, and today, with no apparent changes, I started receiving these messages: Aug 31 13:26:57 NOTICE[27528]: chan_sip.c:5495 sip_reg_timeout: -- Registration for 'user at sipserver' timed out, trying again (Attempt #19) All trunks and extensions went to: sipserver:5060 user 120 Request Sent 011

Hi, I have register a sip user to sip server. I can see after registration * is sending periodic "SIP Request: OPTIONS" messages to server. but it's not getting back any response that should be SIP 200/OK as the documents say. 3130.299707 192.168.2.113 -> 58.ab.cd.ef SIP Request: OPTIONS sip: sipserver.net 3131.299513 192.168.2.113 -> 58.ab.cd.ef SIP Request: OPTIONS sip:

Does anyone know where i can get DIDs in New Zealand. I am look for area code 06. -James --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004

Hi, Is there a specific latency that asterisk accepts? I encountered a problem wherein when the latency was unusually high,my xlite's (i have 2 xlite) cannot register. but when the link suddenly went stable, the x-lite just registered. what i forgot to look at is if the registration packet is reaching my asterisks. ------ when xlite cannot register --------------- Pinging

I'm trying to connect a Cisco 7942 to my Asterisk box. I have a 7960 and 7912 currently connected and functioning. I'm trying to use the recommendations from here: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP I have created a "XMLDefault.cnf.xml" and it took the latest image but the phone states it's unprovisioned? Any

Kevin wrote: > There is a reboot script posted on the wiki to reboot Polycom > telephones. When I execute this script, I get the following messages. > I am concerned as this is causing issues with asterisk and the PRI. > Does anyone have any ideas why this would be happening? > > > > asterisk console: > > -- Remote UNIX connection > -- Remote UNIX

With my thanks to Brian West and his offering in the thread, "Subject: Re: [Asterisk-Users] Remote reload Cisco 7960" I offer PolyReboot.pl, a perl script for rebooting Polycom IP Phones PolyReboot.pl takes an IP address as a single argument and reboots the phone. You must have a cfg file in the Polycom style, i.e., 00ab00cd00ef.cfg - all lower case. Further, you need to use ftp for

Why is SER considered a better SIPserver than asterisk , why is it that SER can handle more clients than asterisk can. And if this is just cause of say poor SIP handling code in asterisk then is there anything being done to fix it. Just wanted to know why SER claims to be better than asterisk as a SIP server. ? -- regards Vikram (http://www.vicramresearch.com)

Hi there, I am trying to setup a configuration that requires me to use SIP and asterisk behind a firewall and over a VPN to a remote office and with some local Phones also. I can't use IAX to my provider because they don't offer it and my handsets ( snom 300 ) also don't support IAX so it's all SIP. The configuration is a follows Asterisk PBX 10.202.17.217/24 ------>|

Hi All, I have a thin cliente here that i want to run asterisk: - National Semicondudor Geode GX1 266MHz Geode 266MHz single chip - NS Cx5530a Southbridge National Semiconductors SC2200 - NS PC97317 in chipset - 32MB Compact Flash - 64MB Ram - 10/100Mbps, Autosense 10/100Mbps, Autosense Realtek 8139C National DP83815 / DP83816 Some tip? I have a ide>flash

Hi, I think that the web-driven SIP Phone (free) doddle (beta version) can be useful with your Asterisk applications. You can pre-fill it with your sip settings (Asterisk host name or IP /?realm / sip user), you just need to setup the HTML link as that: (Attached is the HTML page example) ? /**************************/ simple HTML code example: /*************************/ <html> <head>

Hi everybody, I have some problems making calls from a sip user (HT286) to the pstn trough Digium Wildcard TE110P, i allways have an error : SIP 403 INVITE sip:0170708959@192.168.1.4;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bK6576a5e11afe28bd From: "test" <sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79 To: <sip:0170708959@192.168.1.4;user=phone>

Hi all! I get this error when trying to start asterisk: ERROR[8192]: File asterisk.c, Line 1349 (main): Unable to connect to remote asterisk What can be the problem? Thank you! Miklos iPFONE Telefonia IP Rua Caio Graco 735 S?o Paulo SP iPBX +55 11 3801-3702 UK +44 870 - 3403539 FWD 64662 sip:ipfone@sipserver.com.br www.ipfone.com.br info@ipfone.com.br -------------- next part

hello all, I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record. Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault. the original version of cdr_mysql.so works fine but I need the start time and end

Can asterisk act as a SIP registrar or location server? I would like to be able for a user agent(client) to register with whatever client they are using as "username@domain-name.com". Rather than the entry/username/password that is setup in the sip.conf file. That way a user could log into any SIP enable client and their calls would follow them around. I have read the sip.conf man pages

Hi. I've just bought SIP telephony service from a Swedish telco. I've managed to make and receive calls with kphone. Now I want to set up asterisk to be able to add fancy features like voice mail and recording conversations. But first I have to get the basic setup right. I'm running asterisk and kphone on the same machine, behind at NAT-router. When I make a call (from my regular

Hi list, I have been looking around for a while now, and cant seem to get to the bottom of my problem. My setup is that I have a separate SIP server that servers my SIP subscribers, and I want to use Asterisk purely for voicemail for now. So I set up a common SIP extension at my SIP server, and made Asterisk register it, so that normal users can forward calls to that common extension, and

I've got my 1.2.x Asterisk server registering with a SIP provider using their servers DNS entry, not their IP address. My server is behind a NAT. The setup works like a champ for weeks then I get a call reporting inbound calls are failing. When my server isn't registered, inbound callers get a "disconnected" message; very bad. When it happens, I've found that the IP