similar to: Cisco IOS SIP mime 1.0

A little off-topic: I have the following hardware: 2610 XM NM-2V VIC-2BRI NT/TE IOS loaded: flash:c2600-ipvoice-mz.123-5d.bin" I get the following error while booting: %C542-1-UNKNOWN_VIC: VNM(1), vic daughter card has an unknown id of FF Is the VIC-2BRI compatible with the 2610XM? What IOS needs to be loaded? http://www.cisco.com/en/US/products/hw/modules/ps2641/products_tech_note0918

Does Asterisk 1.0.2 support 302 redirects? With 1.0.1 I get: Got SIP response 302 "Moved Temporarily" When forwarding the call to other SIP server. This is a "bug": http://lists.digium.com/pipermail/asterisk-users/2004-May/045774.html --- Jan Baggen - jbaggen@ip2.nl IP2 Internet BV / http://www.ip2.nl

2 new Cisco 7960 phones are requesting a CTLSEP file, seems like I triggered the universal application loader. I want to load the sip image 7.2 According to this Cisco information: http://www.cisco.com/en/US/customer/products/sw/voicesw/ps4967/products_upgr ade_guides09186a008022a968.html#wp1047292 If the CTLSEP MAC file is not present or is empty, the phone proceeds in nonsecure mode with the

Asterisk is an excellent PBX system, and makes a very good endpoint in the SIP chain for all sorts of things -- IVR systems, voicemail applications, automated messages, etc. It has an extremely well-written CDR engine, so many people mesh it with billing applications to produce accurate accounting information. It also is fully aware of the media stream, which means it's capable of cutting

This for the archives in case it may help someone: I was able to upgrade two Cisco 7940's from firmware P0030301MFG2 to SIP 7.1 as follows: 1. Installed 7.1 images from the Cisco zip file to the TFTP server. 2. Specified "image_version: P0S3-07-1-00" in SIP<MAC>.cnf and SIPDefault.cnf 3. For the older of the two phones, renamed P003-07-1-00.bin to P0S3-07-.bin, making it

Compiled Asterisk from FreeBSD port (0.9.0_2) When I start asterisk it uses 100% cpu. Searches on Google say to comment the noload => chan_oss.so in modules.conf But this is already commented. Make.conf contains some optimizations. modules.conf: ; Asterisk configuration file ; ; Module Loader configuration file ; [modules] autoload=yes ; ; If you want, load the GTK console right away. ;

Anyone see an issue on asterisk 1.2 that it will not accept the invite from a Cisco gateway. If I turn off voice service voip signaling forward unconditional then Asterisk accepts the call but without cname. Below is a trace. Any help is appreciated. Thanks John Bittner Simlab.net voippbx01*CLI> <-- SIP read from 216.86.35.24:63549: INVITE sip:9734333001 at 69.60.198.130:5060 SIP/2.0

I'm new to Asterisk, but, Managed to get it working for outound calls from my ATA --> Asterisk --> Cisco 2620 using SIP. However, I'm having problems with Inbound calls from the Cisco.. Cisco 2620 --> Asterisk --> ATA .. In fact, voice mail won't even work.. This is a snippet of what I'm getting when I try to call the ATA -- Executing

Hi, I am sorry if this message has been reposted, but for some reason I am having problems with posting it. I configured asterisk and zaptel modules with fedora2. I want to be able to load the zaptel wcfxo and wcfxs modules. For now I will use only the Wildcard TDM400P card. I am able to load the modules but I cant configure them using ztcfg or zttool because the tools are compiled to use the

Does anybody have any experience with the ZyXEL Prestige 2000W? I am having problems with the line tear down when I call another extension. If nobody picks up at the other end when I hangup the 2000W, the other extension continues to ring. Is there any way to hangup a SIP call if there is no more traffic? Asterisk seems to think that there is still a connection open. This is pretty annoying

Hi all I have just got a P2000w and experience several problems. Hopefully there is someone out there that has got it working. I saw it on Cebit and the person demonstrating it there told me that it was connected to an Asterisk server on the stand -so it should work. Problem 1: it does not register correctly It get lots of messages like this: Jun 26 19:45:19 NOTICE[1107585968]: chan_sip.c:5630

I can not register via sip to iptel or sipgate and do not see sip into ethereal. I do not unterstand why thats Wudu .. but i am new to asterisk and sip. I am behind a susefirewall2 but asterisk even do not register if it is down. The asterisk is running onto the machine witch is connected to the internet. No answer seems coming back from iptel (sip debug in asterisk). Ports are open (5060,

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten =>

I am trying to make an outgoing call using an iptel account using Asterisk. I have followed a how-to for asterisk and iptel found at http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20client%20SER I am getting the following error message: Got SIP response 403 "That is ugly -- use From=id next time (OB)" back from 195.37.77.101 I'm not quite sure what that means. Does

Hello. I am in a strange situation. I have two asterisk. Asterisk "A" makes a call for asterisk "B" by IAX. Asterisk "B" recives the call and delivers it to Openser by SIP. The problem is openser printing this in the screen: ERROR: parse_to : unexpected char ["] in status 5: <<"David" <sip:>> . ERROR:parse_from_header: bad from header

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten =>

Hi! Then I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer the phone I am ringing. It works fine if I call the 2000W from other phones. I have tried many sip settings. I use this now: [205] type=friend username=205 secret=passwd205 callerid="Zyxel" <205> host=dynamic context=local nat=yes canreinvite=no disallow=all allow=g729

If your phones support PoE, I have had huge success with Zyxel: http://www.amazon.com/ZyXEL-ES1100-16P-16-Port-Ethernet-Unmanaged/dp/B00 5GRETMM/ref=sr_1_3?ie=UTF8&qid=1426296572&sr=8-3&keywords=zyxel+poe If you want to go even cheaper, I have successfully used these as well: http://www.amazon.com/TRENDnet-8-Port-100Mbps-Switch-TPE-S44/dp/B000QYEN

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Anyone have any experience with either of these, I 'd appreciate some feedback? Plus it seems pretty easy to steal a connection with this. Zyxel Prestige 2000W WiSIP thanks, - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin

Hi, Does anyone have suggestions hooking Zyxel Prestige 2002 to Asterisk? I have tested Zyxel Prestige with both supported codecs. Call with G.711 sounds very choppy and cracking. Almost can't understand a word. Today I installed G.729 support into Asterisk but unbearable voice quality remains. It's a little bit better though. I have tested that Zyxel ATA with some commercial SIP