similar to: Got SIP response 400 "Bad Request" ; Cisco 7940 inbound station/station call problem.

Hi, I use * release 1.0.9 with differents phones and softphone, i've got a problem with my Cisco 7940G (last SIP Firmware). Sometimes, when i but a call on hold, the caller has got the music, but when i "resume" the call, then the caller does not hear me (and nothing at all)... I must wait for 10, 20, sometimes 60 seconds before he could hear me again. Any body already had

as far as I know there isn't. I use 80 bytes for G711U that may or may not fix your issue. You can also do a ethereal trace to find out what the actual error is. ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com

We've been experiencing an odd issue lately. I'm not sure when it started because it's not happening on most calls--it seems confined to a couple of our queues. It's consistent though. Here's the CLI output: -- Got SIP response 400 "Bad Request" back from 192.168.249.94 -- SIP/502-9a58 is circuit-busy I've tried a few different Asterisk versions

Hi, I have an Asterisk 1.8.2.3 installed (public IP) with a peer (Polycom IP601) installed behind NAT. When the peer makes a call, it's working without any problem. But when a call is coming back, it ends up with a Got SIP response 400 "Bad Request" back from xx.xx.xx.xx where the xx.xx.xx.xx is the public IP of the peer. And the call drops to the voicemail (congestion at peer

Hi Siggi/Jan, >If so, there's still a load version conflict (although I've >never seen a >7960 or 7940 care about the version communicated through SCCP): > >On the phone, press "Settings", then 4 for load information. >watch out for the "App-Load-ID". On my 7940, this is >"P00305000300". Yours >is most likely a smaller number... >

Hi again, I found chan_skinny and that seems to work pretty good. the SCCP one filled out all the buttons really nice, but skinny seems to be working. How do I fill out the second line button on the phone with skinny.conf? Thanks much! ...Jeff

I've just set up my dial plan to use hints for each of my extensions, and I've set up a buddy watch list on my Polycom IP-501's, however, I keep getting the following messages every 5 or so minutes: -- Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.1.106 -- Incoming call: Got SIP response 500 "Internal Server Error" back from

Hi All, I've two asterisk servers on the same LAN, both 1.4, and I keep getting "Got SIP response 489 "Bad event" back from 192.168.3.10" No idea whats causing it. The only references I can find mentions NATing issues, but these are on the same LAN so NAT shouldn't be an issue. 3.10 does authenticate into the server logging the error. The error appears in the log

Hello: New to asterisk and hoping to use for http://summitcamp.org research station. While trying to use with Inphonex I find that incoming calls drop after about one minute-- -- Got SIP response 420 "Bad Extension" back from 208.239.76.169 == Spawn extension (incoming-inphonex, 210, 1) exited non-zero on 'SIP/inphonex-095bf208' Found that I can use `*CLI> sip

Do somebody using CISCO AS5350 with Asterisk? Which protocol do you using: H323, MGCP, SIP? This direction: [12sp->Asterisk->h323->as5350->isdnPSTN] is ok But reverse: [isdnPSTN->as5350->h323->Asterisk->12sp] cannot hear 12sp, but 12sp hear PSTN (codec g711u) -------------- next part -------------- An HTML attachment was scrubbed... URL:

This may be OT, but I can't seem to find how to do this. I have 7940/7960's with Skinny on them. When you start pressing numbers on the dialpad, you start building a number to dial. When I install SIP, that functionality goes away. You have to hit the speaker button, or lift the handset before you can start dialing. Is there a setting I am missing, or is this just a product of

Hi All, Simple scenario: 7940 SIP><NAT Router><INTERNET><Asterisk SIP B2BUA w/Public IP Inbound/outbound calls work fine 2 way audio, features ok, no issues that I can tell so far. 7940 SIP Using Outbound SIP Proxy><NAT Router><INTERNET><Asterisk SIP w/Public IP Phone registers, call in/out SIP Signaling traversing the proxy ok no audio on phone, SDP

can anyone give me a reference to the retrieval of voicemail from the Asterisk PBX using a cisco 7940 phine running sip image. i have configured a single voicemail box using the script, the corresponding entry in voicemail.conf and configured the extension to use the voicemail box . i can see from the asterisk console the message being passed to the voice mailbox, and correspondingly the sip

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey all, I'm having problems with calls dropping after 15 - 20 seconds from a particular provider. The are using a NexTone gateway. Here are the details: Successful call: INVITE cseq 1 From NexTone 100 Trying cseq 1 From Asterisk 100 Trying cseq 1 From Asterisk 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone INVITE (G711U)

I have a problem where SIP calls from some providers are dropping at 15 minutes. The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS sends calls to an Asterisk server. Below, 'Client' is the IP address of the client's host (running FPBX-2.8.1(1.8.20.0) 'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls 'Asterisk' is the IP

Hi all, I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones. Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK SIP to SIP 4XXX -

What headsets have people found work well with the Cisco 7940 and 7960 phones? To date, I have tried a couple of the headsets within the Plantronics H series (H41-N), and noticed that the volume of my speaking is lower over the headset than on the regular handset. I am currently looking for headsets that are known to work well. I do know that Cisco lists the H-91 and H-101 as certified to

Hello, This is not exactly an Asterisk question, but I was encouraged to seek advice here anyway. The kindness of the * open source community is legendary :) I am getting going with an Asterisk 1.2 box, and I'm having trouble getting good quality transmit sound using handsets with VoIP phones. I'm primarily trying to focus on SPA-941, but also experimenting with Aastra 9113i and Uniden

I'm attempting to access the Pigeonhole documentation but am receiving a 400 Bad Request response from http://pigeonhole.dovecot.org/ . Is this expected? If so, has the documentation moved? Thanks for any help! -Ben

Hi everyone I have successfully compiled and installed OH323 support (finally) into my Asterisk. I want to connect the Asterisk server to our Alcatel OmniPCX Office (OXO) PABX, which has an internal H.323 gateway. I have created the correct dialplans in Asterisk and same in OXO. The OXO only supports G711a G711u G729 and G723.1 codecs. When I call from a SIP phone to OXO using my