similar to: Error 488 Not Acceptable Here

Hello out there, I'm running an Asterisk 1.8.15-cert1 with DAHDI. Today I noticed that Asterisk is signalling to the calling party the current internal CallerID whenever I put a call to another internal phone. Example: Customer calls 020212345-555 -> IVR answers and puts caller to the chosen queue -> Someone picks up the phone (Internal ext. 321) -> CallerID shown on customers

Hi, I'm using http://www.portunity.net/ I configured now asterisk with the following setup: iax.conf: register => XXXXXXX:YYYYYYY@iax.iaxport.de [portunity-out] type=friend host=iax.iaxport.de username=XXXXXXX secret=YYYYYY context=incoming-portunity notransfer=yes [guest] type=user context=default ;callerid="Guest IAX User" And in extensions.conf: [default] ;exten =>

Hi, I have configured asterisk now with ENUM lookups which are working really perfect. Now I want to play a small soundfile before dial the number to inform the caller which protocl is used (SIP, IAX2 or ISDN). How can I do this? With Playback it doesn't seems to work: [iax2-sipport-out] ; with leading 3 using IAX-sipport exten => s,1,NoOp(Dialing ${DIALSTR} with iax2-sipport-out) exten

Hello, I've got a problem at the moment, that setting "transmit_silence = yes" seems to have no effect on Asterisk 1.8-Certified. Although it's enabled and "core show settings" confirms, that it is really enabled, there are no RTP packets sent by Asterisk when waiting for DMTF input or when "Wait()" is called. Also, there seems to be a small gap of 2 or 3

Problem with this is client needs to listen to the call recordings and my interface will only display .wav or .mp3 so they will moan if they have to wait until the next day for today's recordings Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: binary <dreamer.binary at gmail.com> </div><div>Date:01/07/2014 6:09 PM

Hi, I'm having real difficulty in getting calls to go through with Asterisk. I've managed to check that my SIP connection is made to my provider. Below is an email I received from them: ----------------snip--------------------------------snip--------------------------------snip---------------- I am not certain of the reason for rejection but it has to do with the SDP, it does not

I'm creating a test environment for Asterisk. I have Asterisk running on a PC with only a NIC card, No FXO, FXS, TDM cards. I have two Cisco 7960 phones setup for SIP. Within Asterisk, the SIP SHOW PEERS, shows the phones. They don't appear under SIP SHOW REGISTRY. When I call phone 2 from phone 1, I get a message stating it is from Phone 2, stating, Got SIP Response 488 "Not

> On Jun 11, 2006, at 8:15 AM, James Harper wrote: > > > Additionally, just to satisfy myself that I wasn't going mad I changed > > the port from 5060 to 5070 and now things are working, so something is > > definitely playing up on port 5060. > > > If you are behind a NAT perhaps two SIP devices are both trying to use > 5060? > Packets are getting out,

> James Harper wrote: > > >Additionally, just to satisfy myself that I wasn't going mad I changed > >the port from 5060 to 5070 and now things are working, so something is > >definitely playing up on port 5060. > > > >James > > > > > > > You probably have are behind NAT and your NAT device has a SIP ALG. > Changing the port disables

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I'm working on sip trunk between cme 4.x and asterisk (trixbox 1.2.3). Well, the trunk is partially working, asterisk' extensions talk with cme, but - - when cme try to connect to asterisk' number, receives "the number dialed is not in service". - - calls from ISP through asterisk to cme don't work completely,

Are there any known problems with Cisco routers (Cisco 837) and SIP sessions? I have been trying to track down a problem for about 3 hours now and I think the Cisco router is the culprit!!! I keep getting "488 Not acceptable here" messages, which are apparently normally the message you get when a common codec can't be found. I'm also getting "chan_sip.c:3434 process_sdp:

Hi List Asterisk 16.28.0 in use. PJSIP in use Two endpoints Both using IPv6 One Endpoint on UDP, the other via TLS. Both with: t38_udptl=yes ;fax_detect=yes ;fax_detect_timeout=30 rtp_ipv6=yes Both sides are T.38 capable and detect fax tone so no need for fax detection on asterisk. Voice calls between the two work fine. But on a Fax call, I see this situation: A <=> Asterisk

I am helping a friend on one of his sip trunk and couldn't find the way to resolve his problem. His asterisk's problem is like this: 0. When incoming call to one of his sip trunk, Asterisk reply with "488 Not acceptable here". So the call get dropped. 1. Recently upgraded Elastix with Asterisk 1.4.33 2. Was working fine before the upgrade 3. There are total 4 SIP trunks

Hello, I'm trying to send a tif file, using Fax for Asterisk and the call is executed, but when I get the reINVITE with T.38 data, the local server doesn't recognize that we have this capability and sends a 488 message. These are the logs: <--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---> INVITE sip:1234567 at 10.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP

I have a whole bunch of remote devices connected to my Asterisk box, including SPA-3000s, PAP2-NAs and Cisco 7960s. The PAP2-NAs I have only rolled out recently and I am having a problem that is intermittent and inconsistent. It happens to some users but not other users on the same ISP. It happens to users in 2 different countries where the Internet setup (NAT issues) are completely different. It

Hi everyone, we are using custom dfree commands to implement quotas. While these work fine on normal shares, the "dfree command" parameter seems to be ignored in the homes section. Is this correct (and intended)? Best regards Felix IT-Services Telefon 02461 61-9243 E-Mail: f.stolte at fz-juelich.de -------------------------------------------------------------------------------------

The strange thing is its only sometimes my dial string is as follows exten => s,1, Dial (SIP/200,, tT) For that particular route but obviously s,3 as have Ringing () first etc. After she pushes ## 6 times it will go thru sometimes. Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: Kevin Larsen <kevin.larsen at pioneerballoon.com>

Hey guys Is it possible to record with mixmonitor straight into mp3. I am trying to reduce disk space and want my calls to be recorded in mp3 Instead of wav. Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: Sameer Rathod <sameer at hostnsoft.com> </div><div>Date:30/06/2014 9:23 PM (GMT+02:00) </div><div>To:

Hello List; I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and I am getting the following debug, can someone advise me about the solution: <--- SIP read from Provider_IP_Address:5083 --->INVITE sip:22021782 at Asterisk_IP_Address:5060 SIP/2.0?Via: SIP/2.0/UDP Provider_IP_Address:5083;branch=z9hG4bKn1va9h109091cms8h5a0.1?From: "1828444" <sip:1828444 at

Hi all, we use SLES 11 and Lustre 1.8.1.1 + patches and like convert a lustre FS using external journals to one with async journals enabled. Question is whether the procedure: umount <filesystem> on all clients umount <osts> on all OSSes e2fsck <ost-device> on all OSSes for all all OSTs tune2fs -O ^has_journal <ost-device> on all