similar to: Cisco router and "488 Not acceptable here"messages

> 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

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:

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

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

Hello, a person trying to call me by my phone number is getting the error 488 Not acceptable here. I googled that error, seems like this error is normally caused by a failed codec negotation, though I have no clue how I could have read this out of the logs. Anyway, my setup is as follows: Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider The user calling me is also using Sipgate and is calling my

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

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

I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone are registered by the below information *CLI> sip show peers Name/username Host Mask Port Status 2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored 2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored 2000/2000 192.168.22.198 (D)

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

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

Hello, I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec in my ata the the GW return a media not acceptable error. but If i add the g729 codec the all is ok. I see in the config of the cisco where to define codec for imcoming call but not for outgoing *Jul 17 15:57:02.604: Received: INVITE sip:0041787518551 at 192.168.0.110 SIP/2.0 Via: SIP/2.0/UDP

Hello, I'd appreciate your comments on the following problem I'm having, please forgive me if this is something obvious, I've been scratching my head on this for a while: I have Asterisk+Kamailio setup where I'm currently testing inbound calls from outside. I have both webrtc and sip clients, where webrtc peers are defined according to sip.js instructions (

Hi guys, Any idea why I am getting this error when someone tries to send me a T38 Fax?

Kathy, Thank you very much! Peter Olson Alcatel-Lucent Member of Technical Staff Services Technologies 1960 Lucent Lane Rm 7G334 Naperville, IL 60566 peterolson@alcatel-lucent.com Phone: 630 979 0573 Mobile: 630 430 6926 -----Original Message----- From: xen-users-bounces@lists.xensource.com [mailto:xen-users-bounces@lists.xensource.com] On Behalf Of xen-users-request@lists.xensource.com Sent:

Hello, I've been racking my brain over this for much of the day so I thought the list would probably be more helpful. A few days ago I upgraded from Asterisk 1.2 to Asterisk 1.4.5. Everything appeared to be working properly. However, on the first business day, we realized that when transferring calls (not using call parking, using the built in transfer buttons on a Cisco 7960) would not

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

Hi All. I'm running some tests with the latest Asterisk SVN-branch-12-r410493M compiled with fresh github pjsip and srtp 1.4.2 on an i386 centOS machine (2.6.32-358.18.1.el6.i686). As a client I'm using the sipMLP WebRTC javascript softphone running on Chrome 33.0.1750.146 m. I have the softphone correctly registered on the Asterisk machine but as soon as I try to start a new call

hello everybody, i want to configure a sip trunk between my system which has asterisk 11.5.1 and a cisco router. this is my scenario: Freepbx-----my system-----cisco-router----Freepbx my system acts like a router. if i set just one codec in dial-peers on cisco router, every thing is ok and i can make a call. but if i set different codecs in a voice class codec and assign it to dial-peers in

Thought I did already, but I've been pretty absent minded recently :) http://voipnow.com.au Or more specifically: http://voipnow.com.au/xcart/catalog/Saphir-III-ML-PCI-p-16151.html It's the HST Saphir III ML PCI. Price is on the website. Enjoy! James > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- >