similar to: FW: issue with sip 180 responses

I don't expect my SIP provider to provide useful "Remote-Party-ID" information. Therefore, I am using "CONNECTEDLINE(num)=xxx" AND "CONNECTEDLINE(name)=yyy" to populate remote party information from a local database. I am also using the "I" (upper case "i") option for Dial. Generally I like to see to see the remote party name appear on the

Hello, SIP invites are accepted from imitator , but 'SIP 180' is not responded back to imitator. By inspecting the issue , we can *see* the response is generated and sent from asterisk (via asterisk logger ("sip debug" )) , but while sniffing the interface with tcpdump, we can't see 180 response on the interface. We don't have errors on the interface, firewall

Hi All, (retrying to post again - somehow message got blocked) I am running 3.2.7-1.fc16.x86_64 (latest FC 16 kernel) I posted regarding this problem earlier, but after some research and found improvement with new kernel version. so I decided to repost in a new thread to clean things up... The intention of this post is to hopefully be useful and point out some performance numbers that devs can

Dear all I'm realy new to R, so I hope you can help me, as I didn't find any solution in the common books. Since some days I'm trying to create the following plot: A scatterplott showning two different groups side-by-side with according regression lines. Both datasets only have the same five factors, so the scatters will form a kind of column at each factor. When I use

How is this insecure? Most large business and wholesale providers use only IP authentication, relying on a session border controller to do the authentication work resulting in great scalability on the softswitch (since it does not have to act as a proxy as well). If they know your IP, and you know their IP, the only risk is that your IP address can somehow be hijacked. IP authentication is

Chris Abel writes: >Hello everyone! > >I've had this problem for a while and cant figure it out. When an outside >caller calls an extension on my asterisk system, they do not hear any >sort of ringing. Inside extensions calling other extensions do hear >ringing. We have 3 other asterisk systems that are configured the same >way, but do not have this problem. We think it

Dear all I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya [asterisk]-----[mediant 2000]--------[Avaya] when i call from avaya side to ---> asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of

Well, I think I discovered even further why there is no ringback tone available. The following message, is displayed on the console in asterisk. ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Looking more into it, I found that it was related to loading tones for a particular zone. The message is printed

Hi, On one of our asterisk systems that is quite busy, we are seeing the following from 'netstat -s': Udp: 17725210 packets received 36547 packets to unknown port received. 44017 packet receive errors 17101174 packets sent RcvbufErrors: 44017 <--- this When this number increases, we see SIP errors, and in particular Qualify packets are lost, and

Hi, We are running on 5.5 on a HP ProLiant DL360 G6. Kernel version is 2.6.18-194.17.1.el5 (we had also tested with the latest available kernel kernel-2.6.18-238.1.1.el5.x86_64) We running some performance tests using the "iperf" utility. We are seeing very bad and inconsistent performance on the UDP testing. The maximum we could get, was 440 Mbits/sec, and it varies from 250 to 440

I am trying to make an Inalp Smartnode 1200 (SIP-to-ISDN gateway) work with Asterisk. It works ... Partially. We are using the Inalp to connect ISDN phones, it basically acts like an ISDN ATA. First of all, when I make a SIP call to the unit with a simple Dial() command (no "r", so Asterisk shouldn't generate its ringback tone) I hear Asterisk's ringback tone anyway (I'm

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following

I would just like to follow-up on the ringback problem I'm getting from *. As I've said in my previous post, I am not hearing the "real ringback" from the Cisco gateway terminating my call. I don't want to provide false ringback from * (r option of dial), because it'll still give me ringback even if I am suppose to hear announcement or fastbusy. Below is captured ISDN

I am trying to get Asterisk to emulate the sounds of the earlier telephone systems, and the settings in [us-old] are pretty helpful. The only thing lacking is ringback tone, which is not quite as complex as the real phone systems of the day. For example, it is true that a ringback tone commonly used is 420Hz modulated by 40Hz. This is what shows up in [us-old]. But that modulated tone was

Hi, I have the following situation: At a branch , there is a Cisco Call Manager with users all having Cisco phones. Now I put down a Asterisk 1.2.12 box at the branch, which talks H323 via chan_oh323 to the CCM. So calls go from the CCM, go H323 to the local Asterisk box, then I take it via SIP to another Asterisk box. From there I am hooked up to 2 different providers, for Local and

Hi, Saw the below config from tinc’s manual: UDPRcvBuf = bytes (OS default) Sets the socket receive buffer size for the UDP socket, in bytes. If unset, the default buffer size will be used by the operating system. UDPSndBuf = bytes (OS default) Sets the socket send buffer size for the UDP socket, in bytes. If unset, the default buffer size

Hi to all I'm a new user of TDM400P card. The configuration is OK and I have no problem for incoming call. When I try to place a outgoing call towards a PSTN number the call is not always answered. In other words TDM400P seems to not understand that the call has been accepted from the remote party. In other cases (different extension) the call is accepted succesfully. In my opinion TDM400P DSP

are you giving answer()? ..o-------------------------------------------------------o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users

Hello! An extension registered at asterisk 13.23 initiates an external call (pjsip). After the Invite, the callee (-> ISP) sends a 100 Trying 183 Session Progress (*without* SDP) Asterisk now sends to the extension: 183 Session Progress (*with* SDP) 183 Session Progress (*with* SDP) (really two times) The callee meanwhile sends 180 Ringing (*without* SDP) which is