similar to: Problem with Voice Quality

Displaying 20 results from an estimated 6000 matches similar to: "Problem with Voice Quality"

2006 Apr 19
0
Problem with Voice quality, please help
Hi All, We made a VOIP application for PDA's (PALM OS) and we are using both SER and Asterisk. SER is SIP proxy and it routes all the calls to Asterisk. On SER we have RTPProxy also. My problem is that I am getting a weird noise or disturbance for all the calls at an approximate time interval of 100-120 seconds and we are getting this noise consistently. After 5-10 seconds everything
2004 Dec 23
8
asterisk at large
Hello *'s, First Of all Marry Christmas, I want to setup asterisk at large means "my main asterisk server placed in my office(in Pakistan), and some offices outside Pakistan and i want to connect these locations to my main * server (in Pakistan) on remote locations i'll used asterisk can i do this or may be i changed my plans kindly guides me. Thanks In Advance. Adnan Ahmed.
2006 Feb 23
2
Configure DID
Hi All, I am a newbie to Asterisk and I was able to install Asterisk and call out. Recently I purchased two DID's, can someone please tell me or point to some links showing how to configure these DID's for SIP based softphones like Express talk? Thanks, Manoj.
2006 Jan 17
2
Problem configuring Asterisk, Please help me
Hi All, I am a newbie to VOIP and after some problems I was able to install Asterisk. If I start Asterisk I could find "Asterisk Ready" at the end and I am thinking that Asterisk is started successfully. Later after changing my Extensions.conf and ser.conf nothing works, I could still see the message "Asterisk Ready" but when I try using DIAX and connect to Asterisk nothing
2006 Feb 28
1
Problem calling out
Hi All, I installed Asterisk recently and it was working from 2 weeks without a problem until today. Today it started showing strange error Feb 28 03:14:08 WARNING[31430]: chan_sip.c:4826 check_auth: Stale nonce received from '<sip:18006733555@mantragroup.com>' Whatever number I call it displays this, please tell how can I fix this? I have no idea what is happening and the cause
2006 Feb 28
1
Problem with incoming call, Please help
Hi All, I was able to install Asterisk and make outgoing calls. Recently I purchased two DID's and I am facing a problem configuring them to my Asterisk, I hope with the help I get from this list I will be able to configure successfully. Mu errors are Feb 28 08:31:58 NOTICE[19133]: pbx.c:1331 pbx_extension_helper: Cannot find extension context 'context_mantra2' Feb 28 08:31:58
2006 Mar 13
1
Asterisk RealTime Question, Please help
Hi All, I was able to install Asterisk and Asterisk-addons and use them successfully. But I have a problem now, I have many contexts and it looks like Asterisk is unable to find the context given directly in Mysql DB unless I specify it in Extensions.conf to switch it to RealTime. If I add a new context in Mysql then I have to add it in Extensions.conf and reload extensions whenever I need a new
2006 Mar 03
2
Asterisk Fax Question
Hi All, I want to configure fax with Asterisk and I found that we can do this reliably using G711 codec only. Currently my provider is supporting G729 and G711. During the call initiation the call starts with G729 (1'st priority) and somehow if the receiver is unable to receive call then we are providing the Caller to send a fax, but at that point they are using G729 codec. At this point how
2006 Jan 16
1
Problem with installation of rpm's, Please help me.
Hi All, I am a newbie and trying to install Asterisk from instructions given in http://www.voip-info.org/tiki-index.php?page=Asterisk+RPM. We have Centos 3.3 so I downloaded rpm's from ftp://ftp.linuxsys.com/pub/LSE/packages/CentOS-3.4/asterisk-1.0.9/ and tried installing one by one but I get the following errors error: Failed dependencies: asterisk = v1.0.9 is needed by
2006 Jan 27
1
Packeting multiple GSM frames in one IP packet - Help needed.
Hi, We have a task to reduce voice call bandwidth. IP+UDP+RTP are using 40 bytes per packet and for voice GSM FR 33 bytes. We are trying to reduce this bandwidth accommodating multiple GSM frames in one packet. If we want to use per packet 10 GSM frames how to do this using asterisk? Assume the sip client is able to split these packets in to individual GSM frames. Any help will be sincerely
2006 Mar 06
1
Extension 's' in Realtime
Hi All, I was able to insert some extensions in Mysql DB and use them successfully. In Mysql extensions table the priority column is of type tinyint and when I give 's' value for it, it is not accepting that value as it takes only tinyints. Please tell how can I make that column accept values like t,s,i and make it work with asterisk in realtime without any problem? If I change the type
2003 Jun 10
4
PDA's over SIP channels on Asterisk
Is it possible for two PDA's to communicate like telephones via SIP channels on a PC running Asterisk? If that is possible, does there exist any applications that can be installed on a Zaurus 5600, which is a PDA with an Xscale processor running on a Linux OS, that can essentially turn it into a softphone? Thanks in advance for any input, Daniel
2008 Oct 22
3
asterisk video
hi, hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start
2016 Feb 18
2
Asterisk behind RTPproxy | On-Demand SDP engagement
Hi All, I've been wondering if I can instruct asterisk in the dialplan to engage the Media handling for a particular call or not. I've SIP users behind Kamailio & RTPProxy, and I can make use of sip.conf setting "directmediadeny|directmediapermit" to offload media from asterisk for peer-to-peer calls BUT what if someone wants to record a call or engage some feature-code ?
2004 Jun 25
1
SER and NAT
I have a really simple question about a fairly complex problem: I have a Cisco 7960 behind a NAT. I have an Asterisk server behind a different NAT. I have a SER server (with rtpproxy installed) on a public IP adress. I've opened ports with static NAT to * and the Cisco. Without using SER, I can register the phone to *, I can complete calls, I just can't move audio. Reading the
2011 May 12
1
Different IP addresss for SIP and RTP
Hello, is it possible to set an IP address for RTP different than the one used for SIP? I want to use asterisk behind a sip proxy (opensips), but I was thinking if I could avoid having to run rtpproxy on the sip proxy server and let asterisk itself take care of it. So that: Asterisk SIP address : local ip address Asterisk RTP address : global ip address regards, takeshi -------------- next
2009 Jul 27
2
Asterisk and Kamailio NAT problem
Hello Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=ulaw
2005 Oct 11
2
nat and wandering phones
Hi all I'm looking for a solution to this problem. *box--------internet-----------nat-----------softphone We have potential customers who will be travelling the world with laptops/pda's. They need to be able to connect to the asterisk box via ip wherever they are and will have no control over nat whatsoever. I have read that STUN offers this service, but cannot picture in my mind how
2004 Dec 19
1
sip phones in different private networks have one way audio
Hello I have one phone (phone1) in one network, the other (phone2) in public network. both can call the other side; phone1 can be heard by phone2, phone2 can't be heard. I don't have NAT set on both end, but I use rtpproxy on SER. Is NAT still necessary to be set on both phones? Thank you! steven
2005 Sep 30
1
Empty ACK
Hello, I have asterisk connected to SER/RTPProxy which is again connected to a IP-PSTN gateway. When calling with a UA, registered at * to a SIP phone connected to the IP-PSTN gateway, I get 'empty ACKs': U 192.168.0.173:5060 -> 10.254.254.1:5060 ACK SIP/2.0. Via: SIP/2.0/UDP 192.168.0.173:5060;branch=z9hG4bK5cb7d048. Route: