similar to: Music On Hold + canreinvite=yes

Displaying 20 results from an estimated 10000 matches similar to: "Music On Hold + canreinvite=yes"

2005 Aug 14
4
Multiple Asterisk Installations + SER
I'm trying to implement a shared asterisk server for multiple (different) companies. Here's what I've done so far: - I've installed multiple asterisk instances on one server (via vserver). Each * is for one customer, and has it's own extensions (like 100, 101, 102, etc.) Note that the same extension can exist on other * instances - The SIP Clients register themselves with * -
2006 Jan 13
2
Use Grandstream ATA as trunk
Hi All, I have a GSM box, which needs to connect to a analogue phone line. I've plugged the GSM box to a Grandstream ATA (386). This ATA has extension number 600. Now what I want to accomplish is the following: - If a mobile-number is chosen by a user, asterisk needs to call the ATA (600), wait for a few seconds, and then send the mobile-phonenumber. Or, if it's possible, define the
2006 Feb 14
3
Grandstream hold one way audio -URGENT
Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's
2005 Aug 19
1
Nat + Asterisk + Ser (Far end Nat Traversal)
Hello, I have several * servers behind a SER server (in a local ip range). The SER server is also publicy reachable. On the other site, I have SIP clients that are behind another NAT or in the same NAT range as the * server. Can someone give me some directions/hints etc. on how to make this work. I think I should be using MediaProxy with SER. But do the SIP clients need to register at the SER
2005 Aug 30
0
canreinvite = yes with PAP2
Has anyone made this work? For me everything is fine until I switch canreinvite form no to yes. What happens is that asterisk hangs on "attempting native bridge" ... from what I understand "attempting native bridge" means that the RTP is routed through asterisk (just without any codec translation) But it shouldn't do that ... right? ... canreinvite is set to yes ...
2004 Jan 16
2
No subject
Hello! > Date: Thu, 15 Jan 2004 16:53:18 +0100 > From: Kirill Ponomarew <krion@FreeBSD.org> > > > freshly updated ports tree on a 4.9 box is exactly the same as a > > > freshly updated ports tree on a 5.2 box. > >=20 > > Read the users email. They're using specific tags, not "." so there are > > (or may be) some
2005 Feb 17
4
functional difference: canreinvite=yes, no, or update
Can anyone give an example of the difference between the following: canreinvite=no canreinvite=yes canreinvite=update Here is the problem: I have an 800 number sent to me via SIP from a national carrier. Asterisk gets the number and rings my desk phone. Asterisk has 2 NICs, one with public IP and private IP. My phone is on private IP, the inbound call is on public. My phone rings and I answer
2006 Mar 13
3
Callerid on transfer
Hello, Suppose customer A calls attendant. CallerID of A is displayed at the attendant. But, when attendant does a consulted transfer to, let's say, B, the callerID of attendant is displayed at B. When the consulted transfer is succesful, the callerid of attendant is STILL displayed at B. Is it possible to, after a successful transfer change the callerid of the attendant in the callerid of
2006 Feb 11
2
No Voice when canreinvite=no
Hi all I am using Asterisk 1.2.2 on frdora core 4. i have two sip UA. if i put canreinvite=yes voice Ok on both sides. and if i change canreinvite=no there is no voice (media through asterisk) one thing more if i try to use playback application for playing some sound file it is also working (like exten => 500,1,Playback(demo-abouttotry) this is working). here is sip.conf
2003 Oct 19
1
Music on hold...
No, you don't need a sound card. Do you have ztdummy loaded or zaptel device in your system? Regards, Gus ----- Original Message ----- From: "Chris Hariga" <contact@techselesta.com> To: <asterisk-users@lists.digium.com> Sent: Sunday, October 19, 2003 8:19 PM Subject: [Asterisk-Users] Music on hold... > Hi, > > I need a sound card and mpg123 for music on
2006 Mar 30
1
Multicast Music on Hold
As I understand there is provisions for hold in both RFC 2543 (SIP) and RFC 3264 (SDP) and asterisk users the latter. The RFC 2543 method tells the UA its media stream is at 0.0.0.0 where as the method via SDP can tell it to listen to any address/port/ protocol combination, which is how asterisk tell it to listen to the audio stream it presents when it is asked to hold a call. Please
2006 Jan 28
2
RoadRunner
I use SIP over VPN with RR from TWC no problem, connect via WiFi. According to http://www.speakeasy.net/speedtest/ I am getting 3.5Mbps down and 353Kbps up at this time (6:15pm Saturday). My laptop currently has an X-Lite (free version) softphone with GN Netcom USB professional contact center headsets (GN8110 USB XP adapter). We have found that the headset makes a major difference in the quality
2005 Jan 14
2
Passing PIN Numbers
To All If anyone can shed any light on this it would be greatly appreciated. My phones are unable to enter pins numbers correctly when required by the party they are calling. For example I was given an outside number to attend conference bridge. After the call was connected it required me to enter a 4 digit PIN. Now here is the problem whenever I enter a pin it is received twice. For example if
2003 Sep 11
2
SIP client<->NAT<->Asterisk<->NAT<->SIP client. only works with canreinvite=no.
Hi! I have this configuration: SIP client A <-> NAT box A (real external IP) <-> Asterisk server (real IP) <-> (real external IP) NAT box B <-> SIP client B The echo test form any of the clients to the asterisk server is working just fine, even without canreinvite=no. When I try to call from SIP client A to B, wihtout the canreinvite=no in the sip.conf, the call
2005 Jul 27
2
Music on Hold: CPU Intensive Monster
OK. So I did a test last night. All of asterisk's threads where using 0.0% CPU. I made 1 call to our call queue. CPU jumped to average of 9% and stayed around that for the 2 minutes I was in the queue just listening to music on hold. MOH is in MP3 format and I'm using format_mp3. Phone was linksys PAP2-NA using G729. Can I reasonably assume that the 9% was decoding the MP3, then
2005 Aug 22
1
FW: Nat + Asterisk + Ser (Far end Nat Traversal)
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2005 Jan 28
2
Fwd and Tollfree
Hallo all do any of you know if the toll free access to the Netherlands is still working via FWD or Iaxtel? thanks liaan --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Sep 26
2
Early Media in 180 Ringing
Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 180 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP
2006 Dec 11
2
Asterisk Sends 180-RINGING to UA even withprogressinband=yes
Andrew, I don't think it's a Polycom issue. We took Asterisk out of the picture and had our Polycom phones communicate directly with an Audiocodes PSTN gateway. Unlike Asterisk, the audiocodes do not send 180 Ringing before sending 183 Session Progress, and the polycom's play the correct tones in this case. We WANT Asterisk to send progress tones in band. In our case it IS needed.
2007 Sep 09
3
nat=yes
Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal