similar to: One way audio problem

Hi, On some SIP interconnects with devices like Cisco, Dialogic we get SIP invite from different source port every time and asterisk rejects that INVITE. Does anyone knows solution for this? --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications T: +44(0) 871 434 9151 +44(0) 121 620 9151 Email: deepika.nijhawan at oxygen8.com Skype:

Hi, We have free pbx installed on asterisk 1.4.25.1. Mysql is installed and asterisk is connecting to it. CDR modules are all loaded as well. For some reason, it is not creating master.csv and no cdrs are generated. Can anyone help please. --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications T: +44(0) 871 434 9151 +44(0) 121 620 9151

Hi, I have followed steps which were mentioned on forum and given below. Still couldn't get speex working. On test calls getting error "chan_sip.c: sip_call: No audio format found to offer." # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not

Hi, On freepbx (GUI), whatever reason number fails we always get 'all circuits are busy' audio. Does anybody know how to get far end audio when we dial wrong number or when it's busy or unallocated number or failed with some other reason. Thanks, Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL:

----- Asterisk 1.8 will allow to read SIP response codes in the dialplan via ${HASH(SIP_CAUSE,<channel-name>)} Asterisk 1.8 also comes with a 'use_q850_reason' configuration option = for generating and parsing, if available:=20 ----- That will give you what you want if you consider upgrading to v1.8. =09 -----Original Message----- From: asterisk-users-bounces at

Hi, Does anyone has an idea how to tell asterisk to use codec A for first 50 calls and then codec B for rest of the calls. Thanks, Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/6114bf1d/attachment.htm

Hi, I am doing failover routing based on 2 dial commands. First route sends back 4xx response and I don't want it to try 2nd route when it is 4xx response. Can we do failover routing based on SIP 5xx response only ? Thanks Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have connected asterisk 1.6.2.6 with an IVR and ran dahdi_genconf. Dahdi status is not showing alarms but channels are not coming up. It is not showing any channels when i run 'dahdi show channels'. Could anyone help pls. Thanks Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am trying to configure ipsec on asterisk. Have configured /etc/racoon/racoon.conf and /etc/raccoon/psk.txt. Also have policy file in same folder. Have run racoon. Still I can't receive calls. Can anyone please tell if any extra step is needed. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi D'arcy Have you checked your RTP port ranges (I'm sure you have), and also that the server IP for RTP as specified in the initial SIP is correct? Not sure how this will relate to your setup, but we had something similar here using Asterisk 1.8.11.0 on both sides of the connection, via a VOIP service provider in the middle. We had slightly different parameters, e. g. that we would

Hello all, I have a problem where problem with one way audio, and I think it's related to "a=sendonly" and a re-invite. Can anyone please assist? The scenario is as follows.... - We send an INVITE to a peer, and it replies with a "100 Trying", and then a "183 Session Progress" message containing "a=sendonly". - Asterisk plays the caller music on hold,

Hi, I have a couple of users who are having a peculiar problem. On some outbound numbers where there is a deep IVR tree (3+ selections), and then a live person picks up, the live person will be unable to hear them on the phone, but they can hear the live person. I've done packet traces and it appears as though audio is being passed both ways, but the audio from the caller is severely muted

Hi Everyone, I have setup an OpenVPN tunnel between Server A (running Asterisk) and Server B suppling it's SIP Phones with DHCP pool of IPs. So, the tunnel is established nicely and everyone can ping others. "sip show peers" shows the local subnet of the SIP Phones registered (192.168.100.0/24 ). But there is the old bad one-way audio. Calls also drop after few seconds. In the SIP

I'm running Connected to Asterisk 1.4.18-1. We have a T1 PRI and a SIP trunk coming in from our IVR. When people call in on the SIP trunk, I experience the following problem: I put the caller on hold, MOH starts. MOH quits after approx ten seconds. When I pick the call back up off of hold, they can't hear me for about 20 seconds (sometimes longer or not at all) but I can hear them. This

Hi, I am experiencing one-way audio from: SIP Device ----> Asterisk -----> Cisco 7200 The Cisco 7200 has a VXE+ card that will allow you to do SIP. I can pass audio from SIP Device to Asterisk through the Cisco 7200 to the other end, but the Cisco 7200 does not return any audio back to the SIP Device or Asterisk, it seems. I have tried upgrading to 12.3T IOS version, but no

Hi all, I'm getting one way audio when calling over the SIP trunk i.e. end device B (remote end of SIP trunk) can hear device A (softphone registered with Asterisk) but device A can't hear device B. Even though I configured same NAT configurations on other servers and they are working good. The NAT configuration is listed below; localnet=130.0.0.0/130.0.0.0 externhost=12.131.12.13

I am using 0h323 on Asterisk CVS HEAD 19/07/2005. I am dialling a h323 gatekeeper. He can hear me, but I cannot hear him. I have a suspicion that it could be the rtp traffic, since he said that they need rtp traffic from ports 4500 - 65000. So in 0h323.conf i set updstart and udpend, and in rtp.conf i set the ports here. a tcpdump confirms there is two way traffic. unfortunately, a 0h323

Hello everybody, I?ve been trying to solve a problem for several weeks now but it really beats me. There are several hard phones connected to an Innovaphone 3000 VoIP gateway. On the other side I have a SIP softphone connected to Asterisk. The problem I have is that on incoming calls (hardphones to softphone) I only have outgoing audio (from soft to hardphone); everything is OK when I call the

We have zoiper connected over iax to asterisk in Sydney. The call is to asterisk in New York. The caller in NZ can hear clearly. Nothing in NY. Here's the sydney server: -- Accepting AUTHENTICATED call from <zoiperipaddr>: > requested format = speex, > requested prefs = (), > actual format = ulaw, > host prefs = (silk16|ulaw|gsm|g722),

My users dial *120 get to an IVR menu that plays their balance and then ask them for a voucher. Ater the balance is played and the request for the voucher is played the user don't hear any other audio from the asterisk box. I can see the asterisk server playing the files to ask for the voucher again but the user cannot hear any thing. Has any one seens this issue with IVRs. I notice a