Im experiencing a problem with a current setup and I've run out of ways to debug it and come to a resolution. I have two E100P's in a machine which is routing traffic over the internet to a machine that has 1 E400P connected to the PSTN. Clients are able to make calls successfully but when the call is connected experience one-way audio. They cannot hear anything said by the person who is being called. This appears to be intermittent, with occasional times where conversations can be held, but what sounds like silence suppression on the side of the caller. They hear audible clicks and then silence, and assume the person they have called has hung up. On the side of the person being called there are no audible problems at all (except for the fact that most of the time there is nothing being heard on their side). so the link between the two machines is using a 512k connection at one end, and a 2meg connection at the other. an IAX2 trunk between them. the only thing i can think of as a potential cause of this problem is that one is using a cvs thats more recent than the other. im worried about updating the older machine without being sure that its not going to introduce problems as there are many other applications and services running on that server. any help would be much appreciated duncan
>Im experiencing a problem with a current setup and I've run out of ways to >debug it and come to a resolution. > >I have two E100P's in a machine which is routing traffic over the internet >to a machine that has 1 E400P connected to the PSTN. Clients are able to >make calls successfully but when the call is connected experience one-way >audio. They cannot hear anything said by the person who is being called. > >This appears to be intermittent, with occasional times where conversations >can be held, but what sounds like silence suppression on the side of the >caller. They hear audible clicks and then silence, and assume the person >they have called has hung up. On the side of the person being called >there are no audible problems at all (except for the fact that most of the >time there is nothing being heard on their side). > >so the link between the two machines is using a 512k connection at one >end, and a 2meg connection at the other. an IAX2 trunk between them. the >only thing i can think of as a potential cause of this problem is that one >is using a cvs thats more recent than the other. im worried about >updating the older machine without being sure that its not going to >introduce problems as there are many other applications and services >running on that server.ooh replying to my own posts again. bad duncan. some more information on the problems being encountered. if i have an IAX2 Trunk - the person making the call cant be heard but the person recieving the call can hear them if i have a standard call over IAX2 the person recieving the call cant hear anything but can be heard by the person making the call if i have a standard IAX call - both sides can hear each other fine. any ideas where the problem lies based on this new information? duncan
> person recieving the call can hear them > if i have a standard call over IAX2 the person recieving the call cant hear > anything but can be heard by the person making the call > if i have a standard IAX call - both sides can hear each other fine. > > any ideas where the problem lies based on this new information?You should be able to see useful information on the console. One-way audio with IAX2 is, generally speaking, not possible -- especially not without trunking. Turn off the jitter buffer in your iax.conf to be on the safe side. Also, tcpdump is your friend. Remember, IAX2 only uses a single port, so you should be able to see traffic in both directions. Mark
I have a toll-free number inbound from sixTel. This gets answered by my IVR system. If they choose "technical support", I have it dial my SIP phone (SJPhone) and my cell phone (through sixTel) at the same time. If I answer on the SIP phone, all is well. If I try to answer on my cell phone, they can hear me but I cannot hear them. I am using IAX2 with ulaw only as my connection to sixTel. Anyone know where to point me in my search for resolution? --- Kelly D Griffin Network Engineer Tantella Wireless http://tantella.com 479.273.9992 Voice 479.464.8998 Fax
Hi all, Maybe someone encountered similar issue. I have an * with the incoming DID over SIP. * is behind a firewall. I have no issues with other SIP devices connected from the outside network, however on that DID when I receive a call I can hear only incoming audio, no outgoing. If I setup a "playback" with some audio stream, * just disconnects the call right after it receives it. The same issue happens no matter which client is being connected to that DID. For example: [inbound] exten => xxxx225612,1,SetAccount(xxxx225612) exten => xxxx225612,2,Ringing() exten => xxxx225612,3,Dial(SIP/bt101,50) exten => xxxx225612,4,Hangup If I change Dial(SIP/bt101,50) to Dial(IAX2/firefly,50) it does not change anything. This example can only receive audio. This one just answers and disconnects call the same second: [inbound] exten => xxxx225612,1,SetAccount(xxxx225612) exten => xxxx225612,2,Answer exten => xxxx225612,3,Playback(vm-goodbye) exten => xxxx225612,4,Hangup Sincerely, --Andy x6722
Have you looked at the various comments regarding NAT on the wiki? I think you need to set the following in sip.conf nat=yes localnet=192.168.0.0/255.255.255.0 (or whatever) externip=WAN IP address canreinvite=no Regards Cameron ----- Original Message ----- From: "Andrejus Stavickis" <andy@loyalty.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Tuesday, April 19, 2005 3:57 AM Subject: [Asterisk-Users] One-way audio Hi all, Maybe someone encountered similar issue. I have an * with the incoming DID over SIP. * is behind a firewall. I have no issues with other SIP devices connected from the outside network, however on that DID when I receive a call I can hear only incoming audio, no outgoing. If I setup a "playback" with some audio stream, * just disconnects the call right after it receives it. The same issue happens no matter which client is being connected to that DID. For example: [inbound] exten => xxxx225612,1,SetAccount(xxxx225612) exten => xxxx225612,2,Ringing() exten => xxxx225612,3,Dial(SIP/bt101,50) exten => xxxx225612,4,Hangup If I change Dial(SIP/bt101,50) to Dial(IAX2/firefly,50) it does not change anything. This example can only receive audio. This one just answers and disconnects call the same second: [inbound] exten => xxxx225612,1,SetAccount(xxxx225612) exten => xxxx225612,2,Answer exten => xxxx225612,3,Playback(vm-goodbye) exten => xxxx225612,4,Hangup Sincerely, --Andy x6722 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Hi Cameron, Yes I did looked at those pages, and tried to configure NAT and externip and localnet and all that jazz, but still no audio. I'll definitely try to look at those options agai, since the device on outside network can communicate fine. Sincerely, --Andy x6722 "Outsourcing is akin to making a skyscraper taller by taking material from its lower floors." --Byron Katz> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Cameron Beattie > Sent: Wednesday, April 20, 2005 11:41 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] One-way audio > > Have you looked at the various comments regarding NAT on the > wiki? I think you need to set the following in sip.conf > nat=yes localnet=192.168.0.0/255.255.255.0 (or whatever) > externip=WAN IP address canreinvite=no > > Regards > > Cameron > ----- Original Message ----- > From: "Andrejus Stavickis" <andy@loyalty.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Tuesday, April 19, 2005 3:57 AM > Subject: [Asterisk-Users] One-way audio > > > Hi all, > > Maybe someone encountered similar issue. > > I have an * with the incoming DID over SIP. * is behind a firewall. I > have no issues with other SIP devices connected from the outside > network, however on that DID when I receive a call I can hear only > incoming audio, no outgoing. If I setup a "playback" with some audio > stream, * just disconnects the call right after it receives > it. The same > issue happens no matter which client is being connected to > that DID. For > example: > > [inbound] > > exten => xxxx225612,1,SetAccount(xxxx225612) > exten => xxxx225612,2,Ringing() > exten => xxxx225612,3,Dial(SIP/bt101,50) > exten => xxxx225612,4,Hangup > > If I change Dial(SIP/bt101,50) to Dial(IAX2/firefly,50) it does not > change anything. > > This example can only receive audio. > > This one just answers and disconnects call the same second: > > [inbound] > > exten => xxxx225612,1,SetAccount(xxxx225612) > exten => xxxx225612,2,Answer > exten => xxxx225612,3,Playback(vm-goodbye) > exten => xxxx225612,4,Hangup > > Sincerely, > > --Andy > x6722 > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >