Re: [asterisk-users] Sporadic one way audio problem
deny=0.0.0.0/0.0.0.0 permit=XXX.XXX.X.X/29 permit=192.168.1.0/24 Are you sure your provider *always* sends data from this /29? Maybe you have this in your iptables as well and sometimes audio is received from outside this /29 and therefore blocked? -- Andreas Sikkema -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sporadic one way audio problem
Hi, deny=0.0.0.0/0.0.0.0 permit=XXX.XXX.X.X/29 permit=192.168.1.0/24 Are you sure your provider *always* sends data from this /29? I'm sure, yes. Its a MPLS net, only voice inside. Regards, Georg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sporadic one way audio problem
Hi all again, I've got a problem with sporadic one way audio calls, which means sometimes I can't hear the calling party (call is established, but audio is missing). Today I received ~90 calls, one of them got this problem. I've got two networks involved, without NAT: - 192.168.1.X, in there one nic of my server and all the phones - a private net to my provider, in there a nic of my server and the voice switch of my provider My sip.conf: [general] port=5060 bindaddr=0.0.0.0 language=de allowguest=no ;echocancel=yes ;echotraining=yes alwaysauthreject=yes disallow=all allow=alaw deny=0.0.0.0/0.0.0.0 permit=XXX.XXX.X.X/29 permit=192.168.1.0/24 ;jbenable=yes ;jbforce=yes ;jbmaxsize=20 ;jbresyncthreshold=1000 tos=0x10 directmedia=no nat=no directrtpsetup=no [provider] type=peer host=XXX.XXX.X.X insecure=port,invite context=X dtmfmode=rfc2833 directmedia=no nat=no directrtpsetup=no ;qualify=300 [one-phone] [10] type=peer context=X secret=XX host=dynamic ;qualify=300 directmedia=no nat=no directrtpsetup=no dtmfmode=inband Any help greatly appreciated! Thanks, Georg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sporadic one-way audio
2009/10/16 Ishfaq Malik i...@pack-net.co.uk: Brent Davidson wrote: We have several offices running Asterisk version 1.4.20.1, and OSLEC with Rhino R4FXO-EC and one running a Digium TDM800P card for interface to analog lines. All offices are running Snom 300 phones. Phones all have static addresses and are on the same physical network as the server. The problem we are having is that every so often we get someone calling in where we can hear their voice, but they can't hear us. If we immediately call them back everything is fine. The problem affects all offices and also happens when making sip to sip calls from one snom 300 to another. In addition we periodically have calls that drop off in the middle of a conversation like the connection was lost. I haven't been able to replicate any of these problems and the people that are having them can't seem to keep track of when they occur so I can go back and look in the logs. I suspect that both problems may be related though. Possibly a registration issue? Any ideas are welcome. Hi For the sporadic one way audio, check that the codec list in the snom phones is the same as set by the server. The codec list is in the RTP tab of the identities. Hope that helps Ish Also check that your DNS servers are working and can resolve everything. I saw an issue similar to what you describe when a snom300 could not resolve its NTP server address. I cannot explain it, but sorting out DNS sorted out the communication issue. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sporadic one-way audio
Hi For the sporadic one way audio, check that the codec list in the snom phones is the same as set by the server. The codec list is in the RTP tab of the identities. Hope that helps Ish Brent Davidson wrote: We have several offices running Asterisk version 1.4.20.1, and OSLEC with Rhino R4FXO-EC and one running a Digium TDM800P card for interface to analog lines. All offices are running Snom 300 phones. Phones all have static addresses and are on the same physical network as the server. The problem we are having is that every so often we get someone calling in where we can hear their voice, but they can't hear us. If we immediately call them back everything is fine. The problem affects all offices and also happens when making sip to sip calls from one snom 300 to another. In addition we periodically have calls that drop off in the middle of a conversation like the connection was lost. I haven't been able to replicate any of these problems and the people that are having them can't seem to keep track of when they occur so I can go back and look in the logs. I suspect that both problems may be related though. Possibly a registration issue? Any ideas are welcome. Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sporadic one-way audio
We have several offices running Asterisk version 1.4.20.1, and OSLEC with Rhino R4FXO-EC and one running a Digium TDM800P card for interface to analog lines. All offices are running Snom 300 phones. Phones all have static addresses and are on the same physical network as the server. The problem we are having is that every so often we get someone calling in where we can hear their voice, but they can't hear us. If we immediately call them back everything is fine. The problem affects all offices and also happens when making sip to sip calls from one snom 300 to another. In addition we periodically have calls that drop off in the middle of a conversation like the connection was lost. I haven't been able to replicate any of these problems and the people that are having them can't seem to keep track of when they occur so I can go back and look in the logs. I suspect that both problems may be related though. Possibly a registration issue? Any ideas are welcome. Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sporadic One Way Audio
I don't think it's a snom specific issue as I have 5 branch offices all with identical configurations and only one of the 5 is experiencing this problem. I have checked and re-checked the config files until they're practically burned into my brain and everything appears looks OK. Any place there is a config option for the server address it matches the server's physical address. The phones and all computers on the network are statically assigned (these are very small 1-person or 2 at the most offices) so I'm absolutely sure that there is no chance of an IP conflict. The phones and the Asterisk server are plugged in to the same ethernet switch. I have tried tried using both sip debug and Wireshark to try to catch anything relating to this, but the calls seem to happen very rarely and, as usually happens when I'm trying to troubleshoot some sporadic problem, the two people in that office can't ever reliably reproduce this when I happen to be in the office or remotely logged in to the server. To make things even more difficult, the Snom phones have a switch built in to them and the PC's are in the pass-through port. This means that any traffic generated by the phones is in the PC's blind spot. Anyway, I've got about 1000 other projects going right now. Once I get a couple of the more pressing ones knocked out I'll try to spend a couple of days at the problematic office and see if I can't come up with some way to resolve the problem. Thanks for the suggestions, Brent OCG Technical Support wrote: Well, if this is snom specific I can't offer more insight. It really sounds like misconfigured iptables and/or sip helper (conntrack/nat/etc). Are you sure your IP address is right in your sip.conf? If you don;t have NAT set to yes for these phones, they will trust the sip header for IP address and may misroute. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson Sent: October 24, 2008 7:36 PM To: Asterisk Users List Subject: Re: [asterisk-users] Sporadic One Way Audio Importance: High The asterisk server is connected to the PSTN via a Rhino R4FXO-EC card. The lost RTP would have be between the Asterisk server and the phones. There are only 2 phones in the building, 2 lines coming in to the asterisk server and the server is on the same ethernet switch as the phones. The phones are SIP phones. This is a simple PBX system that picks up calls from the analog lines and routes them to the appropriate phone, although it will eventually be linked to a larger system once all the minor bugs are resolved. OCG Technical Support wrote: How is your asterisk server connected to the PSTN? SIP/IAX out...ISDN/T1 out? Etc... Are you looking for lost RTP between * and internal phones or * and external provider? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson Sent: October 24, 2008 5:55 PM To: Asterisk Users List Subject: [asterisk-users] Sporadic One Way Audio I'm having an unusual problem at one of my branch offices. Every now and then they will make a call and the person they call is unable to hear them, but they are able to hear the person. The Asterisk server has only one ethernet interface and is on the same physical network as the 2 snom 300 phones and is connected to the PSTN lines with a Rhino R4FXO-EC card. Usually hanging up and calling back solves the problem, but it is still aggravating to the customer that has been called. Normally I'd suspect that something was only passing packets in one direction, but there is no firewall between the asterisk server and the phones and no iptables or anything like that running on the Asterisk server and sifting through sip debug logs to try to find one call out of maybe 50 has so far proven fruitless. Are there any common issues that might cause this? Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sporadic One Way Audio
I'm having an unusual problem at one of my branch offices. Every now and then they will make a call and the person they call is unable to hear them, but they are able to hear the person. The Asterisk server has only one ethernet interface and is on the same physical network as the 2 snom 300 phones and is connected to the PSTN lines with a Rhino R4FXO-EC card. Usually hanging up and calling back solves the problem, but it is still aggravating to the customer that has been called. Normally I'd suspect that something was only passing packets in one direction, but there is no firewall between the asterisk server and the phones and no iptables or anything like that running on the Asterisk server and sifting through sip debug logs to try to find one call out of maybe 50 has so far proven fruitless. Are there any common issues that might cause this? Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sporadic One Way Audio
How is your asterisk server connected to the PSTN? SIP/IAX out...ISDN/T1 out? Etc... Are you looking for lost RTP between * and internal phones or * and external provider? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson Sent: October 24, 2008 5:55 PM To: Asterisk Users List Subject: [asterisk-users] Sporadic One Way Audio I'm having an unusual problem at one of my branch offices. Every now and then they will make a call and the person they call is unable to hear them, but they are able to hear the person. The Asterisk server has only one ethernet interface and is on the same physical network as the 2 snom 300 phones and is connected to the PSTN lines with a Rhino R4FXO-EC card. Usually hanging up and calling back solves the problem, but it is still aggravating to the customer that has been called. Normally I'd suspect that something was only passing packets in one direction, but there is no firewall between the asterisk server and the phones and no iptables or anything like that running on the Asterisk server and sifting through sip debug logs to try to find one call out of maybe 50 has so far proven fruitless. Are there any common issues that might cause this? Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sporadic One Way Audio
The asterisk server is connected to the PSTN via a Rhino R4FXO-EC card. The lost RTP would have be between the Asterisk server and the phones. There are only 2 phones in the building, 2 lines coming in to the asterisk server and the server is on the same ethernet switch as the phones. The phones are SIP phones. This is a simple PBX system that picks up calls from the analog lines and routes them to the appropriate phone, although it will eventually be linked to a larger system once all the minor bugs are resolved. OCG Technical Support wrote: How is your asterisk server connected to the PSTN? SIP/IAX out...ISDN/T1 out? Etc... Are you looking for lost RTP between * and internal phones or * and external provider? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson Sent: October 24, 2008 5:55 PM To: Asterisk Users List Subject: [asterisk-users] Sporadic One Way Audio I'm having an unusual problem at one of my branch offices. Every now and then they will make a call and the person they call is unable to hear them, but they are able to hear the person. The Asterisk server has only one ethernet interface and is on the same physical network as the 2 snom 300 phones and is connected to the PSTN lines with a Rhino R4FXO-EC card. Usually hanging up and calling back solves the problem, but it is still aggravating to the customer that has been called. Normally I'd suspect that something was only passing packets in one direction, but there is no firewall between the asterisk server and the phones and no iptables or anything like that running on the Asterisk server and sifting through sip debug logs to try to find one call out of maybe 50 has so far proven fruitless. Are there any common issues that might cause this? Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sporadic One Way Audio
Well, if this is snom specific I can't offer more insight. It really sounds like misconfigured iptables and/or sip helper (conntrack/nat/etc). Are you sure your IP address is right in your sip.conf? If you don;t have NAT set to yes for these phones, they will trust the sip header for IP address and may misroute. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson Sent: October 24, 2008 7:36 PM To: Asterisk Users List Subject: Re: [asterisk-users] Sporadic One Way Audio Importance: High The asterisk server is connected to the PSTN via a Rhino R4FXO-EC card. The lost RTP would have be between the Asterisk server and the phones. There are only 2 phones in the building, 2 lines coming in to the asterisk server and the server is on the same ethernet switch as the phones. The phones are SIP phones. This is a simple PBX system that picks up calls from the analog lines and routes them to the appropriate phone, although it will eventually be linked to a larger system once all the minor bugs are resolved. OCG Technical Support wrote: How is your asterisk server connected to the PSTN? SIP/IAX out...ISDN/T1 out? Etc... Are you looking for lost RTP between * and internal phones or * and external provider? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson Sent: October 24, 2008 5:55 PM To: Asterisk Users List Subject: [asterisk-users] Sporadic One Way Audio I'm having an unusual problem at one of my branch offices. Every now and then they will make a call and the person they call is unable to hear them, but they are able to hear the person. The Asterisk server has only one ethernet interface and is on the same physical network as the 2 snom 300 phones and is connected to the PSTN lines with a Rhino R4FXO-EC card. Usually hanging up and calling back solves the problem, but it is still aggravating to the customer that has been called. Normally I'd suspect that something was only passing packets in one direction, but there is no firewall between the asterisk server and the phones and no iptables or anything like that running on the Asterisk server and sifting through sip debug logs to try to find one call out of maybe 50 has so far proven fruitless. Are there any common issues that might cause this? Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sporadic One Way Audio
We have seen cases where an IP address conflict caused something like this. You can take Wireshark traces on the PC (possibly run them in a loop so that you have a pretty long context) and if you have one-way audio be quick to log on to the web interface of the phone and also take a wireshark (PCAP) trace. There are a couple of tools available that may help to track such problems down: http://manageengine.adventnet.com/products/vqmanager, http://palladion.net, www.networkinstruments.de, and www.voipfuture.com. I know some of them offer a 14-days demo, and it tremendeously helped on of our clients to fix network problems. You can also use SNMP tools to poll if the phone has any blackouts regaring network availbility (see http://wiki.snom.com/SNMP). Also the phone sends a statistics at the end of each call. Check the BYE message, there is a counter of received and transmitted packets. Those numbers should be roughtly the same. CS -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Brent Davidson Gesendet: Freitag, 24. Oktober 2008 18:01 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] Sporadic One Way Audio I'm having an unusual problem at one of my branch offices. Every now and then they will make a call and the person they call is unable to hear them, but they are able to hear the person. The Asterisk server has only one ethernet interface and is on the same physical network as the 2 snom 300 phones and is connected to the PSTN lines with a Rhino R4FXO-EC card. Usually hanging up and calling back solves the problem, but it is still aggravating to the customer that has been called. Normally I'd suspect that something was only passing packets in one direction, but there is no firewall between the asterisk server and the phones and no iptables or anything like that running on the Asterisk server and sifting through sip debug logs to try to find one call out of maybe 50 has so far proven fruitless. Are there any common issues that might cause this? Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
