Re: [Asterisk-Users] Cisco 7940 + no audio after MOH
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wed, 24 Aug 2005, Julien wrote: Hi, I use * release 1.0.9 with differents phones and softphone, i've got a problem with my Cisco 7940G (last SIP Firmware). Sometimes, when i but a call on hold, the caller has got the music, but when i resume the call, then the caller does not hear me (and nothing at all)... I must wait for 10, 20, sometimes 60 seconds before he could hear me again. Any body already had this problem ? I use G711u codec for the cisco and i had the probem with a IAX client (using GSM) a PSTN caller (through my SPA3K using 711u) ... Thanks a lot ;) Julien. Is the 794G configured to use a VLAN ? I had a similar problem with VLAN and ARP packets. I got around this by not using VLANS but putting the phone on a separate subnet (with QoS on the phone subnet). - -- Ron Wellsted http://www.wellsted.org.uk [EMAIL PROTECTED] FWD: 519961 N 52.567623, W 2.137621 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iQEVAwUBQwxK/ktP/KMNOfRbAQLSWAf/XhXZdD60KukuTT0B59u6iefG0m94q2nm SNqoZ+9PHrvYmVCRVC4zlH1JQUyMi7w4oed4JHqTAgK0XATEOFXbvjJuiZkNaWnC hl3iCSD3PiPjGkdusvOy/myQzZjCrAZ1DVVSA9Jn0bjtPF9CrEr4zfo9vkQz1ZV9 +mpMJgZPMV6lZ8CTR2ylS6mD/QnqP1sxHPZu2Fyr8ILFpH7gzyCeVg8LT4+IkDNa Es5o4UxQm5lTTnlvZ562dho/GZXflm2FH5CT1lRecmFD3qlulmqC1Ih6kTxWkEZ7 +QLaZrTBiXV4dkF2xtr9NhmvtozpsqvQtBnTiQR/q3Y7uv165pgVFw== =vXem -END PGP SIGNATURE- ___ 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] Cisco 7940 + no audio after MOH
Hi, I use * release 1.0.9 with differents phones and softphone, i've got a problem with my Cisco 7940G (last SIP Firmware). Sometimes, when i but a call on hold, the caller has got the music, but when i resume the call, then the caller does not hear me (and nothing at all)... I must wait for 10, 20, sometimes 60 seconds before he could hear me again. Any body already had this problem ? I use G711u codec for the cisco and i had the probem with a IAX client (using GSM) a PSTN caller (through my SPA3K using 711u) ... Thanks a lot ;) Julien. ___ 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
Re: [Asterisk-Users] Cisco 7940 Outgoing Audio
On Wed, Apr 06, 2005 at 02:27:56PM -0600, Bellows, Jared arranged a set of bits into the following: I'm a Cisco 7940 phone using SCCP. My setup is a private network with the Wow, first time I think we've ever had a *phone* post to the list ;-) * box acting as dhcp server and also tftp server. The phone loads and dials out fine. I can hear the other person, but there is no outgoing audio. I've read that this is an RTP problem and have tried making some changes in /etc/hosts to point to my * box IP but with no luck. When I do a tcpdump I see that the RTP packets are sent to 0.0.0.0. How do I get the phone to send to the * box? If you're using chan_sccp update it to the latest CVS HEAD, that fixes the RTP issues. Thanks, Julien chan_sccp project lead pgp78AbjCsaDM.pgp Description: PGP signature ___ 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
RE: [Asterisk-Users] Cisco 7940 no audio
Title: Message Thanks for the help, I am currently running the latest sip image, it seems to have fixed a lot of bugs.. I did a full rebuild of the server and used the stable cvs, all is working perfectly now. I am actually amazed at the quality of the call using the diva card/capi through isdn. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of AstGrp Sent: 18 April 2004 04:44 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7940 no audio Try upgrading to SIP 6.3. I heard from someone on the IRC Channel about this problem and 6.3 resolved it -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Waddington Posted At: Friday, April 16, 2004 1:04 PM Posted To: Asterisk User Group Conversation: Cisco 7940 no audio Subject: [Asterisk-Users] Cisco 7940 no audio When we receive or make a call to the outside they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi - Asterisk network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this problem, and has a fix? Thanks.
RE: [Asterisk-Users] Cisco 7940 no audio
Title: Message Try upgrading to SIP 6.3. I heard from someone on the IRC Channel about this problem and 6.3 resolved it -gcc -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig WaddingtonPosted At: Friday, April 16, 2004 1:04 PMPosted To: Asterisk User GroupConversation: Cisco 7940 no audioSubject: [Asterisk-Users] Cisco 7940 no audio When we receive or make a call to the outside they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi - Asterisk network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this problem, and has a fix? Thanks.
[Asterisk-Users] Cisco 7940 no audio
When we receive or make a call to the outside they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi - Asterisk network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this problem, and has a fix? Thanks.
Re: [Asterisk-Users] Cisco 7940 no audio
On Apr 16, 2004, at 10:04 AM, Craig Waddington wrote: When we receive or make a call to the outside they can hear us, but we cant hear them. With SIP, missing audio is *usually* either a firewall or NAT issue. Check firewall logs and make sure that you aren't seeing packets being lost. Do you have more then one 7940? If so, can they call each other? Also, when people call into your system, do they get audio from asterisk? Does voicemail work? Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 no audio
It might be worth checking if there's a firewall in the way blocking your upstream audio and also the codecs setup on the cisco and in your asterisk box ?? There's not a lot of detail here to go on, sorry. On Fri, 16 Apr 2004, Craig Waddington wrote: When we receive or make a call to the outside - they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi - Asterisk -- network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this problem, and has a fix? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 no audio
Craig Waddington wrote: When we receive or make a call to the outside they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi - Asterisk network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this problem, and has a fix? Thanks. Make sure you don't have the Cisco phone set to do NAT. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7940 no audio
I will try disallow=all, thanks, Nat is off. Sip.conf below. If I call from my mobile to the isdn number, the Cisco phone rings, I pick up, they can hear me, I can not hear them, if I transfer it to the Sipura I get AUDIO!! It is also happening over IAX with the Cisco phones. I followed a lot of the examples on loligo.com, which were a great help, but this is so hard to troubleshoot as I cannot see any errors in debug, asterisk thinks a good call is in progress. Anything internal is perfect. The CAPI works fine. Its just the audio from the other end. Every now and then I can hear a quick bit of sound. One in 20 calls may work. [general] port=5060 ; Port to bind to bindaddr=0.0.0.0; Address to bind to allow=ulaw allow=alaw tos=lowdelay [20] type=friend username=20 secret=20 canreinvite=no host=dynamic mailbox=20 callerid=Cisco Phone 20 accountcode=20 qualify=yes context=sip Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Cuthie Sent: 16 April 2004 18:37 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 no audio Craig Waddington wrote: When we receive or make a call to the outside - they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi - Asterisk à network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this problem, and has a fix? Thanks. Make sure you don't have the Cisco phone set to do NAT. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7940 no audio
Yes MOH etc work fine for the receiving end, dialing from outside. I have run X-lite and GS phones on the network on a test machine before this one, and it worked great. Though I haven't had a chance to see if they work or not. I will definatley check my Firewall logs, that's a good point, but the sipura works. It seems codec to me, but I have tried many different confs in sip. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Laird Sent: 16 April 2004 18:32 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 no audio On Apr 16, 2004, at 10:04 AM, Craig Waddington wrote: When we receive or make a call to the outside - they can hear us, but we cant hear them. With SIP, missing audio is *usually* either a firewall or NAT issue. Check firewall logs and make sure that you aren't seeing packets being lost. Do you have more then one 7940? If so, can they call each other? Also, when people call into your system, do they get audio from asterisk? Does voicemail work? Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 no audio
On Fri, Apr 16, 2004 at 06:04:04PM +0100, Craig Waddington spake thusly: When we receive or make a call to the outside - they can hear us, but we cant hear them. I have had this problem several times and so far no resolution. However for me it has always been with IAX. I have been told that IAX is supposed to be NAT-safe but that does not seem to be the case for me. For example: SIP (grandstream, snom) -Asterisk-NAT-Asterisk-SIP (grandstream, snom) He can hear me but I can't hear him. In another case I had: IAXclient (soft phone)-NAT-Asterisk-Snom And I could hear him but he could not hear me. Same phone system and settings as above. However as soon as I switched the first users phone to talk directly to my Asterisk box with SIP it worked perfectly. And when I switched the user in the second case to a SIP based soft phone it also worked just fine. SIP has worked better through NAT than IAX (with nat=yes in sip.conf) which is bizarre and contrary to what I have read where IAX should be NAT-safe and SIP not. I have dreams of a world fully converted to IPv6 where NAT no longer exists. Alas, it is but a dream. -- Tracy Reed The attachment is a digital signature. http://copilotconsulting.com More info: http://copilotconsulting.com/sig pgp0.pgp Description: PGP signature
RE: [Asterisk-Users] Cisco 7940 no audio - sip debug
Of Tracy R Reed Sent: 16 April 2004 19:20 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 no audio On Fri, Apr 16, 2004 at 06:04:04PM +0100, Craig Waddington spake thusly: When we receive or make a call to the outside - they can hear us, but we cant hear them. I have had this problem several times and so far no resolution. However for me it has always been with IAX. I have been told that IAX is supposed to be NAT-safe but that does not seem to be the case for me. For example: SIP (grandstream, snom) -Asterisk-NAT-Asterisk-SIP (grandstream, snom) He can hear me but I can't hear him. In another case I had: IAXclient (soft phone)-NAT-Asterisk-Snom And I could hear him but he could not hear me. Same phone system and settings as above. However as soon as I switched the first users phone to talk directly to my Asterisk box with SIP it worked perfectly. And when I switched the user in the second case to a SIP based soft phone it also worked just fine. SIP has worked better through NAT than IAX (with nat=yes in sip.conf) which is bizarre and contrary to what I have read where IAX should be NAT-safe and SIP not. I have dreams of a world fully converted to IPv6 where NAT no longer exists. Alas, it is but a dream. -- Tracy Reed The attachment is a digital signature. http://copilotconsulting.com More info: http://copilotconsulting.com/sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 no audio
Craig Waddington wrote: I will try disallow=all, thanks, Nat is off. Sip.conf below. If I call from my mobile to the isdn number, the Cisco phone rings, I pick up, they can hear me, I can not hear them, if I transfer it to the Sipura I get AUDIO!! It is also happening over IAX with the Cisco phones. I followed a lot of the examples on loligo.com, which were a great help, but this is so hard to troubleshoot as I cannot see any errors in debug, asterisk thinks a good call is in progress. Anything internal is perfect. The CAPI works fine. Its just the audio from the other end. Every now and then I can hear a quick bit of sound. One in 20 calls may work. [general] port=5060 ; Port to bind to bindaddr=0.0.0.0; Address to bind to allow=ulaw allow=alaw tos=lowdelay [20] type=friend username=20 secret=20 canreinvite=no host=dynamic mailbox=20 callerid=Cisco Phone 20 accountcode=20 qualify=yes context=sip Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Cuthie Sent: 16 April 2004 18:37 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 no audio Craig Waddington wrote: When we receive or make a call to the outside - they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi - Asterisk à network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this problem, and has a fix? Thanks. Make sure you don't have the Cisco phone set to do NAT. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Just to be clear, you need at least the following (or at least I did): sip.conf: nat=yes reinvite=no SIPDefault.conf (in your tftp directory) nat_enable=0 -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
