[asterisk-users] One way audio problem
Hi, Asterisk is making a call to a peer. In 200 ok, peer is sending its application server ip in contact field, so asterisk sends ACK to that IP. RTP starts flowing between endpoints and peer plays an IVR and asks for destination number. After entering destination number peer's application server sends INVITE again with different media IP and asterisk accepts with 200 ok. RTP from peer comes from new media IP but asterisk keep sending to their old media IP that came in their 200 ok before and don't send to new one. Hence, I can hear their voice but they can't. Does anyone know how to make asterisk send RTP to new media IP that came in new INVITE within the call? Thanks Deepika -- _ -- 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] One way audio problem, a=sendonly and a re-invite
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, which I believe is correct if we have an a=sendonly. - Then the peer sends a 200 OK which also has a=sendonly, and then sends a re-invite which I've copied and pasted below. - We have canreinvite=no set in sip.conf, but I'm not sure if we should be rejecting this re-invite or not because it does contain a=sendrecv. If it should be rejected what error should Asterisk return, and how can we establish two way audio? - After this re-invite Asterisk replies with a 100 Trying and then a 200 OK which contains a=recvonly. - Call is established but called party cannot hear caller. Here's the re-invite message - note that Asterisk is on port 5070: U 2010/05/05 12:47:38.139701 (peer):5060 - (asterisk):5070 INVITE sip:(called number)@(asterisk):5070 SIP/2.0. Via: SIP/2.0/UDP (peer):5060;branch=z9hG4bK2sansay7330954rdb6594. To: User sip:(called number)@(asterisk):5070;tag=as3ddcc528. From: sip:(called number)@(peer):5060;tag=sansay7330954rdb6594. Call-ID: 58eb52aa414c5e465c3c1a15603093fb@(asterisk). CSeq: 2 INVITE. Contact: sip:(called number)@(peer):5060. Max-Forwards: 69. Content-Type: application/sdp. Content-Length: 297. . v=0. o=Sansay-VSXi 188 1 IN IP4 (peer). s=Session Controller. c=IN IP4 (other unknown IP, maybe of called number?). t=0 0. m=audio 6932 RTP/AVP 18 0 8 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=ptime:20. Any help would be much appreciated! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] One Way Audio Problem
On Thu, 16 Oct 2008, GNUbie wrote: Hello, On Tue, Oct 14, 2008 at 12:07 AM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: A packet trace will probably show exactly what is happening. Try: tcpdump -nlXs 8192 -i eth0 port 5060 You should be able to see the SIP information going back and forth and will probably show you that your NAT rules are applying when they shouldn't. I agree with first turning off your firewall and testing... but if that actually solves the problem you need to know why. This should tell why. Why eth0 when in fact it is not being used AFAIK? My Asterisk box is connected to the LAN via its eth1 interface and the SIP phone is calling from the LAN to the analog telephone via FXO/POTS. Again, below is the call scenario diagram: [SNOM] ==LAN== eth1 [ASTERISK] fxo ==POTS== [ANALOG_TELEPHONE] eth0 || INTERNET You should try on both interfaces. If you see packets on eth0 then your NAT rules are leaking! Try on eth1 to see the SIP headers and tell if your NAT rules are doing what you expect. This is always my first attack... j Please advice. Thank you in advance. Regards, GNUbie ___ -- 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 *** Handled by Will's new toy *** ___ -- 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] One Way Audio Problem
GNUbie wrote: What particular configs are you looking for? Below is my current setup and scenario: [snom] ==LAN== [asterisk] ==FXO/POTS == [analog_telephone/mobile_phone] SNOM is using the 192.168.101.102 IP address Asterisk is using 192.168.101.1 IP address for its eth1 interface FXO port is connected to the POTS SNOM doesn't need to go out to the Internet in this scenario, AFAIK. Below is my current NAT rules: # iptables -L -v -t nat Chain PREROUTING (policy ACCEPT 63795 packets, 7162K bytes) pkts bytes target prot opt in out source destination 11460 760K RETURN 0-- anyany 192.168.101.0/24 !192.168.101.0/24 Chain POSTROUTING (policy ACCEPT 570 packets, 41836 bytes) pkts bytes target prot opt in out source destination 11408 757K MASQUERADE 0-- anyeth0192.168.101.0/24 anywhere Chain OUTPUT (policy ACCEPT 570 packets, 41836 bytes) pkts bytes target prot opt in out source destination Please advice if you need more information from me. Regards, GNUbie Having had many years of experience working with iptables I can tell you that when IP Forwarding is enabled on a Linux machine things can get a bit tricky. In my experience using a Masquerade rule can cause some major weirdness. Try doing this: Instead of the Masquerade rule use: iptables -t nat -A POSTROUTING -i eth1 -o eth0 -j SNAT --to-source public ip of eth0 Also, in the general section of your sip.conf make sure you have: bindaddr=192.168.101.1 to make sure asterisk is not sending sip packets using the public IP then effectively trying to communicate with the phone by Masquerading the packets coming in over the eth1 to eth0. This is more than likely what is happening. (It's normlly bindaddr=0.0.0.0) Good luck, Brent ___ -- 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] One Way Audio Problem
On Thu, Oct 16, 2008 at 09:22:01AM +0800, GNUbie wrote: Hello Daniel, On Tue, Oct 14, 2008 at 12:12 AM, Daniel Hazelbaker [EMAIL PROTECTED] wrote: Might be a stretch, but does the Asterisk log show that the call was answered? I had this problem when interfacing * with an NEC system to do call parking pickup. The NEC would never give a dialtone (nor did it give answer supervision) so * never knew the call got picked up so audio only worked one way. I ended up rigging * to force the line to be considered answered with a patch. Yes, the call has been answered as per Asterisk logs. The CALLER (SNOM SIP Phone) can hear clearly the voice of the target CALLEE (POTS analog telephone) but it is the CALLEE that cannot hear the CALLER's voice. And yet in the output that you showed us, the channels were not in a state of Up. That is: not in a state of finished dialing and stuff and now part of a call. Could you plese double check that? What is the output of 'core show channels' at the time of a call? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] One Way Audio Problem
Hi, Am Donnerstag, den 16.10.2008, 09:37 +0800 schrieb GNUbie: Hello Karsten, On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote: Please post Your sip.conf. Which IP-Address do You configure in the snom for Your asterisk? (eth0 or eth1)? The SNOM 300 is using the NET interface beside the DC 5V port to connect to the LAN. The Asterisk box is using the eth1 to connect to the LAN. As per your instruction, below is my /etc/asterisk/sip.conf : - - - s n i p - - - [general] realm=pbx.domain.com bindport=5060 bindaddr=0.0.0.0 rtptimeout=60 disallow=all allow=ulaw allow=alaw allow=gsm externip=pbx.domain.com localnet=192.168.101.0/255.255.255.0 jbforce=yes allowtransfers=yes maxexpiry=3600 minexpiry=1800 videosupport=no [internal-phones](!) type=friend host=dynamic context=family dtmfmode=rfc2833 insecure=port,invite canreinvite=no nat=no qualify=yes port=5060 [102](internal-phones) username=102 secret=102 callerid=GNUbie102 [EMAIL PROTECTED] - - - s n i p - - - Thanks for the information. In an earlier post You told us, that the local phones talk to asterisk on eth1 using 192.168.101.0 network. Could You please double check, that the phone did not try to register on another IP? The asterisk is IIRC on a dual homed machine. Is Your phone using a DNS lookup to find the asterisk? To which address is that lookup resolved? Another hint: Is Your SNOM using some sort of STUN to lookup an public address? (Just to eliminate some things). HTH, Karsten ___ -- 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] One Way Audio Problem
Hello, On Tue, Oct 14, 2008 at 12:07 AM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: A packet trace will probably show exactly what is happening. Try: tcpdump -nlXs 8192 -i eth0 port 5060 You should be able to see the SIP information going back and forth and will probably show you that your NAT rules are applying when they shouldn't. I agree with first turning off your firewall and testing... but if that actually solves the problem you need to know why. This should tell why. Why eth0 when in fact it is not being used AFAIK? My Asterisk box is connected to the LAN via its eth1 interface and the SIP phone is calling from the LAN to the analog telephone via FXO/POTS. Again, below is the call scenario diagram: [SNOM] ==LAN== eth1 [ASTERISK] fxo ==POTS== [ANALOG_TELEPHONE] eth0 || INTERNET Please advice. Thank you in advance. Regards, GNUbie ___ -- 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] One Way Audio Problem
Hello Daniel, On Tue, Oct 14, 2008 at 12:12 AM, Daniel Hazelbaker [EMAIL PROTECTED] wrote: Might be a stretch, but does the Asterisk log show that the call was answered? I had this problem when interfacing * with an NEC system to do call parking pickup. The NEC would never give a dialtone (nor did it give answer supervision) so * never knew the call got picked up so audio only worked one way. I ended up rigging * to force the line to be considered answered with a patch. Yes, the call has been answered as per Asterisk logs. The CALLER (SNOM SIP Phone) can hear clearly the voice of the target CALLEE (POTS analog telephone) but it is the CALLEE that cannot hear the CALLER's voice. Regards, GNUbie ___ -- 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] One Way Audio Problem
Hello Karsten, On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote: Please post Your sip.conf. Which IP-Address do You configure in the snom for Your asterisk? (eth0 or eth1)? The SNOM 300 is using the NET interface beside the DC 5V port to connect to the LAN. The Asterisk box is using the eth1 to connect to the LAN. As per your instruction, below is my /etc/asterisk/sip.conf : - - - s n i p - - - [general] realm=pbx.domain.com bindport=5060 bindaddr=0.0.0.0 rtptimeout=60 disallow=all allow=ulaw allow=alaw allow=gsm externip=pbx.domain.com localnet=192.168.101.0/255.255.255.0 jbforce=yes allowtransfers=yes maxexpiry=3600 minexpiry=1800 videosupport=no [internal-phones](!) type=friend host=dynamic context=family dtmfmode=rfc2833 insecure=port,invite canreinvite=no nat=no qualify=yes port=5060 [102](internal-phones) username=102 secret=102 callerid=GNUbie102 [EMAIL PROTECTED] - - - s n i p - - - Thank you in advance. Regards, GNUbie ___ -- 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] One Way Audio Problem
Did you try it the magic number of times, three? On Sun, Oct 12, 2008 at 9:57 PM, GNUbie [EMAIL PROTECTED] wrote: Hello Tzafrir, On Mon, Oct 13, 2008 at 2:12 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: This means Zaptel gets silence from Asterisk. What codecs are used? What do you see on 'sip show channels'? I am using the following codecs: # asterisk -rx 'sip show settings' | grep Codecs Codecs: 0xe (gsm|ulaw|alaw) Below is the CLI output: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-081d11d0, Zap/4/1234567) in new stack -- Called 4/1234567 *CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 192.168.101.102 102 3c27a6824ba 00101/2 0x4 (ulaw) No Rx: INVITE 1 active SIP channel *CLI core show channels Channel Location State Application(Data) Zap/4-1 [EMAIL PROTECTED] Dialing AppDial((Outgoing Line)) SIP/102-081d11d0 [EMAIL PROTECTED]:1 RingDial(Zap/4/1234567) 2 active channels 1 active call Can you call from the FXO to Asterisk? (e.g.: to echo test) There is no problem with an inbound calls. I just tried to call the echo test extension number from my mobile phone via FXO/POTS and it works fine. I can hear my own voice. Thank you. Regards, GNUbie ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] One Way Audio Problem
Change all canreinvites to no. On Wed, Oct 15, 2008 at 9:37 PM, GNUbie [EMAIL PROTECTED] wrote: Hello Karsten, On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote: Please post Your sip.conf. Which IP-Address do You configure in the snom for Your asterisk? (eth0 or eth1)? The SNOM 300 is using the NET interface beside the DC 5V port to connect to the LAN. The Asterisk box is using the eth1 to connect to the LAN. As per your instruction, below is my /etc/asterisk/sip.conf : - - - s n i p - - - [general] realm=pbx.domain.com bindport=5060 bindaddr=0.0.0.0 rtptimeout=60 disallow=all allow=ulaw allow=alaw allow=gsm externip=pbx.domain.com localnet=192.168.101.0/255.255.255.0 jbforce=yes allowtransfers=yes maxexpiry=3600 minexpiry=1800 videosupport=no [internal-phones](!) type=friend host=dynamic context=family dtmfmode=rfc2833 insecure=port,invite canreinvite=no nat=no qualify=yes port=5060 [102](internal-phones) username=102 secret=102 callerid=GNUbie102 [EMAIL PROTECTED] - - - s n i p - - - Thank you in advance. Regards, GNUbie ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] One Way Audio Problem
canreinvite defaults to yes, whether specified or not. http://www.voip-info.org/wiki/view/tips If you follow these directions adapting to your particular circumstances and it doesn't work, post your whole sip.conf Start asterisk with verbose set to 3 or so and turn on sip debugging. I get somewhere in the debug, you will see local NAT IPs that don't belong there, or it will just work. Thanks, Steve Totaro On Thu, Oct 16, 2008 at 12:12 AM, Steve Totaro [EMAIL PROTECTED] wrote: Change all canreinvites to no. On Wed, Oct 15, 2008 at 9:37 PM, GNUbie [EMAIL PROTECTED] wrote: Hello Karsten, On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote: Please post Your sip.conf. Which IP-Address do You configure in the snom for Your asterisk? (eth0 or eth1)? The SNOM 300 is using the NET interface beside the DC 5V port to connect to the LAN. The Asterisk box is using the eth1 to connect to the LAN. As per your instruction, below is my /etc/asterisk/sip.conf : - - - s n i p - - - [general] realm=pbx.domain.com bindport=5060 bindaddr=0.0.0.0 rtptimeout=60 disallow=all allow=ulaw allow=alaw allow=gsm externip=pbx.domain.com localnet=192.168.101.0/255.255.255.0 jbforce=yes allowtransfers=yes maxexpiry=3600 minexpiry=1800 videosupport=no [internal-phones](!) type=friend host=dynamic context=family dtmfmode=rfc2833 insecure=port,invite canreinvite=no nat=no qualify=yes port=5060 [102](internal-phones) username=102 secret=102 callerid=GNUbie102 [EMAIL PROTECTED] - - - s n i p - - - Thank you in advance. Regards, GNUbie ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] One Way Audio Problem
Hello Steve, On Thu, Oct 16, 2008 at 12:04 PM, Steve Totaro [EMAIL PROTECTED] wrote: Did you try it the magic number of times, three? I'm sorry. What do you mean? Regards, GNUbie ___ -- 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] One Way Audio Problem
Maybe I have my threads confused but I thought you got one way audio when three calls were made, you only mentioned one call. On Thu, Oct 16, 2008 at 12:44 AM, GNUbie [EMAIL PROTECTED] wrote: Hello Steve, On Thu, Oct 16, 2008 at 12:04 PM, Steve Totaro [EMAIL PROTECTED] wrote: Did you try it the magic number of times, three? I'm sorry. What do you mean? Regards, GNUbie ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] One Way Audio Problem
Sorry, wrong thread, time for bed. I thought this was the thread where the guy was having issues with one way audio on his third call, and his Asterisk server was behind NAT. Good night everyone and have pleasant dreams of 700 point drops in the DOW! OT, did you know if the government took the $700+ billion dollars and did not bail out the greedy banks, we could have immediate relief since for the most part, we could suspend Federal Income tax for everyone. A $300 rebate check, give me a break, how about some real stimulus, a rebate (or lack of theft because there is no law that we as individuals have to pay Federal Income tax, and I dare anyone to point it out, a real law, not something the IRS made up, I don't think they are part of the Legislative branch) weekly or bi-weekly depending on how you get paid. It would be immediate and give more money to the people who need it. All your Fed Income tax pays for anyways is the national debt, the clock just maxed out at $10 trillion. Rather than paying it down below the max and keeping it that way, they are building another one with additional digits. Sorry for a TOTALLY OFF topic post. I screwed up so I thought I might as well rant a little. Apologies in sheer exhaustion, Steve Totaro Thanks, Steve Totaro On Thu, Oct 16, 2008 at 12:46 AM, Steve Totaro [EMAIL PROTECTED] wrote: Maybe I have my threads confused but I thought you got one way audio when three calls were made, you only mentioned one call. On Thu, Oct 16, 2008 at 12:44 AM, GNUbie [EMAIL PROTECTED] wrote: Hello Steve, On Thu, Oct 16, 2008 at 12:04 PM, Steve Totaro [EMAIL PROTECTED] wrote: Did you try it the magic number of times, three? I'm sorry. What do you mean? Regards, GNUbie ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] One Way Audio Problem
Hello Steve, On Thu, Oct 16, 2008 at 12:42 PM, Steve Totaro [EMAIL PROTECTED] wrote: canreinvite defaults to yes, whether specified or not. http://www.voip-info.org/wiki/view/tips If you follow these directions adapting to your particular circumstances and it doesn't work, post your whole sip.conf Start asterisk with verbose set to 3 or so and turn on sip debugging. I get somewhere in the debug, you will see local NAT IPs that don't belong there, or it will just work. My /etc/asterisk/sip.conf is at http://lists.digium.com/pipermail/asterisk-users/2008-October/220256.html and my SIP phone is located within the LAN where the Asterisk box is also part of it. Regards, GNUbie ___ -- 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] One Way Audio Problem
On Mon, Oct 13, 2008 at 09:57:33AM +0800, GNUbie wrote: Hello Tzafrir, On Mon, Oct 13, 2008 at 2:12 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: This means Zaptel gets silence from Asterisk. What codecs are used? What do you see on 'sip show channels'? I am using the following codecs: # asterisk -rx 'sip show settings' | grep Codecs Codecs: 0xe (gsm|ulaw|alaw) Below is the CLI output: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-081d11d0, Zap/4/1234567) in new stack -- Called 4/1234567 *CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 192.168.101.102 102 3c27a6824ba 00101/2 0x4 (ulaw) No Rx: INVITE 1 active SIP channel *CLI core show channels Channel Location State Application(Data) Zap/4-1 [EMAIL PROTECTED] Dialing AppDial((Outgoing Line)) SIP/102-081d11d0 [EMAIL PROTECTED]:1 RingDial(Zap/4/1234567) 2 active channels 1 active call So the call is not established yet, right? This is not a temporary state? Can you call from the FXO to Asterisk? (e.g.: to echo test) There is no problem with an inbound calls. I just tried to call the echo test extension number from my mobile phone via FXO/POTS and it works fine. I can hear my own voice. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] One Way Audio Problem
Hello Tzafrir, On Mon, Oct 13, 2008 at 3:33 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: So the call is not established yet, right? It is already. The CALLER hears the CALLEE's voice but the CALLEE cannot hear the CALLER's voices. This is not a temporary state? What do you mean? Regards, GNUbie ___ -- 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] One Way Audio Problem
On Mon, Oct 13, 2008 at 9:04 AM, GNUbie [EMAIL PROTECTED] wrote: Hello Tzafrir, On Mon, Oct 13, 2008 at 3:33 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: So the call is not established yet, right? It is already. The CALLER hears the CALLEE's voice but the CALLEE cannot hear the CALLER's voices. This is not a temporary state? What do you mean? Regards, GNUbie ___ -- 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 If you are going to dismiss (the most probable) problem (NAT) without posting configs, I am not sure how much help you will get, you will probably be dismissed as well. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] One Way Audio Problem
Hello Steve, On Mon, Oct 13, 2008 at 9:52 PM, Steve Totaro [EMAIL PROTECTED] wrote: If you are going to dismiss (the most probable) problem (NAT) without posting configs, I am not sure how much help you will get, you will probably be dismissed as well. What particular configs are you looking for? Below is my current setup and scenario: [snom] ==LAN== [asterisk] ==FXO/POTS == [analog_telephone/mobile_phone] SNOM is using the 192.168.101.102 IP address Asterisk is using 192.168.101.1 IP address for its eth1 interface FXO port is connected to the POTS SNOM doesn't need to go out to the Internet in this scenario, AFAIK. Below is my current NAT rules: # iptables -L -v -t nat Chain PREROUTING (policy ACCEPT 63795 packets, 7162K bytes) pkts bytes target prot opt in out source destination 11460 760K RETURN 0-- anyany 192.168.101.0/24 !192.168.101.0/24 Chain POSTROUTING (policy ACCEPT 570 packets, 41836 bytes) pkts bytes target prot opt in out source destination 11408 757K MASQUERADE 0-- anyeth0192.168.101.0/24 anywhere Chain OUTPUT (policy ACCEPT 570 packets, 41836 bytes) pkts bytes target prot opt in out source destination Please advice if you need more information from me. Regards, GNUbie ___ -- 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] One Way Audio Problem
On Mon, Oct 13, 2008 at 10:49 AM, GNUbie [EMAIL PROTECTED] wrote: Hello Steve, On Mon, Oct 13, 2008 at 9:52 PM, Steve Totaro [EMAIL PROTECTED] wrote: If you are going to dismiss (the most probable) problem (NAT) without posting configs, I am not sure how much help you will get, you will probably be dismissed as well. What particular configs are you looking for? Below is my current setup and scenario: [snom] ==LAN== [asterisk] ==FXO/POTS == [analog_telephone/mobile_phone] SNOM is using the 192.168.101.102 IP address Asterisk is using 192.168.101.1 IP address for its eth1 interface FXO port is connected to the POTS SNOM doesn't need to go out to the Internet in this scenario, AFAIK. Below is my current NAT rules: # iptables -L -v -t nat Chain PREROUTING (policy ACCEPT 63795 packets, 7162K bytes) pkts bytes target prot opt in out source destination 11460 760K RETURN 0-- anyany 192.168.101.0/24 !192.168.101.0/24 Chain POSTROUTING (policy ACCEPT 570 packets, 41836 bytes) pkts bytes target prot opt in out source destination 11408 757K MASQUERADE 0-- anyeth0192.168.101.0/24 anywhere Chain OUTPUT (policy ACCEPT 570 packets, 41836 bytes) pkts bytes target prot opt in out source destination Please advice if you need more information from me. Regards, GNUbie First, drop firewall/iptables/selinux and try again. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] One Way Audio Problem
On Oct 13, 2008, at 9:29 AM, [EMAIL PROTECTED] wrote: IME: One-way audio problems are almost always casued by NAT gateways and/or incorrect NAT settings in sip.conf and/or incorrect IP address or extenal proxy settings in the SIP phone. And reinvite issues in particular. Those have been the only one-way audio problems I've experienced. Setting reinvite=no fixed everything for me. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- 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] One Way Audio Problem
Hello Norman, On Mon, Oct 13, 2008 at 11:02 PM, Norman Franke [EMAIL PROTECTED] wrote: And reinvite issues in particular. Those have been the only one-way audio problems I've experienced. Setting reinvite=no fixed everything for me. You mean, canreinvite=no? I already have done line on my sip.conf. Thanks. Regards, GNUbie ___ -- 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] One Way Audio Problem
Hello Steve, On Mon, Oct 13, 2008 at 10:59 PM, Steve Totaro [EMAIL PROTECTED] wrote: First, drop firewall/iptables/selinux and try again. I already turned off the firewall and I don't have SELinux on my system and the problem is still there. Regards, GNUbie ___ -- 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] One Way Audio Problem
A packet trace will probably show exactly what is happening. Try: tcpdump -nlXs 8192 -i eth0 port 5060 You should be able to see the SIP information going back and forth and will probably show you that your NAT rules are applying when they shouldn't. I agree with first turning off your firewall and testing... but if that actually solves the problem you need to know why. This should tell why. j On Mon, 13 Oct 2008, GNUbie wrote: Hello Norman, On Mon, Oct 13, 2008 at 11:02 PM, Norman Franke [EMAIL PROTECTED] wrote: And reinvite issues in particular. Those have been the only one-way audio problems I've experienced. Setting reinvite=no fixed everything for me. You mean, canreinvite=no? I already have done line on my sip.conf. Thanks. Regards, GNUbie ___ -- 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 *** Handled by Will's new toy *** ___ -- 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] One Way Audio Problem
Might be a stretch, but does the Asterisk log show that the call was answered? I had this problem when interfacing * with an NEC system to do call parking pickup. The NEC would never give a dialtone (nor did it give answer supervision) so * never knew the call got picked up so audio only worked one way. I ended up rigging * to force the line to be considered answered with a patch. Daniel On Oct 13, 2008, at 8:57 AM, GNUbie wrote: Hello Steve, On Mon, Oct 13, 2008 at 10:59 PM, Steve Totaro [EMAIL PROTECTED] wrote: First, drop firewall/iptables/selinux and try again. I already turned off the firewall and I don't have SELinux on my system and the problem is still there. Regards, GNUbie ___ -- 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] One Way Audio Problem
Hi, Am Montag, den 13.10.2008, 10:00 +0800 schrieb GNUbie: Hello Gordon, On Mon, Oct 13, 2008 at 2:22 AM, Gordon Henderson [EMAIL PROTECTED] wrote: You mention the SIP phone being inside the LAN. Where is the Asterisk box? It is the main gateway of the IP phones and my laptop to the Internet. In this case, the eth1 of the Asterisk box is connected to the LAN and eth0 is connected to the Internet. IME: One-way audio problems are almost always casued by NAT gateways and/or incorrect NAT settings in sip.conf and/or incorrect IP address or extenal proxy settings in the SIP phone. I don't think NAT is involve on this one way audio problem. Please post Your sip.conf. Which IP-Address do You configure in the snom for Your asterisk? (eth0 or eth1)? Regards, Karsten ___ -- 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] One Way Audio Problem
Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio problem only happens when the SIP extension phone (let's call it the CALLER) places an outbound call to a mobile phone or analog telephone (let's call it the CALLEE) via FXO/POTS. The CALLER can hear the CALLEE's voice but the CALLEE cannot hear the CALLER's voice. I used this command ztmonitor 4 -vv -f /tmp/test.raw to monitor the RX/TX but the TX is totally zero. Below is a sample output of the ztmonitor command: - - - s n i p - - - # ztmonitor 4 -vv -f /tmp/test.raw Output to /tmp/test.raw Run e.g., 'sox -r 8000 -s -w -c 1 /tmp/test.raw /tmp/test.raw.wav' to convert. Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX (TX ###* Rx: 718 ( 718) Tx: 0 (0) - - - s n i p - - - Anyone can help me here? Thank you in advance. Regards, GNUbie ___ -- 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] One Way Audio Problem
On Sun, Oct 12, 2008 at 11:53:18PM +0800, GNUbie wrote: Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio problem only happens when the SIP extension phone (let's call it the CALLER) places an outbound call to a mobile phone or analog telephone (let's call it the CALLEE) via FXO/POTS. The CALLER can hear the CALLEE's voice but the CALLEE cannot hear the CALLER's voice. I used this command ztmonitor 4 -vv -f /tmp/test.raw to monitor the RX/TX but the TX is totally zero. Below is a sample output of the ztmonitor command: - - - s n i p - - - # ztmonitor 4 -vv -f /tmp/test.raw Output to /tmp/test.raw Run e.g., 'sox -r 8000 -s -w -c 1 /tmp/test.raw /tmp/test.raw.wav' to convert. Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX (TX ###* Rx: 718 ( 718) Tx: 0 (0) This means Zaptel gets silence from Asterisk. What codecs are used? What do you see on 'sip show channels'? Can you call from the FXO to Asterisk? (e.g.: to echo test) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] One Way Audio Problem
On Sun, 12 Oct 2008, GNUbie wrote: Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. You mention the SIP phone being inside the LAN. Where is the Asterisk box? IME: One-way audio problems are almost always casued by NAT gateways and/or incorrect NAT settings in sip.conf and/or incorrect IP address or extenal proxy settings in the SIP phone. Gordon ___ -- 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] One Way Audio Problem
Hello Tzafrir, On Mon, Oct 13, 2008 at 2:12 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: This means Zaptel gets silence from Asterisk. What codecs are used? What do you see on 'sip show channels'? I am using the following codecs: # asterisk -rx 'sip show settings' | grep Codecs Codecs: 0xe (gsm|ulaw|alaw) Below is the CLI output: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-081d11d0, Zap/4/1234567) in new stack -- Called 4/1234567 *CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 192.168.101.102 102 3c27a6824ba 00101/2 0x4 (ulaw) No Rx: INVITE 1 active SIP channel *CLI core show channels Channel Location State Application(Data) Zap/4-1 [EMAIL PROTECTED] Dialing AppDial((Outgoing Line)) SIP/102-081d11d0 [EMAIL PROTECTED]:1 RingDial(Zap/4/1234567) 2 active channels 1 active call Can you call from the FXO to Asterisk? (e.g.: to echo test) There is no problem with an inbound calls. I just tried to call the echo test extension number from my mobile phone via FXO/POTS and it works fine. I can hear my own voice. Thank you. Regards, GNUbie ___ -- 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] One Way Audio Problem
Hello Gordon, On Mon, Oct 13, 2008 at 2:22 AM, Gordon Henderson [EMAIL PROTECTED] wrote: You mention the SIP phone being inside the LAN. Where is the Asterisk box? It is the main gateway of the IP phones and my laptop to the Internet. In this case, the eth1 of the Asterisk box is connected to the LAN and eth0 is connected to the Internet. IME: One-way audio problems are almost always casued by NAT gateways and/or incorrect NAT settings in sip.conf and/or incorrect IP address or extenal proxy settings in the SIP phone. I don't think NAT is involve on this one way audio problem. Thank you. Regards, GNUbie ___ -- 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] One way audio problem on gateway to PSTN after some time, no NAT involved
Did you solved this Problem? I have the same problem, and i can't solve it, did you know anything about? Thanks Nico On Thu, 14 Sep 2006, Kai Militzer wrote: Hello everyone, since some weeks I experience strange problems on my gateways to the PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that SER -- Asterisk A -- Asterisk B (chan_ss7) -- PSTN What happens is, that after a while (uptime was a least two days) the gateway starts to not transmit audio to the PSTN on outgoing calls, but the caller can still hear the called party. There is no NAT involved and firewall rules allow the RTP ports defined in rtp.conf on both asterisk (A and B) machines. The SIP packages look good, no errors messages from asterisk or anything else, so I have really no idea what causes it and I cannot reproduce it except by waiting till it happens again. :( Now the strange thing is, that if I restart the asterisk all works fine again. A reload does not help, only a restart. Until now I came across this phenomenon two times on different machines and it all started about three weeks ago. Before that I ran asterisk 1.2.10 on the machines and then updated to 1.2.11. I looked through the Changelog but coulnd't find anything that seems related, but I guess it's a bug that was introduced somewhere between 1.2.10 and 1.2.11 ... Does anyone else have similar problems? Regards, Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller Lütticher Straße 10 Tel 0241/701333-14 [EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way audio problem on gateway to PSTN after some time, no NAT involved
Hi, I have similar symptoms (usually one-way audio like you, but sometimes echoed, distorded, or low volume sound), in a simpler configuration, using just SIP with a few phones and a TDM400 card with two FXOs: Asterisk -- PSTN I have kernel 2.6.18-XEN and using Asterisk 1.4 François. [EMAIL PROTECTED] wrote: Did you solved this Problem? I have the same problem, and i can't solve it, did you know anything about? Thanks Nico On Thu, 14 Sep 2006, Kai Militzer wrote: Hello everyone, since some weeks I experience strange problems on my gateways to the PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that SER -- Asterisk A -- Asterisk B (chan_ss7) -- PSTN What happens is, that after a while (uptime was a least two days) the gateway starts to not transmit audio to the PSTN on outgoing calls, but the caller can still hear the called party. There is no NAT involved and firewall rules allow the RTP ports defined in rtp.conf on both asterisk (A and B) machines. The SIP packages look good, no errors messages from asterisk or anything else, so I have really no idea what causes it and I cannot reproduce it except by waiting till it happens again. :( Now the strange thing is, that if I restart the asterisk all works fine again. A reload does not help, only a restart. Until now I came across this phenomenon two times on different machines and it all started about three weeks ago. Before that I ran asterisk 1.2.10 on the machines and then updated to 1.2.11. I looked through the Changelog but coulnd't find anything that seems related, but I guess it's a bug that was introduced somewhere between 1.2.10 and 1.2.11 ... Does anyone else have similar problems? Regards, Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller Lütticher Straße 10 Tel 0241/701333-14 [EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way audio problem on gateway to PSTN after some time, no NAT involved
Hello everyone, since some weeks I experience strange problems on my gateways to the PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that SER -- Asterisk A -- Asterisk B (chan_ss7) -- PSTN What happens is, that after a while (uptime was a least two days) the gateway starts to not transmit audio to the PSTN on outgoing calls, but the caller can still hear the called party. There is no NAT involved and firewall rules allow the RTP ports defined in rtp.conf on both asterisk (A and B) machines. The SIP packages look good, no errors messages from asterisk or anything else, so I have really no idea what causes it and I cannot reproduce it except by waiting till it happens again. :( Now the strange thing is, that if I restart the asterisk all works fine again. A reload does not help, only a restart. Until now I came across this phenomenon two times on different machines and it all started about three weeks ago. Before that I ran asterisk 1.2.10 on the machines and then updated to 1.2.11. I looked through the Changelog but coulnd't find anything that seems related, but I guess it's a bug that was introduced somewhere between 1.2.10 and 1.2.11 ... Does anyone else have similar problems? Regards, Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller Lütticher Straße 10 Tel 0241/701333-14 [EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way audio problem on gateway to PSTN after some time, no NAT involved
Hi Kai, we had a similar problem with a PBX which had PSTN lines and SIP phones: sometimes some phones had one way calls...the caller couldn't hear. We hadn't tried to restart but we reduced the number of RTP ports (rtp.conf if memory helps!) to a range of 200 (it depends from the number of simultaneous calls you have). That seemed to work! Hope it may help! Giorgio Incantalupo Kai Militzer wrote: Hello everyone, since some weeks I experience strange problems on my gateways to the PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that SER -- Asterisk A -- Asterisk B (chan_ss7) -- PSTN What happens is, that after a while (uptime was a least two days) the gateway starts to not transmit audio to the PSTN on outgoing calls, but the caller can still hear the called party. There is no NAT involved and firewall rules allow the RTP ports defined in rtp.conf on both asterisk (A and B) machines. The SIP packages look good, no errors messages from asterisk or anything else, so I have really no idea what causes it and I cannot reproduce it except by waiting till it happens again. :( Now the strange thing is, that if I restart the asterisk all works fine again. A reload does not help, only a restart. Until now I came across this phenomenon two times on different machines and it all started about three weeks ago. Before that I ran asterisk 1.2.10 on the machines and then updated to 1.2.11. I looked through the Changelog but coulnd't find anything that seems related, but I guess it's a bug that was introduced somewhere between 1.2.10 and 1.2.11 ... Does anyone else have similar problems? Regards, Kai ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
