Re: [Asterisk-Users] internal external SIP
On Thursday 08 July 2004 22:49, Ian D. Wlloughby wrote: I am guessing the problem is that your internal clients can see the external SIP clients but not the other way round. The clients have to be able to make a physical connection to each other. You are not using any NAT capabilities I guess as your internal clients have their own network to access the server on. If you set nat on in sip.conf for one of your internal clients and get it to register on the public network, does this work? Yes, the internal clients can see the external but not the other way round. I thought that canreinvite=no meant that the clients didn't need to be able to talk directly - just be registered on the same * box. Jon ___ 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] internal external SIP
On Thursday 08 July 2004 23:04, Soren Rathje wrote: bindaddr = 0.0.0.0 ; Local interface externip = xxx.xxx.xxx.xxx ; Public IP address localnet = 192.168.0.0/255.255.0.0 ; All RFC 1918 addresses are local networks localnet = 10.0.0.0/255.0.0.0; Also RFC1918 localnet = 172.16.0.0/12 ; Another RFC1918 with CIDR notation localnet = 169.254.0.0/255.255.0.0 ; Zero conf local network Also, I saw some fixes to RTP address binding in CVS today. Hard to tell really without a trace.. Okay, I've made some changes. I've moved the local phones to public IP's. So now everything is connecting effectively from the internet to the * box. Things are still the same as before - I can initiate calls from local phones to remote ones. If a remote phone tries to initiate the call, the internal phone rings. When I pickup the internal phone, the call isn't completed. I've included a trace below of an incomming call. I don't know which bits are relevant so I've pasted it all. Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.145.37.29;branch=z9hG4bKd27e2ee695c179d0 From: Pete Murphy sip:[EMAIL PROTECTED];tag=2f9ab5f04daa9e4e To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 7711 INVITE User-Agent: Grandstream SIP UA 1.0.4.26 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 270 v=0 o=2003 8000 8000 IN IP4 82.145.37.29 s=SIP Call c=IN IP4 82.145.37.29 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=ptime:20 12 headers, 13 lines Using latest request as basis request Sending to 82.145.37.29 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format ULAW Found audio format UNKN Found audio format GSM Found audio format UNKN Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format G726-32 Found description format G728 Capabilities: us - 524302, them - 285/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 82.145.37.29;branch=z9hG4bKd27e2ee695c179d0 From: Pete Murphy sip:[EMAIL PROTECTED];tag=2f9ab5f04daa9e4e To: sip:[EMAIL PROTECTED];tag=as584623c0 Call-ID: [EMAIL PROTECTED] CSeq: 7711 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=7c6b65eb Content-Length: 0 to 82.145.37.29:5060 Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.145.37.29;branch=z9hG4bKd27e2ee695c179d0 From: Pete Murphy sip:[EMAIL PROTECTED];tag=2f9ab5f04daa9e4e To: sip:[EMAIL PROTECTED];tag=as584623c0 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 7711 ACK User-Agent: Grandstream SIP UA 1.0.4.26 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Length: 0 11 headers, 0 lines Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.145.37.29;branch=z9hG4bKd04bc72a0d281db8 From: Pete Murphy sip:[EMAIL PROTECTED];tag=2f9ab5f04daa9e4e To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Proxy-Authorization: DIGEST username=2003, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=7c6b65eb, response=2d2400a30b257419c48ac5dd6747 Call-ID: [EMAIL PROTECTED] CSeq: 7712 INVITE User-Agent: Grandstream SIP UA 1.0.4.26 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 270 v=0 o=2003 8000 8000 IN IP4 82.145.37.29 s=SIP Call c=IN IP4 82.145.37.29 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=ptime:20 13 headers, 13 lines Using latest request as basis request Sending to 82.145.37.29 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format ULAW Found audio format UNKN Found audio format GSM Found audio format UNKN Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format G726-32 Found description format G728 Capabilities: us - 524302, them - 285/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Looking for 2000 in remote list_route: hop: sip:[EMAIL PROTECTED] Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 82.145.37.29;branch=z9hG4bKd04bc72a0d281db8 From: Pete Murphy sip:[EMAIL PROTECTED];tag=2f9ab5f04daa9e4e To: sip:[EMAIL PROTECTED];tag=as17a6c60a Call-ID: [EMAIL PROTECTED] CSeq: 7712 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL,
Re: [Asterisk-Users] internal external SIP
From: Jon Lawrence Okay, I've made some changes. I've moved the local phones to public IP's. So now everything is connecting effectively from the internet to the * box. Things are still the same as before - I can initiate calls from local phones to remote ones. If a remote phone tries to initiate the call, the internal phone rings. When I pickup the internal phone, the call isn't completed. .. snip .. to 82.145.37.29:5060 Jul 9 12:41:49 WARNING[5126]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 7712 (Response) set_destination: Parsing sip:[EMAIL PROTECTED] for address/port to send to set_destination: set destination to 81.168.4.69, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 81.168.4.67:5060;branch=z9hG4bK201b0b71 From: 2003 sip:[EMAIL PROTECTED];tag=as3f8ccbff To: sip:[EMAIL PROTECTED];tag=0939785f3bc7641e Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 What are your codec settings in sip.conf ?? Could you try (can be set at client level): disallow=all allow=ulaw -- Soren ___ 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] internal external SIP
On Friday 09 July 2004 15:30, Soren Rathje wrote: What are your codec settings in sip.conf ?? Could you try (can be set at client level): disallow=all allow=ulaw codec's are set to allow all. I can't see how this would help. I can talk fine from local client to remote so the codecs must be correct. Jon ___ 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] internal external SIP
On Friday 09 July 2004 15:30, Soren Rathje wrote: What are your codec settings in sip.conf ?? Could you try (can be set at client level): disallow=all allow=ulaw codec's are set to allow all. I can't see how this would help. I can talk fine from local client to remote so the codecs must be correct. Ok, then I suggest you have a look at http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+grandstream+budgetone -- Soren ___ 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] internal external SIP
[EMAIL PROTECTED] (Jon Lawrence) writes: codec's are set to allow all. Thats your problem. I tried this too as an experiment and asterisk appears to take all to mean all codecs you can think of, not just the ones you have converters for. Instead of all you may want to try listing the codecs asterisk actually has (this is from -current): ; ; codecs: a_mu adpcm alaw g726 gsm ilbc lpc10 ulaw ; disallow=all allow=ulaw allow=alaw allow=gsm allow=adpcm allow=g726 allow=ilbc ;; allow=lpc10 (robotman) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] internal external SIP
On Friday 09 July 2004 18:42, Wolfgang S. Rupprecht wrote: [EMAIL PROTECTED] (Jon Lawrence) writes: codec's are set to allow all. Thats your problem. No it's not. I'm not saying that it won't fix it - it might. I've just put my local phone back on the internal network, moved the remote phone onto a vlan that have a ipsec vpn to my internal network - guess what, everything worked. If the problem was down to me having all codec's allowed then this should not have worked - at least I don't think it should have :) I tried this too as an experiment and asterisk appears to take all to mean all codecs you can think of, not just the ones you have converters for. Instead of all you may want to try listing the codecs asterisk actually has (this is from -current): ; ; codecs: a_mu adpcm alaw g726 gsm ilbc lpc10 ulaw ; disallow=all allow=ulaw allow=alaw allow=gsm allow=adpcm allow=g726 allow=ilbc ;; allow=lpc10 (robotman) I'll try this any way - since it's something I've not tried. If this does cure my problems, I'll be throughly confused. Jon ___ 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] internal external SIP
In response to myself. Setting specific codecs has indeed fixed the problem. Q - how and why ? when the remote phone has a vpn directly to my internal LAN, everything works perfectly when codecs=all. But when it's connecting in from a public IP everything goes pear shaped. Can anyone give a even a clue as to why this happens ? Or is it like many other things that don't make sense - it just does :) Thanks to everyone that offered advise. Jon ___ 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] internal external SIP
Hi all, I've got a problem with external sip clients. My * box has 2 nics, one to my internal network and one on a public IP. There are external sip clients (on public IPs) and internal clients on the internal nic. both clients can register fine. I can phone external clients from the internal clients and the connection works perfectly. But if an external client phones an internal one, the internal phone rings, but when the phone is picked up the external call disappears. Both internal and external have canreinvite=no Can anyone give me any ideas where to start looking into this. Regards, Jon ___ 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] internal external SIP
I am guessing the problem is that your internal clients can see the external SIP clients but not the other way round. The clients have to be able to make a physical connection to each other. You are not using any NAT capabilities I guess as your internal clients have their own network to access the server on. If you set nat on in sip.conf for one of your internal clients and get it to register on the public network, does this work? R's Ian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Lawrence Sent: 08 July 2004 21:00 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] internal external SIP Hi all, I've got a problem with external sip clients. My * box has 2 nics, one to my internal network and one on a public IP. There are external sip clients (on public IPs) and internal clients on the internal nic. both clients can register fine. I can phone external clients from the internal clients and the connection works perfectly. But if an external client phones an internal one, the internal phone rings, but when the phone is picked up the external call disappears. Both internal and external have canreinvite=no Can anyone give me any ideas where to start looking into this. Regards, Jon ___ 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] internal external SIP
Hi all, I've got a problem with external sip clients. My * box has 2 nics, one to my internal network and one on a public IP. There are external sip clients (on public IPs) and internal clients on the internal nic. both clients can register fine. I can phone external clients from the internal clients and the connection works perfectly. But if an external client phones an internal one, the internal phone rings, but when the phone is picked up the external call disappears. Both internal and external have canreinvite=no Can anyone give me any ideas where to start looking into this. bindaddr = 0.0.0.0 ; Local interface externip = xxx.xxx.xxx.xxx ; Public IP address localnet = 192.168.0.0/255.255.0.0 ; All RFC 1918 addresses are local networks localnet = 10.0.0.0/255.0.0.0; Also RFC1918 localnet = 172.16.0.0/12 ; Another RFC1918 with CIDR notation localnet = 169.254.0.0/255.255.0.0 ; Zero conf local network Also, I saw some fixes to RTP address binding in CVS today. Hard to tell really without a trace.. -- Soren ___ 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
