hello, i have trouble with nat + sip outgoing call.when make an outgoing call to a sip gateway, i have no sound. i have 2 sip gateway, one is asterisk. asterisk is on public ip and private ip other sip gateway is on public ip phone are cisco and grandstream on private ip on the same subnet as asterisk. phone are connected by sip to asterisk (i have try with or without nat=yes) incoming call from sip gateway to asterisk : no problem outgoing call from asterisk to sip gateway : no sound no sip error if i put a phone on public ip, incoming and outgoing call work fine i haven't make any port translation since phone and asterisk are on the same private subnet. phone register on the private ip of asterisk i have zaphfc connected to isdn and incoming and outgoing call work fine (with phone on public or private ip) any idea what's wrong or try to fix this problem ? thanks for help
Webn1 a ?crit :>hello, > >i have trouble with nat + sip outgoing call.when make an outgoing call to a >sip gateway, i have no sound. > >i have 2 sip gateway, one is asterisk. > >asterisk is on public ip and private ip >other sip gateway is on public ip >phone are cisco and grandstream on private ip on the same subnet as >asterisk. >phone are connected by sip to asterisk (i have try with or without nat=yes) > >incoming call from sip gateway to asterisk : no problem >outgoing call from asterisk to sip gateway : no sound > >no sip error >if i put a phone on public ip, incoming and outgoing call work fine > >i haven't make any port translation since phone and asterisk are on the same >private subnet. >phone register on the private ip of asterisk > >i have zaphfc connected to isdn and incoming and outgoing call work fine >(with phone on public or private ip) > >any idea what's wrong or try to fix this problem ? > >codec?>thanks for help > >-- Daniel
I have the following situation. My Asterisk Box is behind firewall ( for example 10.1.1.2 ) I have mapped 5060,10000-10010 and in rtp.conf I have said this range of prots 10000-10010. I'm tring to dial a PSTN from another PC with Sip phone in internet with external ip. I can hear the voice from the PSTN , but The Other Side can't hear me. I ran Ethereal and so that all rtp packets going from the calling phone are with destination 10.1.1.2. What to do to configured it right ? in Sip.conf [general] nat=yes externip=213.x.x.x [sipphone] [damencho] type=friend username=damencho host=dynamic nat=yes canreinvite=no
Hi; Using asterisk@home and it working well in network but when can not logged in over internet although the server is reachable Does anybody has any idea? Thanks Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi m?mk?n olmayabilir. Mesaji alan kisi, mesajin g?nderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekk?rler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050524/2f1c2243/attachment.htm
go in to the asterisk@home set up
http://yourip/maint
inter in your user name and password go to
config edit
sip.conf
go to the general section
Note: if you are behind a NAT Firewall, you will probably need to add the
following lines to hte [General] section of your sip.conf file. Adjust the
numbers as needed to match your configuration:
externip=66.5.21.6
replace the above with your public ip address
localnet=192.168.5.0/255.255.255.0
the 192 adddress mentioned above needs to be replaced with the ip address of
your asterisk server eg mine is on 192.168.15.101
so it would look like
localnet=192.168.15.101/255.255.255.0
hth
hank
email:
hanksmith4@earthlink.net
gmail:
hanksmith5@gmail.com
msn messenger:
hanksmith4@earthlink.net
aim:
hanksmith5
skype:
hanksmith5
----- Original Message -----
From: Bet?l G?zl?ko?lu
To: asterisk-users@lists.digium.com
Sent: Tuesday, May 24, 2005 3:57 AM
Subject: [Asterisk-Users] nat problem
Hi;
Using asterisk@home and it working well in network but when can not logged in
over internet although the server is reachable
Does anybody has any idea?
Thanks
Betul
Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun
kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli
olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi m?mk?n
olmayabilir. Mesaji alan kisi, mesajin g?nderilmek istendigi kisi veya kurulus
degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza
yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve
orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi
rica ederiz. Tesekk?rler - Hassangroup
Important note : This e-mail transmission is intended only for the use of the
individual or entity to which it is addressed, and may contain information that
is privileged, confidential and that may not be made public by law or agreement.
If the recipient of this message is not the intended recipient or entity, you
are hereby notified that any further dissemination, distribution or copying of
this information is strictly prohibited. If you have received this communication
in error, please notify us immediately by telephone and return the original
message to us to the above address or destroy it. Thank you - Hassangroup
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Hi, I am experiencing a hard to solve problem with my VoIP provider. I can make calls without any problem but I can not receive any. Actually, calls arive to * but the phone just does not ring. I believe must be a problem with NAT but I think I have a good config: - Extensions have nat=always and qualify=yes - Have introduced in sip.conf Externip and localnet - ADSL modem/router is redirected to my * server - With sip debug I can see the call arrives Am I misssing something that someone else can see? Appreciate any hint. Thanks ===================================ASTERISK VERSION: Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q SIP DEBUG CAPTURE <-- SIP read from 62.22.20.194:5060: INVITE sip:34700758288001@87.218.175.120:5060 SIP/2.0 Record-Route: <sip:62.22.20.194;ftag=08ff6000ff05ff10ff00000e0c4effff;lr> Via: SIP/2.0/UDP 62.22.20.194;branch=z9hG4bK90bf.b9c560e1.0 Via: SIP/2.0/UDP 62.22.20.207:5060;branch=z9hG4bK-10ff6000ff05ff10ff00000e0c4effff From: <sip:690351498@62.22.20.207;user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff To: <sip:34700758288001@62.22.20.194:5060;user=phone> Call-ID: d2c76000bf05c0108000000e0c4ef4b3@siphit-1 CSeq: 1 INVITE Contact: <sip:690351498@62.22.20.207;user=phone> Max-Forwards: 9 User-Agent: MERA MSIP v.1.0.2 Cisco-Guid: 908093991-393679323-3151091529-1429652222 Content-Type: application/sdp Content-Length: 216 v=0 o=- 1153435071 1153435071 IN IP4 62.22.20.207 s=- c=IN IP4 62.22.20.207 t=0 0 m=audio 59320 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (14 headers 10 lines)--- Using INVITE request as basis request - d2c76000bf05c0108000000e0c4ef4b3@siphit-1 Sending to 62.22.20.194 : 5060 (non-NAT) Found peer 'Peoplecall' Reliably Transmitting (NAT) to 62.22.20.194:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 62.22.20.194;branch=z9hG4bK90bf.b9c560e1.0;received=62.22.20.194 Via: SIP/2.0/UDP 62.22.20.207:5060;branch=z9hG4bK-10ff6000ff05ff10ff00000e0c4effff From: <sip:690351498@62.22.20.207;user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff To: <sip:34700758288001@62.22.20.194:5060;user=phone>;tag=as476d14de Call-ID: d2c76000bf05c0108000000e0c4ef4b3@siphit-1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:34700758288001@87.218.175.74> Proxy-Authenticate: Digest realm="asterisk", nonce="008d23b0" Content-Length: 0 --- Scheduling destruction of call 'd2c76000bf05c0108000000e0c4ef4b3@siphit-1' in 15000 ms asterisk1*CLI> <-- SIP read from 62.22.20.194:5060: ACK sip:34700758288001@87.218.175.120:5060 SIP/2.0 Via: SIP/2.0/UDP 62.22.20.194;branch=z9hG4bK90bf.b9c560e1.0 From: <sip:690351498@62.22.20.207;user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff Call-ID: d2c76000bf05c0108000000e0c4ef4b3@siphit-1 To: <sip:34700758288001@62.22.20.194:5060;user=phone>;tag=as476d14de CSeq: 1 ACK User-Agent: OpenSer (1.0.0 (i386/linux)) Content-Length: 0 --- (8 headers 0 lines)--- REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 62.22.20.194:5060: REGISTER sip:sip.peoplecall.com SIP/2.0 Via: SIP/2.0/UDP 87.218.175.74:5060;branch=z9hG4bK4a6abe4f;rport From: <sip:34700758288001@sip.peoplecall.com>;tag=as79fdfc26 To: <sip:34700758288001@sip.peoplecall.com> Call-ID: 1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1 CSeq: 421 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="34700758288001", realm="sip.peoplecall.com", algorithm=MD5, uri="sip:sip.peoplecall.com", nonce="44c0059db2d71f523aeb30399a54a4a32d8aeed6", response="ee782a37bae7eed1a0a881147c733ede", opaque="" Expires: 120 Contact: <sip:34700758288001@87.218.175.74> Event: registration Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 62.22.20.194:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK4a6abe4f;rport=5060 From: <sip:34700758288001@sip.peoplecall.com>;tag=as79fdfc26 To: <sip:34700758288001@sip.peoplecall.com>;tag=555271b30cfd40f8a3b4837b054360a3.975d Call-ID: 1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1 CSeq: 421 REGISTER Contact: <sip:34700758288001@192.168.1.104:5060>;expires=120 Server: OpenSer (1.0.0 (i386/linux)) Content-Length: 0 --- (9 headers 0 lines)--- Scheduling destruction of call '1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1' in 32000 ms Destroying call 'd2c76000bf05c0108000000e0c4ef4b3@siphit-1' asterisk1*CLI> sip no debug SIP Debugging Disabled -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060720/d1acd7ee/attachment.htm