Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine; However, when the _remote_ XLite calls the local XLite, things work OK until precisely 20s, where Asterisk decides to hang up, and displays the following error message in the console: =================WARNING[593]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on transmission e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. for seqno 2 (Critical Response) WARNING[593]: chan_sip.c:1972 retrans_pkt: Hanging up call e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. - no reply to our critical packet. == Spawn extension (my-phones, local-xlite-extension, 1) exited non-zero on 'SIP/unused-008008e4' ================= I'm no SIP expert, but based on the debug session, before deciding to hang up, Asterisk tries 6 times to send an OK message to the remote XLite, and doesn't seem to get an answer. FWIW, after Asterisk has hung up, the remote XLite remains off-hook, oblivious to this error and keeps displaying "Call established": www.pastebin.com/x6MgnrpG There's also this oddity on line 50: "Scheduling destruction of SIP dialog". FWIW, in sip.conf, for the remote XLite user, I tried "nat=no" and "nat=yes", with no difference. I'm actually not sure how to configure a remote user which happens to be listed in sip.conf (it's behind a NAT router but it registers with Asterisk, so... is it NATed or not?), and am surprised it actually rings and sends/receives voice with no problem, regardless of this parameter. I found discussions about using "t1min=500" in sip.conf, but it made no difference either. Has someone already experienced this and knows what can be done? Any hint much appreciated.
On 22 December 2010 12:44, Gilles <codecomplete at free.fr> wrote:> Hello > > ? ? ? ?I have an Asterisk 1.4 server and two XLite softphones, where > Asterisk and the local XLite phone are located in a LAN behind a NAT > router, and the remote XLite phone is located elsewhere on the Net > behind its own NAT router: > > http://img252.imageshack.us/img252/3749/asterisknat.png > > I'm having the following issue: When the _local_ XLite calls out the > remote XLite, everything works fine; However, when the _remote_ XLite > calls the local XLite, things work OK until precisely 20s, where > Asterisk decides to hang up, and displays the following error message > in the console: > > =================> WARNING[593]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on > transmission > e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. for seqno > 2 (Critical Response) > > WARNING[593]: chan_sip.c:1972 retrans_pkt: Hanging up call > e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. - no > reply to our critical packet. > ?== Spawn extension (my-phones, local-xlite-extension, 1) exited > non-zero on 'SIP/unused-008008e4' > =================> > I'm no SIP expert, but based on the debug session, before deciding to > hang up, Asterisk tries 6 times to send an OK message to the remote > XLite, and doesn't seem to get an answer. FWIW, after Asterisk has > hung up, the remote XLite remains off-hook, oblivious to this error > and keeps displaying "Call established": > > www.pastebin.com/x6MgnrpG > > There's also this oddity on line 50: "Scheduling destruction of SIP > dialog". > > FWIW, in sip.conf, for the remote XLite user, I tried "nat=no" and > "nat=yes", with no difference. I'm actually not sure how to configure > a remote user which happens to be listed in sip.conf (it's behind a > NAT router but it registers with Asterisk, so... is it NATed or not?), > and am surprised it actually rings and sends/receives voice with no > problem, regardless of this parameter. > > I found discussions about using "t1min=500" in sip.conf, but it made > no difference either. > > Has someone already experienced this and knows what can be done? > > Any hint much appreciated.Look in the XLite advanced network settings and disable the 2 timeout settings (RTP and RTCP?). This is not always necessary, but there are sufficient cases where the packets XLite expects appear early on, but do not persist, thus causing a hangup. I think the default timeout is 20 seconds. Cheers, Steve
Hello, you have a typicall nat issue. Asterisk receives messages from the phone but cannot send any messages back (thats why it tries to resend the 200 ok message 6 times). try setting qualify=yes to your sip peers config to keep the nat port open. best regards stefan Am 22.12.10 13:44, schrieb Gilles:> Hello > > I have an Asterisk 1.4 server and two XLite softphones, where > Asterisk and the local XLite phone are located in a LAN behind a NAT > router, and the remote XLite phone is located elsewhere on the Net > behind its own NAT router: > > http://img252.imageshack.us/img252/3749/asterisknat.png > > I'm having the following issue: When the _local_ XLite calls out the > remote XLite, everything works fine; However, when the _remote_ XLite > calls the local XLite, things work OK until precisely 20s, where > Asterisk decides to hang up, and displays the following error message > in the console: > > =================> WARNING[593]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on > transmission > e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. for seqno > 2 (Critical Response) > > WARNING[593]: chan_sip.c:1972 retrans_pkt: Hanging up call > e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. - no > reply to our critical packet. > == Spawn extension (my-phones, local-xlite-extension, 1) exited > non-zero on 'SIP/unused-008008e4' > =================> > I'm no SIP expert, but based on the debug session, before deciding to > hang up, Asterisk tries 6 times to send an OK message to the remote > XLite, and doesn't seem to get an answer. FWIW, after Asterisk has > hung up, the remote XLite remains off-hook, oblivious to this error > and keeps displaying "Call established": > > www.pastebin.com/x6MgnrpG > > There's also this oddity on line 50: "Scheduling destruction of SIP > dialog". > > FWIW, in sip.conf, for the remote XLite user, I tried "nat=no" and > "nat=yes", with no difference. I'm actually not sure how to configure > a remote user which happens to be listed in sip.conf (it's behind a > NAT router but it registers with Asterisk, so... is it NATed or not?), > and am surprised it actually rings and sends/receives voice with no > problem, regardless of this parameter. > > I found discussions about using "t1min=500" in sip.conf, but it made > no difference either. > > Has someone already experienced this and knows what can be done? > > Any hint much appreciated. > > > -- > _____________________________________________________________________ > -- 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-- F?r weitere Fragen stehen wir gerne unter voip at sil.at oder 059944 - 2440 zur Verf?gung. Mit freundlichen Gr?ssen -- Stefan Schmidt Sysadmin/VOIP // voip at sil.at // Tel 059944-2440// ------------------------------------------------- SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // -------------------------------------------------
On Wed, 22 Dec 2010 13:18:38 +0000, Steve Davies <davies147 at gmail.com> wrote:>Look in the XLite advanced network settings and disable the 2 timeout >settings (RTP and RTCP?). This is not always necessary, but there are >sufficient cases where the packets XLite expects appear early on, but >do not persist, thus causing a hangup. I think the default timeout is >20 seconds.Thanks for the tip, but I get the same problem with SJPhone and PhonerLite, so it looks like a problem in Asterisk.
This is a NAT issue like noted before. Try: localnet=192.168.0.0/ <http://192.168.0.0/24>255.255.255.0 instead of: localnet=192.168.0.0/24 <http://192.168.0.0/24>Also, make sure you have all your VPN connections as localnet and other side subnet as localnet as well if you are using VPN. Otherwise, open the neccessary ports needed for SIP and RTP. If you note your router type someone might be able to help more specifically. -Bruce On Wed, Dec 22, 2010 at 12:27 PM, Gilles <codecomplete at free.fr> wrote:> On Wed, 22 Dec 2010 13:18:38 +0000, Steve Davies <davies147 at gmail.com> > wrote: > >Look in the XLite advanced network settings and disable the 2 timeout > >settings (RTP and RTCP?). This is not always necessary, but there are > >sufficient cases where the packets XLite expects appear early on, but > >do not persist, thus causing a hangup. I think the default timeout is > >20 seconds. > > Thanks for the tip, but I get the same problem with SJPhone and > PhonerLite, so it looks like a problem in Asterisk. > > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20101222/14ffde84/attachment.htm>