Lincoln King-Cliby wrote: > Hi All, > > I've looked through the archives and tried several variations in Google, and > I haven't found anything on-point... So I'm hoping someone here may be able > to help this relative Asterisk neophyte shed some light on an issue: > > I have a box running Asterisk 1.4.22 in our lab with several Cisco 7961G > phones and an AEX804E card (4 FXO, hardware echo cancellation). > > The server and all phones are on the same subnet (10.2.0.x/255.255.255.0) of > the local LAN with no NAT, routing, firewall, etc., etc. between the server > and the phones. > > Periodically I'm seeing calls placed from the 7961s through anything on the > PBX that requires digit entry (the Auto Attendant, Voicemail, etc.) > 'randomly' drop; extension-to-extension calls extension-to-PSTN, and > PSTN-to-extension calls never have any issues whatsoever. Nor have I been > able to duplicate the issues hopping around auto attendants on an inbound > PSTN call. > > When the call drops, the phone still thinks that it is connected, but the > audio path is cut off and something similar to the following is dumped to the > console > > -- <SIP/1103-b71184e0> Playing 'vm-password' (language 'en') > [Oct 28 14:23:07] WARNING[9423]: chan_sip.c:1958 retrans_pkt: Maximum retries > exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Response) > -- See doc/sip-retransmit.txt. > [Oct 28 14:23:07] WARNING[9423]: chan_sip.c:1980 retrans_pkt: Hanging up call > [EMAIL PROTECTED] - no reply to our critical packet (see > doc/sip-retransmit.txt). > > All of the results Google has turned up, and the doc/sip-retransmit.txt file > point to problems with things in the middle of the path between the server > and the phone (NAT, firewall, "SIP middle box", proxy) that simply don't > exist in the configuration that we're using. > > I suspect it's an issue with the way the Cisco phones are dealing with DTMF > to Asterisk or Asterisk dealing with the DTMF from Cisco but that's where I > go off into unknown territory. (FWIW, until the call drops everything works > fine, pressing a button triggers the desired action, and audio quality is > fantastic) > > I've rolled the firmware on the phones up and down with no noticeable change, > and I also upgraded to Asterisk 1.4.22 version of Asterisk (I had been > running 1.4.21.2, and there are fewer dropped calls with .22 but it's still > way too often to be acceptable) > > Any suggestions are greatly appreciated, but please be explicit... short of > editing the configuration files and "make install" my Asterisk experience is > rather limited. > > Thanks in advance, > > Lincoln
It's hard to diagnose a problem like this without a full SIP trace, but given the problem you are describing, it looks like Asterisk is sending a SIP INVITE that is not being replied to with a 200 OK. It wasn't clear in the scenario you presented why Asterisk would be sending an INVITE out anywhere though, so I'm not sure where this is originating. Is Asterisk dialing out to another box which contains the voicemail and auto-attendant services? If so, and if the box which provides these services is another Asterisk server, be sure that the second Asterisk server has an Answer in the dialplan for these calls. By the way, to see a SIP trace inside Asterisk, you can issue the command "sip set debug" in the CLI. Then all SIP messages will be written anywhere where you are logging verbose messages. Mark Michelson _______________________________________________ -- 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
