Hello everyone, I've suffering cut offs after 6 or 7 seconds a call is answered, incoming calls are working fine, but outgoing ones show the gollowing messages when are being dropped: [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6399ms with no response [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3670 retrans_pkt: Hanging up call ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). This is happening with my PBX hosted on an external network and peers on my local network. It seems the SIP ACK is not being received properly. I'm using asterisk 1.8.19.0 on Debian 6.0.6 and 1.8.11.1 on Centos 5.9 Elder D. Arohuanca Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130515/ea647eca/attachment.htm>
asterisk is behind nat? On Wed, May 15, 2013 at 8:18 PM, Daniel - Asterisk <earohuanca at gmail.com>wrote:> Hello everyone, > > I've suffering cut offs after 6 or 7 seconds a call is answered, incoming > calls are working fine, but outgoing ones show the gollowing messages when > are being dropped: > > [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt: > Retransmission timeout reached on transmission > ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical > Response) -- See > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > Packet timed out after 6399ms with no response > [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3670 retrans_pkt: Hanging > up call ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. - no reply to our > critical packet (see > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). > This is happening with my PBX hosted on an external network and peers on > my local network. > > It seems the SIP ACK is not being received properly. > > I'm using asterisk 1.8.19.0 on Debian 6.0.6 and 1.8.11.1 on Centos 5.9 > > Elder D. Arohuanca > Lima - Peru > > -- > _____________________________________________________________________ > -- 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/20130515/2c24b46f/attachment.htm>
When the call is snswered, is there 2-way audio? Seems a natting issue. On Wednesday, May 15, 2013, Daniel - Asterisk wrote:> Hello everyone, > > I've suffering cut offs after 6 or 7 seconds a call is answered, incoming > calls are working fine, but outgoing ones show the gollowing messages when > are being dropped: > > [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt: > Retransmission timeout reached on transmission > ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical > Response) -- See > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > Packet timed out after 6399ms with no response > [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3670 retrans_pkt: Hanging > up call ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. - no reply to our > critical packet (see > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). > This is happening with my PBX hosted on an external network and peers on > my local network. > > It seems the SIP ACK is not being received properly. > > I'm using asterisk 1.8.19.0 on Debian 6.0.6 and 1.8.11.1 on Centos 5.9 > > Elder D. Arohuanca > Lima - Peru >-- Sent from Gmail Mobile -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130515/269aa615/attachment.htm>
Philipp von Klitzing
2013-May-19 01:10 UTC
[asterisk-users] Cut offs on outgoing SIP calls
Hi!> I've suffering cut offs after 6 or 7 seconds a call is answered, > incoming calls are working fine, but outgoing ones show the gollowing > messages when are being dropped > [...] > It seems the SIP ACK is not being received properly.I can confirm this issue: In my case it happens with calls coming in from a patton ISDN gateway to Asterisk 1.8.20.1. The calls is processed and passed to a snom phone, audio flows fine for a few seconds, but then Asterisk terminates the call. Interestingly this never happens on internal calls (from snom to snom). Downgrading to Asterisk 1.4 makes the issue go away as well. Have you tried 1.8.22? I haven't yet, but it seems to come with a fix for a deadlock in the SIP channel which *might* solve the issue we are both experiencing (see ASTERISK-21389). Philipp