Re: [asterisk-users] Cut offs on outgoing SIP calls
Hey Philipp, I will try soon the new version and let you know. Currently my users are pointing to a PBX in my local-private network with no problems. When I use wireshark I see my internal peers trying to send the ACK packets 4 or 5 times until hangup, at the same time the PBX are requesting that very packet many times until it decides to hangup (as you can see in previous message). The funny thing happens when I restart my router, everything works fine, but 2 or 3 hours later calls start getting cut-offs again. I'm not very used to routers but if someone have some tip on Cisco 2811 it will be great. Definitely it's a NAT issue, any help is welcome. Elder D. Arohuanca Lima - Peru On Sat, May 18, 2013 at 8:10 PM, Philipp von Klitzing phil...@vklitzing.com wrote: 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 -- _ -- 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 -- _ -- 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
Re: [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 -- _ -- 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
[asterisk-users] Cut offs on outgoing SIP calls
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
Re: [asterisk-users] Cut offs on outgoing SIP calls
asterisk is behind nat? On Wed, May 15, 2013 at 8:18 PM, Daniel - Asterisk earohua...@gmail.comwrote: 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 -- _ -- 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
Re: [asterisk-users] Cut offs on outgoing SIP calls
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 -- _ -- 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
Re: [asterisk-users] Cut offs on outgoing SIP calls
Users (softphones) are behind a NAT, Asterisk has its own public ip address On Wed, May 15, 2013 at 1:30 PM, Asghar Mohammad asghar...@gmail.comwrote: asterisk is behind nat? On Wed, May 15, 2013 at 8:18 PM, Daniel - Asterisk earohua...@gmail.comwrote: 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Cut offs on outgoing SIP calls
There was 2-way audio and suddenly, the calls when down. On Wed, May 15, 2013 at 1:30 PM, Gertjan Baarda gertjan.baa...@gmail.comwrote: 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Cut offs on outgoing SIP calls
please show us peer configuration. On Wed, May 15, 2013 at 9:46 PM, Daniel - Asterisk earohua...@gmail.comwrote: Users (softphones) are behind a NAT, Asterisk has its own public ip address On Wed, May 15, 2013 at 1:30 PM, Asghar Mohammad asghar...@gmail.comwrote: asterisk is behind nat? On Wed, May 15, 2013 at 8:18 PM, Daniel - Asterisk earohua...@gmail.comwrote: 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 -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Cut offs on outgoing SIP calls
Current configuration follows: [general] context=default allowguest=no alwaysauthreject=yes allowoverlap=yes allowtransfer=yes tcpenable=no tlsenable=no srvlookup=yes vmexten=vm rtcachefriends=yes nat=no directmedia=nonat directrtpsetup=no videosupport=yes maxcallbitrate=384 disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm allow=ilbc allow=speex allow=g726 allow=g723 mohinterpret=default mohsuggest=default dtmfmode=rfc2833 timer1b=6 transport=udp [carrier-1] host=a.b.c.d type=friend context=from-pstn disallow=all allow=ulaw,alaw qualify=yes trunk=yes [90102] secret=xx mailbox=90102@default cid_number=NX accountcode=401 type=friend host=dynamic port=5060 qualify=yes nat=yes transport=udp context=users disallow=all allow=ulaw,alaw,g729,gsm,speex,ilbc,h264,h263p,h263 directmedia=no canreinvite=no videosupport=no On Wed, May 15, 2013 at 2:47 PM, Asghar Mohammad asghar...@gmail.comwrote: please show us peer configuration. On Wed, May 15, 2013 at 9:46 PM, Daniel - Asterisk earohua...@gmail.comwrote: Users (softphones) are behind a NAT, Asterisk has its own public ip address On Wed, May 15, 2013 at 1:30 PM, Asghar Mohammad asghar...@gmail.comwrote: asterisk is behind nat? On Wed, May 15, 2013 at 8:18 PM, Daniel - Asterisk earohua...@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 -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Cut offs on outgoing SIP calls
sip set debug peer 90102 and check in log why call drop or upload log somewhere. configuration seems ok. On Wed, May 15, 2013 at 10:09 PM, Daniel - Asterisk earohua...@gmail.comwrote: Current configuration follows: [general] context=default allowguest=no alwaysauthreject=yes allowoverlap=yes allowtransfer=yes tcpenable=no tlsenable=no srvlookup=yes vmexten=vm rtcachefriends=yes nat=no directmedia=nonat directrtpsetup=no videosupport=yes maxcallbitrate=384 disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm allow=ilbc allow=speex allow=g726 allow=g723 mohinterpret=default mohsuggest=default dtmfmode=rfc2833 timer1b=6 transport=udp [carrier-1] host=a.b.c.d type=friend context=from-pstn disallow=all allow=ulaw,alaw qualify=yes trunk=yes [90102] secret=xx mailbox=90102@default cid_number=NX accountcode=401 type=friend host=dynamic port=5060 qualify=yes nat=yes transport=udp context=users disallow=all allow=ulaw,alaw,g729,gsm,speex,ilbc,h264,h263p,h263 directmedia=no canreinvite=no videosupport=no -- _ -- 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
