Hello list, I'm getting droped calls on an asterisk installation. When on GS phone dials another one, the call is dropped after some (usually random) time but most of the tome within 3 to 20 seconds. I think the cause is stated on the logs, see bellow, and is related with the message "Didn't get a frame from channel: SIP/3805-df43", but I can't figure why. asterisk logs: ------------------------------------- Mar 5 15:57:26 DEBUG[1116957488]: build_route: Contact hop: <sip:192.168.60.106> Mar 5 15:57:26 VERBOSE[1217669936]: -- SIP/3805-df43 answered SIP/-08122450 Mar 5 15:57:26 VERBOSE[1217669936]: -- Attempting native bridge of SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:26 DEBUG[1116957488]: Stopping retransmission on 'c9f88915cb5c25fd@192.168.60.107' of Response\ 25663: Found Mar 5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524 Mar 5 15:57:26 DEBUG[1217669936]: Ooh, format changed from UNKN to ULAW Mar 5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel: SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse counter Mar 5 15:57:38 DEBUG[1217669936]: is not a local user Mar 5 15:57:38 VERBOSE[1217669936]: == Spawn extension (local, 3805, 1) exited non-zero on 'SIP/-0812245\0' ----------------- The scenario: 1 server (redhat 9), asterisk (stable) and a 2 x hisax ISDN BRI. One of the BRI boards is used to dial out (ppp) on one channel and a mgetty on the other channel. The other board is in ptp and used by *. The phones are Grandstream BT101 and Handytone and are all on a switched network (3 procurve switches, stacked). The configs are ok, since the same files on another server work ok (no dropped calls), but I can post them if needed. Any help will be greatly appreciated. Thanks in advance, --- Paulo Loureiro
There is new firmware that may help http://www.grandstream.com/BETATEST/. Grandstream acknowledges this problem. They say it is a codec issue with asterisk. I don't know if this update addresses this problem but it may be worth a try.> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Paulo Loureiro > Sent: Friday, March 05, 2004 10:26 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] dropped calls > > Hello list, > > I'm getting droped calls on an asterisk installation. When on GS phone > dials another one, the call is dropped after some (usually > random) time > but most of the tome within 3 to 20 seconds. > I think the cause is stated on the logs, see bellow, and is > related with > the message "Didn't get a frame from channel: SIP/3805-df43", but I > can't figure why. > > > asterisk logs: > ------------------------------------- > Mar 5 15:57:26 DEBUG[1116957488]: build_route: Contact hop: > <sip:192.168.60.106> > Mar 5 15:57:26 VERBOSE[1217669936]: -- SIP/3805-df43 answered > SIP/-08122450 > Mar 5 15:57:26 VERBOSE[1217669936]: -- Attempting native > bridge of > SIP/-08122450 and SIP/3805-df43 > Mar 5 15:57:26 DEBUG[1116957488]: Stopping retransmission on > 'c9f88915cb5c25fd@192.168.60.107' of Response\ 25663: Found > Mar 5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524 > Mar 5 15:57:26 DEBUG[1217669936]: Ooh, format changed from > UNKN to ULAW > Mar 5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel: > SIP/3805-df43 > Mar 5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels > SIP/-08122450 and SIP/3805-df43 > Mar 5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse > counter > Mar 5 15:57:38 DEBUG[1217669936]: is not a local user > Mar 5 15:57:38 VERBOSE[1217669936]: == Spawn extension > (local, 3805, > 1) exited non-zero on 'SIP/-0812245\0' > ----------------- > > The scenario: > 1 server (redhat 9), asterisk (stable) and a 2 x hisax ISDN BRI. > One of the BRI boards is used to dial out (ppp) on one channel and a > mgetty on the other channel. The other board is in ptp and used by *. > The phones are Grandstream BT101 and Handytone and are all on > a switched > network (3 procurve switches, stacked). > > The configs are ok, since the same files on another server work ok (no > dropped calls), but I can post them if needed. > > > Any help will be greatly appreciated. > > Thanks in advance, > > > > --- Paulo Loureiro > > >
Lately, I have been experiencing unexpected hangups just when the a call has been established. This effects a small percentage of all calls coming from sip phone which are terminated on a zap pri channel. I turned on sip and pri debugging and it almost looks like the ACK message coming back from the sip agent in response to the "200 ok" message from the asterisk box which signaled the successful call setup would trigger a DISCONNECT message on the zap pri side. Asterisk spits the following line on the console just before issuing the DISCONNECT: NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peeerstate Connect Request I suspect this might have to do with the sip agents (all Grandstream ATAs/phones) as not all my users are affected. Has anybody of you seen this before? Thilo
I have been experiencing dropped calls on my iax2 connections between my Asterisk server and my ITSP providers, I use Teliax and Voxee but it seems to happen on both so I don't think it is the provider. I don't see any packet loss at the time so I don't think it is poor Internet connectivity, what else can I look for? Using Asterisk 1.2.6 but had this problem on 1.2.5 also. -- Chris Mason -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.