[Asterisk-Users] Unable to forward frame
Hi,
I get this error in the log file when I call from my mobile to the Asterisk
server, but hang up the mobile before anyone picks up.
Normally I would not worry about it, but I have been having some bad
experiences (only recently, after about 9 months of good operation) with
asterisk, although there have been related issues with Telco lines /
equipment and also some Asterisk initiated CRC errors after upgrading to
1.2. So I have downgraded to 1.0.9.
So I can isolate everything I have just installed a VERY VERY simple dial
plan.
The setup is
Telco --- TDM 4 Port BRI --- Ericsspn BP250
Extensions.conf (all of it)
[default]
exten = s,1,Dial(ZAP/g4/211,45,t)
[dialstring]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
[te405p-frombp250]
exten = _3XX,1,Answer
exten = _3XX,2,Dial(Sip/${EXTEN},6000,t)
exten = _3XX,3,Hangup
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g1/${EXTEN},6000,t)
exten = _X.,3,Hangup
[te405p-intelstra]
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g4/${EXTEN},6000,t)
exten = _X.,3,Hangup
[from-sip]
exten = s,1,Dial(SIP/3332,45,t)
exten = _0X.,1,Answer
exten = _0X.,2,Dial(Zap/g1/${EXTEN:1},6000,t)
exten = _0X.,3,Hangup
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g4/${EXTEN},6000,t)
exten = _X.,3,Hangup
Zapata.conf
[channels]
context=default
musiconhold=default
switchtype=euroisdn
usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=0.0
txgain=0.0
group=1
context=te405p-intelstra
;context=te405p-ext
pridialplan=local
signalling=pri_cpe
;overlapdial=yes
callerid=asreceived
channel=1-15, 17-31
;channel=32-46, 48-62
group=4
context=te405p-frombp250
;context=te405p-in
pridialplan=local
signalling=pri_net
overlapdial=yes
callerid=asreceived
channel=94-108, 110-124
;channel=32-46, 48-62
Zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=2,0,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
span=3,0,0,ccs,hdb3,crc4
bchan=63-77
dchan=78
bchan=79-93
span=4,0,0,ccs,hdb3,crc4
bchan=94-108
dchan=109
bchan=110-124
loadzone=au
defaultzone=au
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Re: [Asterisk-Users] Unable to forward frame
Hi,
I think there should be only one timing source, but you have 3 here...
Zaptel.conf
span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
span=3,0,0,ccs,hdb3,crc4
span=4,0,0,ccs,hdb3,crc4
Not sure if this is causing the problem though.
Andy
On 3/15/06, James Sturges [EMAIL PROTECTED] wrote:
Hi,
I get this error in the log file when I call from my mobile to the Asterisk
server, but hang up the mobile before anyone picks up.
Normally I would not worry about it, but I have been having some bad
experiences (only recently, after about 9 months of good operation) with
asterisk, although there have been related issues with Telco lines /
equipment and also some Asterisk initiated CRC errors after upgrading to
1.2. So I have downgraded to 1.0.9.
So I can isolate everything I have just installed a VERY VERY simple dial
plan.
The setup is
Telco --- TDM 4 Port BRI --- Ericsspn BP250
Extensions.conf (all of it)
[default]
exten = s,1,Dial(ZAP/g4/211,45,t)
[dialstring]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
[te405p-frombp250]
exten = _3XX,1,Answer
exten = _3XX,2,Dial(Sip/${EXTEN},6000,t)
exten = _3XX,3,Hangup
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g1/${EXTEN},6000,t)
exten = _X.,3,Hangup
[te405p-intelstra]
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g4/${EXTEN},6000,t)
exten = _X.,3,Hangup
[from-sip]
exten = s,1,Dial(SIP/3332,45,t)
exten = _0X.,1,Answer
exten = _0X.,2,Dial(Zap/g1/${EXTEN:1},6000,t)
exten = _0X.,3,Hangup
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g4/${EXTEN},6000,t)
exten = _X.,3,Hangup
Zapata.conf
[channels]
context=default
musiconhold=default
switchtype=euroisdn
usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=0.0
txgain=0.0
group=1
context=te405p-intelstra
;context=te405p-ext
pridialplan=local
signalling=pri_cpe
;overlapdial=yes
callerid=asreceived
channel=1-15, 17-31
;channel=32-46, 48-62
group=4
context=te405p-frombp250
;context=te405p-in
pridialplan=local
signalling=pri_net
overlapdial=yes
callerid=asreceived
channel=94-108, 110-124
;channel=32-46, 48-62
Zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=2,0,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
span=3,0,0,ccs,hdb3,crc4
bchan=63-77
dchan=78
bchan=79-93
span=4,0,0,ccs,hdb3,crc4
bchan=94-108
dchan=109
bchan=110-124
loadzone=au
defaultzone=au
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RE: [Asterisk-Users] Unable to forward frame
exten = _3XX,1,Answer
exten = _3XX,2,Dial(Sip/${EXTEN},6000,t)
exten = _3XX,3,Hangup
Why do you Answer before you Dial here? I had a problem where calls were
misbehaving and someone asked me that same question. Without really
understanding why I removed the Answer and it then just worked.
I think the idea is that Dial connects the incoming channel - still in
the 'ringing' state - to the outgoing channel, and then the outgoing
channel takes care of answering the incoming call (or not).
Or maybe you know what you are doing and there is a perfectly good
reason to Answer before Dial.
James
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RE: [Asterisk-Users] Unable to forward frame
It has to do with transcoding. If Asterisk cannot 'speak' the codec it
cannot answer the call and is 'unable to forward the frame.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
James Harper
Sent: Wednesday, March 15, 2006 5:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Unable to forward frame
exten = _3XX,1,Answer
exten = _3XX,2,Dial(Sip/${EXTEN},6000,t) exten = _3XX,3,Hangup
Why do you Answer before you Dial here? I had a problem where
calls were misbehaving and someone asked me that same
question. Without really understanding why I removed the
Answer and it then just worked.
I think the idea is that Dial connects the incoming channel -
still in the 'ringing' state - to the outgoing channel, and
then the outgoing channel takes care of answering the
incoming call (or not).
Or maybe you know what you are doing and there is a perfectly
good reason to Answer before Dial.
James
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RE: [Asterisk-Users] Unable to forward frame
Thanks for the input everyone.
I though the second digit in the span = was the timing attribute, so only
getting master timing on span 1.
We have had timing issues, can we confirm this?
Thanks again
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andy Kuo
Sent: Thursday, 16 March 2006 6:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Unable to forward frame
Hi,
I think there should be only one timing source, but you have 3 here...
Zaptel.conf
span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
span=3,0,0,ccs,hdb3,crc4
span=4,0,0,ccs,hdb3,crc4
Not sure if this is causing the problem though.
Andy
On 3/15/06, James Sturges [EMAIL PROTECTED] wrote:
Hi,
I get this error in the log file when I call from my mobile to the
Asterisk
server, but hang up the mobile before anyone picks up.
Normally I would not worry about it, but I have been having some bad
experiences (only recently, after about 9 months of good operation) with
asterisk, although there have been related issues with Telco lines /
equipment and also some Asterisk initiated CRC errors after upgrading to
1.2. So I have downgraded to 1.0.9.
So I can isolate everything I have just installed a VERY VERY simple dial
plan.
The setup is
Telco --- TDM 4 Port BRI --- Ericsspn BP250
Extensions.conf (all of it)
[default]
exten = s,1,Dial(ZAP/g4/211,45,t)
[dialstring]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
[te405p-frombp250]
exten = _3XX,1,Answer
exten = _3XX,2,Dial(Sip/${EXTEN},6000,t)
exten = _3XX,3,Hangup
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g1/${EXTEN},6000,t)
exten = _X.,3,Hangup
[te405p-intelstra]
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g4/${EXTEN},6000,t)
exten = _X.,3,Hangup
[from-sip]
exten = s,1,Dial(SIP/3332,45,t)
exten = _0X.,1,Answer
exten = _0X.,2,Dial(Zap/g1/${EXTEN:1},6000,t)
exten = _0X.,3,Hangup
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g4/${EXTEN},6000,t)
exten = _X.,3,Hangup
Zapata.conf
[channels]
context=default
musiconhold=default
switchtype=euroisdn
usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=0.0
txgain=0.0
group=1
context=te405p-intelstra
;context=te405p-ext
pridialplan=local
signalling=pri_cpe
;overlapdial=yes
callerid=asreceived
channel=1-15, 17-31
;channel=32-46, 48-62
group=4
context=te405p-frombp250
;context=te405p-in
pridialplan=local
signalling=pri_net
overlapdial=yes
callerid=asreceived
channel=94-108, 110-124
;channel=32-46, 48-62
Zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=2,0,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
span=3,0,0,ccs,hdb3,crc4
bchan=63-77
dchan=78
bchan=79-93
span=4,0,0,ccs,hdb3,crc4
bchan=94-108
dchan=109
bchan=110-124
loadzone=au
defaultzone=au
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Re: [Asterisk-Users] Unable to forward frame/voice
A brief update for those who are interested. This turned out to be my problem (DOH!). After doing a PRI debug on the span on the telco side to a log file, I found that the 'Unable to forward voice' message corresponded one for one with the following progress message: Message type: PROGRESS (3) [08 02 84 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] In other words, it looks like the failed calls are failing because the numbers are invalid. I verified this with a regular POTS phone on a few, and that was, in fact, the case. A suggestion, if anyone is interested, would be that 'Unable to forward voice' is maybe not the best message for app_dial to put in the asterisk log when this condition occurrs. It would have been nice (and easier to figure out) if what went into the log was something more along the lines of the actual cause for what happened. Thanks, Christian Quoting Christian M. Watts [EMAIL PROTECTED]: Hi, We've exhausted our internal capabilities as well as Sangoma tech support and were hoping someone with some expertise could help us with a pointer. Briefly, our issue is as follows. Periodically (several times an hour), we get either of the following error messages in our asterisk messages log. These correspond with dropped outbound calls on a one-to-one basis when the second error happens. The first error sometimes causes a dropped call and sometimes does not: Jun 30 16:40:27 WARNING[5395] app_dial.c: Unable to forward frame Jun 30 16:45:07 WARNING[5455] app_dial.c: Unable to forward voice Our hardware is as follows: Compaq DL380 Dual PIII 1Ghz, 1.2 GB RAM, Onboard SmartArray for SCSI RAID Sangoma A102U dual-port T1 card Digi Datafire T1 fax/modem board Our software is as follows: Linux 2.4.30 Asterisk, Zaptel and Libpri from CVS HEAD as of 6/28/05 Sangoma wanpipe 2.3.3-beta11 (latest as of this post) Patton electronic's latest drivers and firmware for our Digi Datafire board (still no 2.6 Linux support, which is why we're on 2.4) Hylafax 4.2.1 driving the Digi Datafire The path (for the problem calls) looks like this: Digi Datafire - Sangoma Port B - Sangoma Port A - Telco Basically, sending a fax over a PRI with asterisk doing TDM bridging in the middle. We have confirmed the following (based on similar posts to this list related to the same problem with Digium boards as well as Sangoma tech support assistance): 1. Sangoma Port A takes clocking from the telco 2. Sangoma Port B retransmits A's clocking and acts as master 3. Sangoma tech support says our configs are correct 4. Zaptel.conf is set up with Sangoma Port A as the primary clock source, and Port B to not be used as a clock source 5. LBO, switch options, etc. are correct for the environment (since 98% of outbound calls are fine, this seems fairly obvious) 6. ISDN Transfer Capability gets properly set to 3K1AUDIO for calls 7. No IRQ sharing on the system 8. IDE DMA mode is irrelevant, since there are no IDE disks in the system (other than the CDROM) We have tried the following: 1. Asterisk, libpri and zaptel versions from 6/1/2005, 6/15/2005 and 6/28/2005 - no change in behavior 2. Wanpipe drivers 2.3.3-beta8 and 2.3.3-beta11 - no change in behavior 3. Wanpipe configured both with and without the D-Channel hardware HDLC - no change in behavior 4. Firmware versions 24 (shipped version) and 25 (latest version) on the Sangoma card - no change in behavior 5. callprogress and busydetect both 'yes' and 'no' in zapata.conf (currently 'no') - no change in behavior 6. Added SetTransferCapability(3K1AUDIO) to our dialplan, just to be sure - no change in behavior General environment: 1. We run TDM only, no VoIP protocols are in use. SIP, IAX2, MGCP are all noload in modules.conf. 2. This problem occurs with as few as one simultaneous channel active and as many as 15 simultaneous channels active with equal frequency (i.e.: not load related). The load on the box is negligible in any case, plenty of RAM is free, etc. 3. Restarting asterisk does seem to cause the problem not to re-present itself for 30 minutes to 2 hours. When asterisk is restarted, the Sangoma and Zaptel kernel modules are also unloaded and reloaded. Again, any pointers or help would be greatly appreciated. Thanks, Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Christian Watts EC Data Systems, Inc. 303.991.6020 - Voice 303.991.6021 - Fax ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update
[Asterisk-Users] Unable to forward frame/voice
Hi, We've exhausted our internal capabilities as well as Sangoma tech support and were hoping someone with some expertise could help us with a pointer. Briefly, our issue is as follows. Periodically (several times an hour), we get either of the following error messages in our asterisk messages log. These correspond with dropped outbound calls on a one-to-one basis when the second error happens. The first error sometimes causes a dropped call and sometimes does not: Jun 30 16:40:27 WARNING[5395] app_dial.c: Unable to forward frame Jun 30 16:45:07 WARNING[5455] app_dial.c: Unable to forward voice Our hardware is as follows: Compaq DL380 Dual PIII 1Ghz, 1.2 GB RAM, Onboard SmartArray for SCSI RAID Sangoma A102U dual-port T1 card Digi Datafire T1 fax/modem board Our software is as follows: Linux 2.4.30 Asterisk, Zaptel and Libpri from CVS HEAD as of 6/28/05 Sangoma wanpipe 2.3.3-beta11 (latest as of this post) Patton electronic's latest drivers and firmware for our Digi Datafire board (still no 2.6 Linux support, which is why we're on 2.4) Hylafax 4.2.1 driving the Digi Datafire The path (for the problem calls) looks like this: Digi Datafire - Sangoma Port B - Sangoma Port A - Telco Basically, sending a fax over a PRI with asterisk doing TDM bridging in the middle. We have confirmed the following (based on similar posts to this list related to the same problem with Digium boards as well as Sangoma tech support assistance): 1. Sangoma Port A takes clocking from the telco 2. Sangoma Port B retransmits A's clocking and acts as master 3. Sangoma tech support says our configs are correct 4. Zaptel.conf is set up with Sangoma Port A as the primary clock source, and Port B to not be used as a clock source 5. LBO, switch options, etc. are correct for the environment (since 98% of outbound calls are fine, this seems fairly obvious) 6. ISDN Transfer Capability gets properly set to 3K1AUDIO for calls 7. No IRQ sharing on the system 8. IDE DMA mode is irrelevant, since there are no IDE disks in the system (other than the CDROM) We have tried the following: 1. Asterisk, libpri and zaptel versions from 6/1/2005, 6/15/2005 and 6/28/2005 - no change in behavior 2. Wanpipe drivers 2.3.3-beta8 and 2.3.3-beta11 - no change in behavior 3. Wanpipe configured both with and without the D-Channel hardware HDLC - no change in behavior 4. Firmware versions 24 (shipped version) and 25 (latest version) on the Sangoma card - no change in behavior 5. callprogress and busydetect both 'yes' and 'no' in zapata.conf (currently 'no') - no change in behavior 6. Added SetTransferCapability(3K1AUDIO) to our dialplan, just to be sure - no change in behavior General environment: 1. We run TDM only, no VoIP protocols are in use. SIP, IAX2, MGCP are all noload in modules.conf. 2. This problem occurs with as few as one simultaneous channel active and as many as 15 simultaneous channels active with equal frequency (i.e.: not load related). The load on the box is negligible in any case, plenty of RAM is free, etc. 3. Restarting asterisk does seem to cause the problem not to re-present itself for 30 minutes to 2 hours. When asterisk is restarted, the Sangoma and Zaptel kernel modules are also unloaded and reloaded. Again, any pointers or help would be greatly appreciated. Thanks, Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
