Re: [asterisk-users] High resident memory with 11.14.0 ?
On Wed, Nov 26, 2014 at 3:20 PM, James Lamanna jlama...@gmail.com wrote:
On Tue, Nov 25, 2014 at 10:21 AM, James Lamanna jlama...@gmail.com
wrote:
On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan mjor...@digium.com
wrote:
On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna jlama...@gmail.com
wrote:
Also, how big does the cache in frame.c grow to?
I've recompiled with MALLOC_DEBUG on that server:
asterisk -rx memory show summary
1780466242 bytes (1780181594 cache) in2352909 allocations in file
frame.c
...
Seems like a ridiculous cache.
I'm not going to respond to your new thread, since it is the same
discussion as this one.
The frame cache is a per-thread local cache of frames that prevents
having to re-allocate frames as they pass through Asterisk. Clearly,
something is abusing it.
I think you'll need to provide some more information on how you're
producing this situation. Specifically:
* Channel technologies involved, and the formats on the channels
* Dialplan that reproduces the problem
Are you using any non-core dialplan applications or channel drivers?
This PBX has about 100 registered SIP clients, along with 23 PRI
channels, 2 inbound/outbound SIP trunks and around 100 IAXModems registered
to it. It primarily handles faxing.
I am not using any non-standard channel drivers. I am using the T.38
gateway funcionality.
The jist of the dialplan is this: (example of the PRI and a SIP trunk,
inbound)
[pri-in]
exten = _X.,1,Set(__FROM_DID=${EXTEN})
exten = _X.,n,Set(FAX_IDX=700)
exten = _X.,n,Set(MAX_IDX=719)
exten = _X.,n,Goto(dial-hylafax,s,1)
[sip-trunk-in]
exten = _X.,1(normal),Set(__FROM_DID=${EXTEN})
exten = _X.,n,Set(FAX_IDX=950)
exten = _X.,n,Set(MAX_IDX=959)
exten = _X.,n,Set(FAXOPT(gateway)=yes)
exten = _X.,n,Goto(dial-hylafax,s,1)
[dial-hylafax]
exten = s,1,GotoIf($[${FROM_DID:0:1} = 1]?prune:cont)
exten = s,n(prune),Set(__FROM_DID=${FROM_DID:1})
exten = s,n(cont),GotoIf($[${FAX_IDX} = ${MAX_IDX}]?tryfax:nofax)
exten = s,n(tryfax),Set(STATE=${DEVICE_STATE(Custom:iaxmodem${FAX_IDX})})
exten = s,n,NoOp(${STATE})
exten = s,n,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=INUSE)
exten = s,n,Dial(IAX2/iaxmodem${FAX_IDX}/${FROM_DID},60,g)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s,n(nofax),Playtones(busy)
exten = s,n,NoOp(NO MODEMS AVAILABLE)
exten = s,n,Wait(20)
exten = s,n,Hangup()
exten = s-ANSWER,1,NoOp(IAXMODEM HANGUP)
exten = s-ANSWER,n,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=NOT_INUSE)
exten = s-ANSWER,n,Hangup()
exten = _s-.,1,Set(FAX_IDX=${MATH(1+${FAX_IDX},i)})
exten = _s-.,n,Goto(s,1)
exten = h,1,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=NOT_INUSE)
The current state requires me to restart Asterisk almost every day.
I'm also seeing this on a completely different machine after upgrading
from Asterisk10 to 11.
I'm wondering if this is a problem in the SLIN converter?
I do use SLIN with iaxmodem.
Also of note,
A quick valgrind run and attempting to send a few faxes produces a bunch of
these in the valgrind output:
==30640== 217,259 bytes in 287 blocks are definitely lost in loss record
1,778 of 1,789
==30640==at 0x4C267CC: calloc (vg_replace_malloc.c:467)
==30640==by 0x4DC50E: ast_frdup (utils.h:523)
==30640==by 0x47125F: __ast_queue_frame (channel.c:1284)
==30640==by 0x1EF75589: __do_deliver (chan_iax2.c:3102)
==30640==by 0x1EF76C5A: schedule_delivery (chan_iax2.c:4374)
==30640==by 0x1EF8F497: socket_process_helper (chan_iax2.c:12010)
==30640==by 0x1EF99C37: iax2_process_thread (chan_iax2.c:12030)
==30640==by 0x56C458: dummy_start (utils.c:1192)
==30640==by 0x5E359C9: start_thread (pthread_create.c:300)
==30640==by 0x270326FF: ???
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Re: [asterisk-users] High resident memory with 11.14.0 ?
On Tue, Nov 25, 2014 at 10:21 AM, James Lamanna jlama...@gmail.com wrote:
On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan mjor...@digium.com
wrote:
On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna jlama...@gmail.com
wrote:
Also, how big does the cache in frame.c grow to?
I've recompiled with MALLOC_DEBUG on that server:
asterisk -rx memory show summary
1780466242 bytes (1780181594 cache) in2352909 allocations in file
frame.c
...
Seems like a ridiculous cache.
I'm not going to respond to your new thread, since it is the same
discussion as this one.
The frame cache is a per-thread local cache of frames that prevents
having to re-allocate frames as they pass through Asterisk. Clearly,
something is abusing it.
I think you'll need to provide some more information on how you're
producing this situation. Specifically:
* Channel technologies involved, and the formats on the channels
* Dialplan that reproduces the problem
Are you using any non-core dialplan applications or channel drivers?
This PBX has about 100 registered SIP clients, along with 23 PRI channels,
2 inbound/outbound SIP trunks and around 100 IAXModems registered to it. It
primarily handles faxing.
I am not using any non-standard channel drivers. I am using the T.38
gateway funcionality.
The jist of the dialplan is this: (example of the PRI and a SIP trunk,
inbound)
[pri-in]
exten = _X.,1,Set(__FROM_DID=${EXTEN})
exten = _X.,n,Set(FAX_IDX=700)
exten = _X.,n,Set(MAX_IDX=719)
exten = _X.,n,Goto(dial-hylafax,s,1)
[sip-trunk-in]
exten = _X.,1(normal),Set(__FROM_DID=${EXTEN})
exten = _X.,n,Set(FAX_IDX=950)
exten = _X.,n,Set(MAX_IDX=959)
exten = _X.,n,Set(FAXOPT(gateway)=yes)
exten = _X.,n,Goto(dial-hylafax,s,1)
[dial-hylafax]
exten = s,1,GotoIf($[${FROM_DID:0:1} = 1]?prune:cont)
exten = s,n(prune),Set(__FROM_DID=${FROM_DID:1})
exten = s,n(cont),GotoIf($[${FAX_IDX} = ${MAX_IDX}]?tryfax:nofax)
exten = s,n(tryfax),Set(STATE=${DEVICE_STATE(Custom:iaxmodem${FAX_IDX})})
exten = s,n,NoOp(${STATE})
exten = s,n,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=INUSE)
exten = s,n,Dial(IAX2/iaxmodem${FAX_IDX}/${FROM_DID},60,g)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s,n(nofax),Playtones(busy)
exten = s,n,NoOp(NO MODEMS AVAILABLE)
exten = s,n,Wait(20)
exten = s,n,Hangup()
exten = s-ANSWER,1,NoOp(IAXMODEM HANGUP)
exten = s-ANSWER,n,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=NOT_INUSE)
exten = s-ANSWER,n,Hangup()
exten = _s-.,1,Set(FAX_IDX=${MATH(1+${FAX_IDX},i)})
exten = _s-.,n,Goto(s,1)
exten = h,1,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=NOT_INUSE)
The current state requires me to restart Asterisk almost every day.
I'm also seeing this on a completely different machine after upgrading
from Asterisk10 to 11.
I'm wondering if this is a problem in the SLIN converter?
I do use SLIN with iaxmodem.
-- James
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Re: [asterisk-users] High resident memory with 11.14.0 ?
On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan mjor...@digium.com wrote:
On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna jlama...@gmail.com wrote:
Also, how big does the cache in frame.c grow to?
I've recompiled with MALLOC_DEBUG on that server:
asterisk -rx memory show summary
1780466242 bytes (1780181594 cache) in2352909 allocations in file
frame.c
...
Seems like a ridiculous cache.
I'm not going to respond to your new thread, since it is the same
discussion as this one.
The frame cache is a per-thread local cache of frames that prevents
having to re-allocate frames as they pass through Asterisk. Clearly,
something is abusing it.
I think you'll need to provide some more information on how you're
producing this situation. Specifically:
* Channel technologies involved, and the formats on the channels
* Dialplan that reproduces the problem
Are you using any non-core dialplan applications or channel drivers?
This PBX has about 100 registered SIP clients, along with 23 PRI channels,
2 inbound/outbound SIP trunks and around 100 IAXModems registered to it. It
primarily handles faxing.
I am not using any non-standard channel drivers. I am using the T.38
gateway funcionality.
The jist of the dialplan is this: (example of the PRI and a SIP trunk,
inbound)
[pri-in]
exten = _X.,1,Set(__FROM_DID=${EXTEN})
exten = _X.,n,Set(FAX_IDX=700)
exten = _X.,n,Set(MAX_IDX=719)
exten = _X.,n,Goto(dial-hylafax,s,1)
[sip-trunk-in]
exten = _X.,1(normal),Set(__FROM_DID=${EXTEN})
exten = _X.,n,Set(FAX_IDX=950)
exten = _X.,n,Set(MAX_IDX=959)
exten = _X.,n,Set(FAXOPT(gateway)=yes)
exten = _X.,n,Goto(dial-hylafax,s,1)
[dial-hylafax]
exten = s,1,GotoIf($[${FROM_DID:0:1} = 1]?prune:cont)
exten = s,n(prune),Set(__FROM_DID=${FROM_DID:1})
exten = s,n(cont),GotoIf($[${FAX_IDX} = ${MAX_IDX}]?tryfax:nofax)
exten = s,n(tryfax),Set(STATE=${DEVICE_STATE(Custom:iaxmodem${FAX_IDX})})
exten = s,n,NoOp(${STATE})
exten = s,n,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=INUSE)
exten = s,n,Dial(IAX2/iaxmodem${FAX_IDX}/${FROM_DID},60,g)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s,n(nofax),Playtones(busy)
exten = s,n,NoOp(NO MODEMS AVAILABLE)
exten = s,n,Wait(20)
exten = s,n,Hangup()
exten = s-ANSWER,1,NoOp(IAXMODEM HANGUP)
exten = s-ANSWER,n,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=NOT_INUSE)
exten = s-ANSWER,n,Hangup()
exten = _s-.,1,Set(FAX_IDX=${MATH(1+${FAX_IDX},i)})
exten = _s-.,n,Goto(s,1)
exten = h,1,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=NOT_INUSE)
The current state requires me to restart Asterisk almost every day.
I'm also seeing this on a completely different machine after upgrading from
Asterisk10 to 11.
-- James
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Re: [asterisk-users] High resident memory with 11.14.0 ?
cat /proc/cpuinfo lists 4 cores. So even if that's not showing hyperthreading, maximum 8. By your rule, that would be 8 cores * 0.5GB = 4GB memory. I've seen resident memory be up over 6GB. On Sat, Nov 22, 2014 at 1:29 PM, Freddi Hansen f...@danovation.dk wrote: Its up to 5.8G of resident memory with 28321 calls processed. The OOM killer is going to kill this soon at this rate (8GB RAM machine). This seems like a pretty serious problem. It looks like I'll need to restart asterisk every night Hi the number of cpu cores that you see with top times 512Mbyte is the level of ram that's needed e.g. a hp-gen8 with 2 octo core cpu's and hyperthreading enabled will be ( 2 x 8 x 2 x 0,5 gb ) = 16 gb + a bit exstra. So from start memory usage increases until it reaches 17.3 gb and then stabilizes. at that level. You can disables hypertreading and cut your ram usage to half of that. I can't see what hardware you are using but I think you need to check that the rule above fits your hardware. b.r. Freddi On Fri, Nov 21, 2014 at 10:53 AM, James Lamanna jlama...@gmail.com wrote: Hi, I have an Asterisk server that's been running now for around 2 days. I've noticed that the resident memory seems to be very high for its current call load: PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 18321 asterisk 20 0 8050m 5.2g 6968 S 13 66.2 363:11.80 asterisk $ asterisk -rx core show channels 24 active channels 12 active calls 25216 calls processed This server has a bunch of IAXModems hooked up to it and is mainly used as a Fax gateway to hylafax. Is this normal? 5.2Gig of memory used after 2 days with only 12 currently active calls? I am not using any realtime peers. There are 100 registered SIP peers on this server as well. Thanks. -- James -- _ -- 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] High resident memory with 11.14.0 ?
Also, how big does the cache in frame.c grow to? I've recompiled with MALLOC_DEBUG on that server: asterisk -rx memory show summary 1780466242 bytes (1780181594 cache) in2352909 allocations in file frame.c ... Seems like a ridiculous cache. On Mon, Nov 24, 2014 at 9:02 AM, James Lamanna jlama...@gmail.com wrote: cat /proc/cpuinfo lists 4 cores. So even if that's not showing hyperthreading, maximum 8. By your rule, that would be 8 cores * 0.5GB = 4GB memory. I've seen resident memory be up over 6GB. On Sat, Nov 22, 2014 at 1:29 PM, Freddi Hansen f...@danovation.dk wrote: Its up to 5.8G of resident memory with 28321 calls processed. The OOM killer is going to kill this soon at this rate (8GB RAM machine). This seems like a pretty serious problem. It looks like I'll need to restart asterisk every night Hi the number of cpu cores that you see with top times 512Mbyte is the level of ram that's needed e.g. a hp-gen8 with 2 octo core cpu's and hyperthreading enabled will be ( 2 x 8 x 2 x 0,5 gb ) = 16 gb + a bit exstra. So from start memory usage increases until it reaches 17.3 gb and then stabilizes. at that level. You can disables hypertreading and cut your ram usage to half of that. I can't see what hardware you are using but I think you need to check that the rule above fits your hardware. b.r. Freddi On Fri, Nov 21, 2014 at 10:53 AM, James Lamanna jlama...@gmail.com wrote: Hi, I have an Asterisk server that's been running now for around 2 days. I've noticed that the resident memory seems to be very high for its current call load: PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 18321 asterisk 20 0 8050m 5.2g 6968 S 13 66.2 363:11.80 asterisk $ asterisk -rx core show channels 24 active channels 12 active calls 25216 calls processed This server has a bunch of IAXModems hooked up to it and is mainly used as a Fax gateway to hylafax. Is this normal? 5.2Gig of memory used after 2 days with only 12 currently active calls? I am not using any realtime peers. There are 100 registered SIP peers on this server as well. Thanks. -- James -- _ -- 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
[asterisk-users] Size of frame.c cache in Asterisk 11?
(Starting a new email topic for this specific issue) Hi, What is the maximum size of the frame.c cache in Asterisk 11 and why does it constantly increase? This is what I'm up to already: $ asterisk -rx memory show summary 3667584471 bytes (3667366799 cache) in4846685 allocations in file frame.c ~$ asterisk -rx core show uptime System uptime: 2 days, 11 hours, 12 minutes, 12 seconds Last reload: 2 days, 11 hours, 12 minutes, 12 seconds $ asterisk -rx core show channels 34 active channels 17 active calls 13824 calls processed This seems like very odd behavior. Thanks. -- James -- _ -- 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] High resident memory with 11.14.0 ?
Hi, I have an Asterisk server that's been running now for around 2 days. I've noticed that the resident memory seems to be very high for its current call load: PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 18321 asterisk 20 0 8050m 5.2g 6968 S 13 66.2 363:11.80 asterisk $ asterisk -rx core show channels 24 active channels 12 active calls 25216 calls processed This server has a bunch of IAXModems hooked up to it and is mainly used as a Fax gateway to hylafax. Is this normal? 5.2Gig of memory used after 2 days with only 12 currently active calls? I am not using any realtime peers. There are 100 registered SIP peers on this server as well. Thanks. -- James -- _ -- 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] High resident memory with 11.14.0 ?
Its up to 5.8G of resident memory with 28321 calls processed. The OOM killer is going to kill this soon at this rate (8GB RAM machine). This seems like a pretty serious problem. It looks like I'll need to restart asterisk every night On Fri, Nov 21, 2014 at 10:53 AM, James Lamanna jlama...@gmail.com wrote: Hi, I have an Asterisk server that's been running now for around 2 days. I've noticed that the resident memory seems to be very high for its current call load: PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 18321 asterisk 20 0 8050m 5.2g 6968 S 13 66.2 363:11.80 asterisk $ asterisk -rx core show channels 24 active channels 12 active calls 25216 calls processed This server has a bunch of IAXModems hooked up to it and is mainly used as a Fax gateway to hylafax. Is this normal? 5.2Gig of memory used after 2 days with only 12 currently active calls? I am not using any realtime peers. There are 100 registered SIP peers on this server as well. Thanks. -- James -- _ -- 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] Setting channel musicclass from AGI
Hi Matt,
So this actually works (haven't had a chance to try it)?
SET VARIABLE CHANNEL(musicclass) default
Because musicclass is piece of channel information.
Referencing ${musicclass} is not the same thing.
Thanks.
-- James
On Sun, Oct 5, 2014 at 8:05 PM, Matthew Jordan mjor...@digium.com wrote:
On Sun, Oct 5, 2014 at 6:40 PM, James Lamanna jlama...@gmail.com wrote:
Hi,
Since SetMusicOnHold() is being deprecated, how do we set the channel
musicclass from an AGI script?
Last time I checked you can't call dialplan functions from AGI.
Actually, you can. Any time you can evaluate or set a channel
variable, you can also evaluate or set a dialplan function. Hence, you
can use both 'get variable' [1] or 'set variable' [2]. You could also
use 'exec' and call the Set dialplan application directly.
[1] https://wiki.asterisk.org/wiki/display/AST/AGICommand_get+variable
[2] https://wiki.asterisk.org/wiki/display/AST/AGICommand_set+variable
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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[asterisk-users] Setting channel musicclass from AGI
Hi, Since SetMusicOnHold() is being deprecated, how do we set the channel musicclass from an AGI script? Last time I checked you can't call dialplan functions from AGI. Thanks. -- James -- _ -- 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] Duplicated DTMF issues
Hi, I have a 1.8.22 Asterisk (Box A) connected to a 1.4.32 Asterisk box (Box B) through SIP. The 1.4.32 box is then connected to the PSTN through PRIs. I've noticed there are occasions where I am seeing duplicated DTMF. I've verified from the SIP trace from the phone that there is only a single '3' being pressed. It appears as though the DTMF end (without a begin) that is detected on Box B is being turned into a duplicate '3'. Does anyone have an idea how to fix this? Do I need to lower the requested DTMF duration? Thanks. -- James This is what the DTMF logs look like. Log on Box A (from phone): [Jul 3 13:56:24] DTMF[30040] channel.c: DTMF begin '3' received on SIP/3401-00034777 [Jul 3 13:56:24] DTMF[30040] channel.c: DTMF begin passthrough '3' on SIP/3401-00034777 [Jul 3 13:56:24] DTMF[30040] channel.c: DTMF end '3' received on SIP/3401-00034777, duration 90 ms [Jul 3 13:56:24] DTMF[30040] channel.c: DTMF end accepted with begin '3' on SIP/3401-00034777 [Jul 3 13:56:24] DTMF[30040] channel.c: DTMF end '3' detected to have actual duration 77 on the wire, emulation will be triggered on SIP/3401-00034777 [Jul 3 13:56:24] DTMF[30040] channel.c: DTMF end '3' has duration 77 but want minimum 80, emulating on SIP/3401-00034777 [Jul 3 13:56:24] DTMF[30040] channel.c: DTMF end emulation of '3' queued on SIP/3401-00034777 Log on Box B (from Box A): [Jul 3 13:56:24] DTMF[14562] channel.c: DTMF end '3' received on SIP/dp-pbx0-00022756, duration 0 ms [Jul 3 13:56:24] DTMF[14562] channel.c: DTMF begin emulation of '3' with duration 100 queued on SIP/dp-pbx0-00022756 [Jul 3 13:56:24] DTMF[14562] channel.c: DTMF end emulation of '3' queued on SIP/dp-pbx0-00022756 Log on Box B (from PSTN): [Jul 3 13:56:24] DTMF[14568] channel.c: DTMF begin '3' received on Zap/41-1 [Jul 3 13:56:24] DTMF[14568] channel.c: DTMF begin passthrough '3' on Zap/41-1 [Jul 3 13:56:24] DTMF[14568] channel.c: DTMF end '3' received on Zap/41-1, duration 172 ms [Jul 3 13:56:24] DTMF[14568] channel.c: DTMF end accepted with begin '3' on Zap/41-1 [Jul 3 13:56:24] DTMF[14568] channel.c: DTMF end passthrough '3' on Zap/41-1 [Jul 3 13:56:24] DTMF[14568] channel.c: DTMF begin '3' received on Zap/41-1 [Jul 3 13:56:24] DTMF[14568] channel.c: DTMF begin passthrough '3' on Zap/41-1 [Jul 3 13:56:24] DTMF[14568] channel.c: DTMF end '3' received on Zap/41-1, duration 172 ms [Jul 3 13:56:24] DTMF[14568] channel.c: DTMF end accepted with begin '3' on Zap/41-1 [Jul 3 13:56:24] DTMF[14568] channel.c: DTMF end passthrough '3' on Zap/41-1 -- _ -- 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] Voicemail Prepend not working properly on 1.8.18
Hi, I have a problem with forwarding a voicemail and prepending a message to it. If a user just forwards a voicemail, everything works fine. However, if a user prepends a message to the voicemail when forwarding, the voicemail that is forwarded only contains the prepended message and not the original voicemail message. Also, I continue to have voicemails and recordings that are recording the '#' to end the message. Thanks. -- James -- _ -- 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] DTMF Blips at end of Record() - 1.8.18
On Wed, Feb 20, 2013 at 10:49 AM, James Lamanna jlama...@gmail.com wrote: Hi, I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this? The DTMF is coming in through rfc2833 and not inband. I have another PBX running the same exact version of asterisk that's newer that doesn't exhibit the same problem. I'm wondering if it is a timing thing? The PBX with the issue seems to respond slower to DTMF (Background() takes longer to get interrupted, etc..) It is an older box so it is using res_timing_dahdi (dahdi_dummy) and the newer box is using res_timing_timerfd. Any suggestions would be welcome. Thanks. -- James -- _ -- 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] DTMF Blips at end of Record() - 1.8.18
Hi, I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this? The DTMF is coming in through rfc2833 and not inband. Thanks. -- James -- _ -- 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] Congestion() forcing PRI channels to be not available
On Thu, Dec 20, 2012 at 2:22 AM, Steve Davies davies...@gmail.com wrote: On 19 December 2012 21:54, Christopher Harrington ch...@acsdi.com wrote: You probably already know this, but 1.4x is very old (released in 2006) and is officially end-of-life. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions You might get more help or better behavior by updating to a newer more current version of Asterisk, such as 1.8 which will be receiving bug fixes into October 2014. On Wed, Dec 19, 2012 at 3:47 PM, James Lamanna jlama...@gmail.comwrote: Hi, I have a PSTN Asterisk box that's connected to other dialplan PBXes through IAX2. Recently this box was upgraded to 1.4.44 with the latest DAHDI version. I've noticed that if one of the dialplan PBXes calls Congestion(), the PRI will return ISDN code 34 (as its supposed to do). However, the issue is that subsequent calls into that PRI channel are immediately responded by a Code 44 (channel not available) even though there is no live call on the channel. Has anyone else experienced this behavior? Its a pretty crippling behavior since all of our channels eventually become unresponsive until a 'dahdi restart' is issued. Thanks. -- James I believe that what you are describing is a very old bug, which is fixed somewhere in the 1.8 timeline when the interface between DAHDI and Asterisk is changed slightly. I encountered the same issue some time ago. I do not recall the exact conditions under which the issue happens, but I believe it is the attempt to cancel an unanswered inbound call with a specific subset of cause codes. If you are using an older Asterisk version, the only workaround is to use Playtones + Hangup() instead of sending the Congestion() or Busy() cause codes. Regards, Steve Thanks Steve. It must have been introduced between DAHDI 2.4.0 and 2.6.1 or between Asterisk 1.4.35 and 1.4.44. I had a box running Asterisk 1.4.35 + DAHDI 2.4.0 and I never had any issues. -- James -- _ -- 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] Congestion() forcing PRI channels to be not available
Hi, I have a PSTN Asterisk box that's connected to other dialplan PBXes through IAX2. Recently this box was upgraded to 1.4.44 with the latest DAHDI version. I've noticed that if one of the dialplan PBXes calls Congestion(), the PRI will return ISDN code 34 (as its supposed to do). However, the issue is that subsequent calls into that PRI channel are immediately responded by a Code 44 (channel not available) even though there is no live call on the channel. Has anyone else experienced this behavior? Its a pretty crippling behavior since all of our channels eventually become unresponsive until a 'dahdi restart' is issued. Thanks. -- James -- _ -- 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] Static on calls - v1.8.15.0
Hi, I'm testing out a server with asterisk 1.8.15.0 on it. I'm experiencing static occurring on almost 90% of calls on this particular server. All test phones are using SIP, and calls to/from PSTN servers are delivered using IAX2. I have other production servers running 1.4.x that do not have this issue that use the same PSTN connections. I haven't seen any ethernet errors or anything like that. Load is minimal since this is still a test server. The server itself has 16GB of RAM and Dual Quad Core Xeon E5345s. I'm sort of baffled as to where to start looking for the root cause of this issue, but it appears to be isolated to only this machine. Thanks. -- _ -- 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] Static on calls - v1.8.15.0
On Thu, Nov 8, 2012 at 8:47 AM, Richard Mudgett rmudg...@digium.com wrote: I'm testing out a server with asterisk 1.8.15.0 on it. I'm experiencing static occurring on almost 90% of calls on this particular server. All test phones are using SIP, and calls to/from PSTN servers are delivered using IAX2. I have other production servers running 1.4.x that do not have this issue that use the same PSTN connections. I haven't seen any ethernet errors or anything like that. Load is minimal since this is still a test server. The server itself has 16GB of RAM and Dual Quad Core Xeon E5345s. I'm sort of baffled as to where to start looking for the root cause of this issue, but it appears to be isolated to only this machine. You might have an A-law/u-law mismatch in the audio path. That kind of mismatch sounds like static on the line. Hmm, would translation from ulaw - gsm cause that as well? I noticed in 1.8 apparently iax2 allow=ulaw is off... -- James -- _ -- 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] Forcing SIP trunk matching order?
On Mon, Jul 2, 2012 at 12:13 AM, Olle E. Johansson o...@edvina.net wrote: No. This is probably because you are using phone numbers as names of devices with type=friend in sip.conf. That's generally a bad idea. The SIP channel matches an incoming call this way: 1. Take the From: user name and match with the list of type=user and type=friend 2. Take the sender's IP and port and match with the list of peers 3. Send the call to the context defined in the [general] section of sip conf In Asterisk 1.4 and hopefully 1.8 the last peer in sip.conf will match first. In 1.8 the internal strcutures was changed, but I hope that this functionality was retained. We had a dicussion about it, but I personally haven't tested the result. One needs to know the matching order, so if 1.8 doesn't behave that way, we need to fix it. The recommended way is to NOT use anything that likely will end up as a caller ID as names of devices in sip.conf. The name is whatever you have within square brackets above definitions of type=friend or type=user. The username= option is another option, not the name of the device. The quick way to solve your problems is to stop using type=friend and start using type=peer instead. Hi Ollie, You are correct, I do have callerID-type names as accounts in sip.conf. The hosts are set to dynamic. Is this a problem with type=peer? Would the deny/allow suggestion posted earlier also work with type=friend? Thanks. -- James -- _ -- 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] VoIP Company looking for Asterisk/VoIP Engineer
Hi, I work for a VoIP provider in Southern California. We are looking for someone very knowledgeable in Asterisk/VoIP to help work on the following: - Maintenance of current Asterisk servers, updating Asterisk, monitoring load, and other sysadmin tasks - Devise and implement scalability strategies so that adding additional capacity is easy and does not compromise anything about the current system - Troubleshooting call quality issuses through our network (jitter, audio dropouts..) Candidates should have the following experience: - Minimum 3 years working with VoIP/Asterisk - Have worked in an environment with a significant number of phones (500) - Experience working with Cisco networking devices - QoS knowledge is a huge plus. Having experience with VoIP over carrier-class wireless links is a definite plus. This is a part-time contractor position. We are located in Southern California, and while having someone local would be ideal, telecommuting is an option. Hourly rate DOE. Please email all resumes directly to me at jlama...@gmail.com Thank you. -- _ -- 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] Restart single dahdi span
Hi, Is it possible yet to restart a single Dahdi span in any version of Asterisk? (instead of all of them) Thanks. -- James -- _ -- 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] Asterisk 1.4.42 NOTIFY replies ignore NAT setting
Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back to that port, which is incorrect. (Sonicwalls and other routers love to do this, even with Symmetric NAT on). The reason for this is that the NOTIFY reply does not attempt to lookup the SIP peer and check its NAT flags. I've seen some nasty From: header string parsing code + find_peer() that does this, but I was wondering if there's an easier way. Thanks. -- James -- _ -- 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] Asterisk 1.4.42 NOTIFY replies ignore NAT setting
On Fri, Dec 30, 2011 at 6:02 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 12/30/2011 04:07 AM, James Lamanna wrote: Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back to that port, which is incorrect. (Sonicwalls and other routers love to do this, even with Symmetric NAT on). The reason for this is that the NOTIFY reply does not attempt to lookup the SIP peer and check its NAT flags. I've seen some nasty From: header string parsing code + find_peer() that does this, but I was wondering if there's an easier way. Since Asterisk does not initiate subscriptions, these NOTIFY requests arriving to the Asterisk system must be 'unsolicited'. As such, they don't have an associated SIP dialog structure, so there's no simple way to know whether they are associated with a known peer or not. You say that Asterisk's behavior is 'incorrect', but it's only 'incorrect' because you believe it should be looking up any associated peer and using that peer's NAT setting; Asterisk's behavior as you've quoted is *correct* according to the RFC3261 rules for how replies should be sent, assuming that the top-most Via header does not have an 'rport' parameter present in it. The *proper* way to solve this problem is to have the UA sending the NOTIFY request include the 'rport' parameter in the top-most Via header of the request; if that is done, then whatever UA receives the request will be able to properly respond, even if the request crosses a NAT. Another way to solve it, if the sending UA cannot be changed to emit proper SIP requests, is to modify Asterisk to attempt a peer lookup when it is going to reply to request that it cannot associate with any known dialog, and then have the peer configured with 'nat=yes' (in the case of 1.4.42). A third option is to set 'nat=yes' in the [general] section of sip.conf, so that Asterisk will reply using rport-style behavior regardless of whether the request could be associated with a peer or not. Thanks Kevin. I'll have to turn rport on on all my Linksys/Cisco phones and give it a shot. -- James -- _ -- 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] Asterisk 1.4.42 NOTIFY replies ignore NAT setting
On Fri, Dec 30, 2011 at 8:35 AM, James Lamanna jlama...@gmail.com wrote: On Fri, Dec 30, 2011 at 6:02 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 12/30/2011 04:07 AM, James Lamanna wrote: Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back to that port, which is incorrect. (Sonicwalls and other routers love to do this, even with Symmetric NAT on). The reason for this is that the NOTIFY reply does not attempt to lookup the SIP peer and check its NAT flags. I've seen some nasty From: header string parsing code + find_peer() that does this, but I was wondering if there's an easier way. Since Asterisk does not initiate subscriptions, these NOTIFY requests arriving to the Asterisk system must be 'unsolicited'. As such, they don't have an associated SIP dialog structure, so there's no simple way to know whether they are associated with a known peer or not. You say that Asterisk's behavior is 'incorrect', but it's only 'incorrect' because you believe it should be looking up any associated peer and using that peer's NAT setting; Asterisk's behavior as you've quoted is *correct* according to the RFC3261 rules for how replies should be sent, assuming that the top-most Via header does not have an 'rport' parameter present in it. The *proper* way to solve this problem is to have the UA sending the NOTIFY request include the 'rport' parameter in the top-most Via header of the request; if that is done, then whatever UA receives the request will be able to properly respond, even if the request crosses a NAT. Another way to solve it, if the sending UA cannot be changed to emit proper SIP requests, is to modify Asterisk to attempt a peer lookup when it is going to reply to request that it cannot associate with any known dialog, and then have the peer configured with 'nat=yes' (in the case of 1.4.42). A third option is to set 'nat=yes' in the [general] section of sip.conf, so that Asterisk will reply using rport-style behavior regardless of whether the request could be associated with a peer or not. Thanks Kevin. I'll have to turn rport on on all my Linksys/Cisco phones and give it a shot. Hi Kevin, That doesn't appear to work correctly: The response does not come back to 34972 even though rport is in the Via. U xxx.234:34972 - yyy..7:5060 NOTIFY sip:yyy.7 SIP/2.0..Via: SIP/2.0/UDP 10.132.38.19:6957;branch=z9hG4bK-25ea41f0;rport..From: 1316 sip:1316@yyy.7;tag=80f427ae9e884ado0..To: sip:yyy .7..Call-ID: 4fa38a62-b7d76...@10.132.38.19..cseq: 1 NOTIFY..Max-Forwards: 70..Contact: 1316 sip:1316@10.132.38.19:6957..Event: keep-alive..User-Agent: Linksys/SPA942-6.1.3( a)..Content-Length: 0 # U yyy.7:5060 - xxx.234:6957 SIP/2.0 481 No subscription..Via: SIP/2.0/UDP 10.132.38.19:6957;branch=z9hG4bK-25ea41f0;received=xxx.234;rport=34972..From: 1316 sip:1316@yyy.7;tag=80f427ae9e884 ado0..To: sip:yyy.7;tag=as07ad17b5..Call-ID: 4fa38a62-b7d76...@10.132.38.19..cseq: 1 NOTIFY..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTI FY, INFO..Supported: replaces..Content-Length: 0 -- _ -- 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] Asterisk 1.4.42 NOTIFY replies ignore NAT setting
On Fri, Dec 30, 2011 at 11:55 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 12/30/2011 12:29 PM, James Lamanna wrote: On Fri, Dec 30, 2011 at 8:35 AM, James Lamannajlama...@gmail.com wrote: On Fri, Dec 30, 2011 at 6:02 AM, Kevin P. Flemingkpflem...@digium.com wrote: On 12/30/2011 04:07 AM, James Lamanna wrote: Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back to that port, which is incorrect. (Sonicwalls and other routers love to do this, even with Symmetric NAT on). The reason for this is that the NOTIFY reply does not attempt to lookup the SIP peer and check its NAT flags. I've seen some nasty From: header string parsing code + find_peer() that does this, but I was wondering if there's an easier way. Since Asterisk does not initiate subscriptions, these NOTIFY requests arriving to the Asterisk system must be 'unsolicited'. As such, they don't have an associated SIP dialog structure, so there's no simple way to know whether they are associated with a known peer or not. You say that Asterisk's behavior is 'incorrect', but it's only 'incorrect' because you believe it should be looking up any associated peer and using that peer's NAT setting; Asterisk's behavior as you've quoted is *correct* according to the RFC3261 rules for how replies should be sent, assuming that the top-most Via header does not have an 'rport' parameter present in it. The *proper* way to solve this problem is to have the UA sending the NOTIFY request include the 'rport' parameter in the top-most Via header of the request; if that is done, then whatever UA receives the request will be able to properly respond, even if the request crosses a NAT. Another way to solve it, if the sending UA cannot be changed to emit proper SIP requests, is to modify Asterisk to attempt a peer lookup when it is going to reply to request that it cannot associate with any known dialog, and then have the peer configured with 'nat=yes' (in the case of 1.4.42). A third option is to set 'nat=yes' in the [general] section of sip.conf, so that Asterisk will reply using rport-style behavior regardless of whether the request could be associated with a peer or not. Thanks Kevin. I'll have to turn rport on on all my Linksys/Cisco phones and give it a shot. Hi Kevin, That doesn't appear to work correctly: The response does not come back to 34972 even though rport is in the Via. U xxx.234:34972 - yyy..7:5060 NOTIFY sip:yyy.7 SIP/2.0..Via: SIP/2.0/UDP 10.132.38.19:6957;branch=z9hG4bK-25ea41f0;rport..From: 1316 sip:1316@yyy.7;tag=80f427ae9e884ado0..To:sip:yyy .7..Call-ID: 4fa38a62-b7d76...@10.132.38.19..cseq: 1 NOTIFY..Max-Forwards: 70..Contact: 1316 sip:1316@10.132.38.19:6957..Event: keep-alive..User-Agent: Linksys/SPA942-6.1.3( a)..Content-Length: 0 # U yyy.7:5060 - xxx.234:6957 SIP/2.0 481 No subscription..Via: SIP/2.0/UDP 10.132.38.19:6957;branch=z9hG4bK-25ea41f0;received=xxx.234;rport=34972..From: 1316sip:1316@yyy.7;tag=80f427ae9e884 ado0..To:sip:yyy.7;tag=as07ad17b5..Call-ID: 4fa38a62-b7d76...@10.132.38.19..cseq: 1 NOTIFY..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTI FY, INFO..Supported: replaces..Content-Length: 0 That would be a bug then; the 481 response was not sent to the proper port. It's strange though, because the rport parameter was properly updated with the 'perceived port', and the received parameter was added as well. Could this be because this is sent through a temporary' response, rather than the traditional allocation? (it uses transmit_response_using_temp) Thanks. -- James -- _ -- 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] DTMF between sip trunks and PRIs
Hi, I'm looking for some advice on how to solve DTMF issues. I have 2 boxes, one which is the connection to the PSTN (PSTN) through PRIs and SIP trunks, and a second (PBX) which has UAs registered to it. We have a customer that has an existing pbx that we trunk analog lines to using a GXW-4008. The GXW is set to dtmfmode inband. This seems to provide the best outbound DTMF. The issue I'm currently having is with inbound DTMF. PBX and PSTN are connected through a standard sip trunk. Both machines are on the same physical switch. Here are the results I've seen: PBX - PSTN using rfc2833 | Incoming call on PRI | DTMF on pbx voicemail system fails (dup/missing digits) PBX - PSTN using inband | Incoming call on PRI | DTMF on pbx voicemail system is correct PBX - PSTN using rfc2833 | Incoming call on SIP | DTMF on pbx voicemail system is correct PBX - PSTN using inband | Incoming call on SIP | DTMF on pbx voicemail system is correct All asterisk versions are 1.4.35. PRI card is a Sangoma A104 with HW DTMF detection. Does asterisk just have a problem converting the DTMF from the D-channel to rfc2833? The DTMF log looks ok (I dialed '642'), so I'm not sure where the issue is coming in. [Jul 4 21:05:44] DTMF[9769] channel.c: DTMF begin '6' received on Zap/15-1 [Jul 4 21:05:44] DTMF[9769] channel.c: DTMF begin passthrough '6' on Zap/15-1 [Jul 4 21:05:44] DTMF[9769] channel.c: DTMF end '6' received on Zap/15-1, duration 100 ms [Jul 4 21:05:44] DTMF[9769] channel.c: DTMF end accepted with begin '6' on Zap/15-1 [Jul 4 21:05:44] DTMF[9769] channel.c: DTMF end passthrough '6' on Zap/15-1 [Jul 4 21:05:45] DTMF[9769] channel.c: DTMF begin '4' received on Zap/15-1 [Jul 4 21:05:45] DTMF[9769] channel.c: DTMF begin passthrough '4' on Zap/15-1 [Jul 4 21:05:45] DTMF[9769] channel.c: DTMF end '4' received on Zap/15-1, duration 100 ms [Jul 4 21:05:45] DTMF[9769] channel.c: DTMF end accepted with begin '4' on Zap/15-1 [Jul 4 21:05:45] DTMF[9769] channel.c: DTMF end passthrough '4' on Zap/15-1 [Jul 4 21:05:45] DTMF[9769] channel.c: DTMF begin '2' received on Zap/15-1 [Jul 4 21:05:45] DTMF[9769] channel.c: DTMF begin passthrough '2' on Zap/15-1 [Jul 4 21:05:46] DTMF[9769] channel.c: DTMF end '2' received on Zap/15-1, duration 100 ms [Jul 4 21:05:46] DTMF[9769] channel.c: DTMF end accepted with begin '2' on Zap/15-1 [Jul 4 21:05:46] DTMF[9769] channel.c: DTMF end passthrough '2' on Zap/15-1 Thanks. -- James -- _ -- 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] queue_log in MySQL database
Hi Jonas, On Thu, Jan 13, 2011 at 8:19 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, can /var/log/messages/queue_log be saved in a MySQL database ?? So it would be easier to work with... I don't think Asterisk has this support built-in...maybe 1.8 does? However, what I do to manage queue_log is I have a small daemon that I have written in Python that watches the queue_log file, parses each incoming line, and stores it in a MySQL table. -- James -- _ -- 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] DTMF not being heard correctly by far end conference system
Hi Duncan, On Wed, Jan 12, 2011 at 10:13 AM, Duncan Turnbull dun...@e-simple.co.nz wrote: Hi Thorsten Thanks very much, at this point my preference is rfc2833 but I will try some other options. The system is generating audible tones (that I can hear), although I think the audio is generated by the last sip device in the network so if thats so I don't have any control of it. Probably then I have to go to inband to get some control back, I am not sure what I lose from this, or change upstream provider (although the current provider works from a different system) In my DTMF experience I have found a few IVRs and conference systems out there that won't accept my DTMF, even though its DTMF that I can see going out over PRI channels. My guess is that these systems use too tight of a duration window on their DTMF detectors. In your case I'm guessing that for some reason the SIP DTMF tones are coming out with too short of a duration. I believe you can fiddle with the dtmf tone duration and spacing in channel.c but I don't know if that will fix the issue. Is it possible to get the DTMF specs from the manufacturer of the conference system? -- James -- _ -- 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] Failed SIP registration kicks registered device off?
HI Ye, On Mon, Jan 10, 2011 at 10:04 AM, Ye Liu jaux...@gmail.com wrote: Hi folks, I'm currently running a modified version of Asterisk 1.6.1.1, I observed an unexpected behavior of my system today: 1. SIP device A successfully registered extension 100; 2. SIP device B tried to register extension 100 but with wrong password, so registration failed; 3. A then showed it was unregistered! Failed registration of device B shouldn't kick A off, I expect A stay online and work properly in this situation. Could anyone confirm this? Because my asterisk is modified, I'm not sure this behavior is in vanilla asterisk or it is caused by my own code. AFAIK, Asterisk does not support simultaneous registration from more than one device on the same extension. That is why you are seeing this behavior. As soon as B tries to register, the registration of A is 'overwritten'. If you need this behavior, you might want to try and look into a different UA Registrar like OpenSIPS, which supports this. Thank you! -- Ye Liu (AKA @jaux) -- James -- _ -- 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] Asterisk replying to wrong port for NOTIFY messages
Hi Jeff, On Thu, Jan 6, 2011 at 11:28 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 5 Jan 2011, James Lamanna wrote: See the following SIP trace. Where in the world does Asterisk get port 1025 to respond to? This is asterisk 1.6.x. Hi James, I'm sure it would be the NAT translated port on the public side of the customer's firewall... Unfortunately its not. All clients are on symmetric NAT. Here's an ngrep trace, you can see the NAT port in the VIA is the same as the source port: U xxx.xxx.xxx.44:8155 - xx.xxx.xxx.7:5060 NOTIFY sip:pbx1.warp2biz.com SIP/2.0..Via: SIP/2.0/UDP 192.168.1.127:8155;branch=z9hG4bK-4b50c77d..From: zz sip:zzz...@pbx1.example.com;tag=5281a88170274fa2o0..To: sip:pbx1.example.com..Call-ID: c914b8d-532f2...@192.168.1.127..cseq: 14492 NOTIFY..Max-Forwards: 70..Con tact: zz sip:zzz...@192.168.1.127:8155..Event: keep-alive..User-Agent: Cisco/SPA509G-7.4.6-0002fdff9097..Content-Length: 0 # U xx.xxx.xxx.7:5060 - xx.xxx.xxx.44:1025 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.127:8155;branch=z9hG4bK-4b50c77d;received=xx.xxx.xxx.44..From: zz sip:zzz...@pbx1.example.com;tag=5281a88170274fa2o0..To: sip:pbx1.example.com;tag=as62dac391..Call-ID: c914b8d-532f2...@192.168.1.127..cseq: 14492 NOTIFY..User- Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Content-Length: 0 -- James j Thanks. -- James --- SIP read from zzz.zzz.zzz.44:9363 --- NOTIFY sip:pbx1.mydomain.com SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3^M From: xxx-xxx- sip:xxx...@pbx1.mydomain.com;tag=467525dd6fac949do0^M To: sip:pbx1.mydomain.com^M Call-ID: 707176dd-38f47...@192.168.1.140^m CSeq: 118907 NOTIFY^M Max-Forwards: 70^M Contact: xxx-xxx- sip:xx...@192.168.1.140:9363^M Event: keep-alive^M User-Agent: Cisco/SPA509G-7.4.6-0002fdff90a4^M Content-Length: 0^M ^M - [Jan 5 13:46:36] VERBOSE[3919] logger.c: --- (11 headers 0 lines) --- [Jan 5 13:46:36] VERBOSE[3919] logger.c: --- Transmitting (no NAT) to zzz.zzz.zzz.44:1025 --- SIP/2.0 200 OK^M Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3;received=zzz.zzz.zzz.44^M From: xxx-xxx- sip:xx...@pbx1.mydomain.com;tag=467525dd6fac949do0^M To: sip:pbx1.mydomain.com;tag=as0493c604^M Call-ID: 707176dd-38f47...@192.168.1.140^m CSeq: 118907 NOTIFY^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO^M Supported: replaces^M Content-Length: 0^M -- _ -- 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
[asterisk-users] Asterisk replying to wrong port for NOTIFY messages
See the following SIP trace. Where in the world does Asterisk get port 1025 to respond to? This is asterisk 1.6.x. Thanks. -- James --- SIP read from zzz.zzz.zzz.44:9363 --- NOTIFY sip:pbx1.mydomain.com SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3^M From: xxx-xxx- sip:xxx...@pbx1.mydomain.com;tag=467525dd6fac949do0^M To: sip:pbx1.mydomain.com^M Call-ID: 707176dd-38f47...@192.168.1.140^m CSeq: 118907 NOTIFY^M Max-Forwards: 70^M Contact: xxx-xxx- sip:xx...@192.168.1.140:9363^M Event: keep-alive^M User-Agent: Cisco/SPA509G-7.4.6-0002fdff90a4^M Content-Length: 0^M ^M - [Jan 5 13:46:36] VERBOSE[3919] logger.c: --- (11 headers 0 lines) --- [Jan 5 13:46:36] VERBOSE[3919] logger.c: --- Transmitting (no NAT) to zzz.zzz.zzz.44:1025 --- SIP/2.0 200 OK^M Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3;received=zzz.zzz.zzz.44^M From: xxx-xxx- sip:xx...@pbx1.mydomain.com;tag=467525dd6fac949do0^M To: sip:pbx1.mydomain.com;tag=as0493c604^M Call-ID: 707176dd-38f47...@192.168.1.140^m CSeq: 118907 NOTIFY^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO^M Supported: replaces^M Content-Length: 0^M -- _ -- 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] DAHDI 2.4.0 produces HDLC errors with echo canceler
On Mon, Nov 29, 2010 at 10:02 AM, Shaun Ruffell sruff...@digium.com wrote: On 11/27/2010 11:03 AM, James Lamanna wrote: Hi, After upgrading to DAHDI 2.4.0 from Zaptel, I noticed a lot of HDLC errors on my console: [Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 [Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 These errors prevented calls from being made and received on my PRI spans. This seems similar to bug 15498: https://issues.asterisk.org/view.php?id=15498 Which says this was fixed in 2.2...so maybe it got back into 2.4? I can get rid of the errors by disabling the mg2 echo canceler in /etc/dadhi/system.conf. Do you have a hardware echocan module installed on your card? If so, it's strange indeed that the error goes away when you disable mg2. What is the complete output of your /etc/dahdi/system.conf? Nope, No h/w echo canceler on this card. Here's system.conf: span=1,1,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 # Global data loadzone= us defaultzone = us -- James -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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
[asterisk-users] DAHDI 2.4.0 produces HDLC errors with echo canceler
Hi, After upgrading to DAHDI 2.4.0 from Zaptel, I noticed a lot of HDLC errors on my console: [Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 [Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 These errors prevented calls from being made and received on my PRI spans. This seems similar to bug 15498: https://issues.asterisk.org/view.php?id=15498 Which says this was fixed in 2.2...so maybe it got back into 2.4? I can get rid of the errors by disabling the mg2 echo canceler in /etc/dadhi/system.conf. The PRI card I'm using is a Digium TE122. I'd prefer not having to run with the echo canceler off of course... [1] active=yes alarms=OK description=Wildcard TE122 Card 0 name=WCT1/0 manufacturer=Digium devicetype=Wildcard TE122 location=PCI Bus 03 Slot 03 basechan=1 totchans=24 irq=35 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF Thanks. -- James -- _ -- 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] SIP NOTIFY to make linksys/cisco SPA BLF go yellow
Hi, I was wondering if anyone stumbled upon the correct event in a sip NOTIFY (from a SUBSCRIBE) to make the BLF lamps on a Linksys/Cisco SPA9xx/5xx go yellow? I'm trying to differentiate between On the Phone and DND with the BLF. Thanks. -- James -- _ -- 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] Polling DND status of a Linksys SPA9xx/5xx phone?
Hi, Is there a way to poll the DND status of a Linksys SPA9xx/5xx phone? The reason I ask is that I'm trying to implement DND + BLF on asterisk. However, the DND softkey on the Linksys phone does not send any feature codes to asterisk. On the flip side, if you disable the Vertical Activation Codes on the phone, then dialing the feature code doesn't display 'Do Not Disturb' on the phone. What I need is an indication on the phone that it is on DND, AND an indication through BLF to other users that a particular phone is on DND. Any ideas? Thanks. -- James -- _ -- 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] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)
On Tue, Jun 22, 2010 at 6:33 PM, Ryan Wagoner rswago...@gmail.com wrote: -- The out of dialog support was the trick for 1.6.2.9 since it has support for sending a keep-alive. I have attached a modified version of your patch that worked for me. Do you mind if I attach the modified version of the patch to my issue report? Hey Ryan, I have no problems with that, go right ahead. -- James Ryan -- _ -- 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] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)
On Tue, Jun 22, 2010 at 8:57 PM, Andres and...@telesip.net wrote: completely as well. Below I've posted a patch that responds with a 200 OK to these keep-alive requests, and I believe also solves the temporary loss of registration problem, though more testing in different environments for those who experience this problem would be greatly appreciated. The patch is against 1.4.32. A workaround we have used for a long time is to simply change the config on the Linksys phones to send an empty packet as a keep-alive. There is obviously no response from asterisk but it keeps the NAT bindings alive and well on every router we have tested. Hi Andres, I have noticed that on Linksys phones that have a short REGISTER time, the lack of NAT keep alive responses can cause the phone to no longer be able to register. That's why I've spent a lot of effort to hopefully make these keep-alives supported. Andres http://www.neuroredes.com -- James -- James -- _ -- 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] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)
If you've used Linksys phones against recent Asterisk 1.4.x you may have noticed that they may drop registration for a quick bit and then go back to being ok if your phone is behind NAT. If you turn Asterisk's sip debug information on, you'll probably find errors like these in your logs: NOTICE[30179] chan_sip.c: Correct auth, but based on stale nonce received from '11 sip:999...@208.90.186.10' I believe I have determined that this is caused by a bug in the Linksys firmware that is related to the NAT Keep-Alive packets. Because recent Asterisk 1.4.x's do not establish a SIP dialog for NOTIFY requests, the 489 Bad Event replies were going back to the wrong address if your phone was behind NAT. This lack of reply would cause the next REGISTER message to use the same nonce as the previous REGISTER, resulting in the stale nonce errors and temporarily dropping registration. I've also seen the lack of response to the NAT keep-alive cause the phone to stop being able to register completely as well. Below I've posted a patch that responds with a 200 OK to these keep-alive requests, and I believe also solves the temporary loss of registration problem, though more testing in different environments for those who experience this problem would be greatly appreciated. The patch is against 1.4.32. -- James keep_alive_fix.diff Description: Binary data -- _ -- 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] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)
On Tue, Jun 22, 2010 at 12:06 PM, Stefan Schmidt s...@sil.at wrote: James Lamanna schrieb: If you've used Linksys phones against recent Asterisk 1.4.x you may have noticed that they may drop registration for a quick bit and then go back to being ok if your phone is behind NAT. If you turn Asterisk's sip debug information on, you'll probably find errors like these in your logs: NOTICE[30179] chan_sip.c: Correct auth, but based on stale nonce received from '11 sip:999...@208.90.186.10' I believe I have determined that this is caused by a bug in the Linksys firmware that is related to the NAT Keep-Alive packets. Because recent Asterisk 1.4.x's do not establish a SIP dialog for NOTIFY requests, the 489 Bad Event replies were going back to the wrong address if your phone was behind NAT. This lack of reply would cause the next REGISTER message to use the same nonce as the previous REGISTER, resulting in the stale nonce errors and temporarily dropping registration. I've also seen the lack of response to the NAT keep-alive cause the phone to stop being able to register completely as well. Below I've posted a patch that responds with a 200 OK to these keep-alive requests, and I believe also solves the temporary loss of registration problem, though more testing in different environments for those who experience this problem would be greatly appreciated. The patch is against 1.4.32. -- James Hello, you also just could set the NAT KEEP ALIVE MESSAGE on Ext 1 from $NOTIFY to $OPTIONS and make this extension in your default context: exten = s,1,hangup and you also would get a 200 ok for the keep alive package. IMHO a stale nonce would only occur when a user tries to register faster than 3600s cause of the register timeout used in asterisk. Maybe you should also try to set a higher register timeout on your phones. but i dont have an 1.4 system running, only around 2k of linksys phones on a 1.2.40 and 300 on 1.6.1.18 and i dont see this problem there. I'm not sure how this works. The OPTIONS message fails chan_sip.c:parse_request() so the OPTIONS message never gets processed. The options message I receive from a Linksys942 6.1.3(a) looks like this: --- SIP read from xxx.xxx.xxx.xxx:8037 --- OPTIONS - -- James -- _ -- 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] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)
On Tue, Jun 22, 2010 at 4:31 PM, Ryan Wagoner rswago...@gmail.com wrote: On Tue, Jun 22, 2010 at 6:26 PM, James Lamanna jlama...@gmail.com wrote: On Tue, Jun 22, 2010 at 12:06 PM, Stefan Schmidt s...@sil.at wrote: James Lamanna schrieb: If you've used Linksys phones against recent Asterisk 1.4.x you may have noticed that they may drop registration for a quick bit and then go back to being ok if your phone is behind NAT. If you turn Asterisk's sip debug information on, you'll probably find errors like these in your logs: NOTICE[30179] chan_sip.c: Correct auth, but based on stale nonce received from '11 sip:999...@208.90.186.10' I believe I have determined that this is caused by a bug in the Linksys firmware that is related to the NAT Keep-Alive packets. Because recent Asterisk 1.4.x's do not establish a SIP dialog for NOTIFY requests, the 489 Bad Event replies were going back to the wrong address if your phone was behind NAT. This lack of reply would cause the next REGISTER message to use the same nonce as the previous REGISTER, resulting in the stale nonce errors and temporarily dropping registration. I've also seen the lack of response to the NAT keep-alive cause the phone to stop being able to register completely as well. Below I've posted a patch that responds with a 200 OK to these keep-alive requests, and I believe also solves the temporary loss of registration problem, though more testing in different environments for those who experience this problem would be greatly appreciated. The patch is against 1.4.32. -- James Hello, you also just could set the NAT KEEP ALIVE MESSAGE on Ext 1 from $NOTIFY to $OPTIONS and make this extension in your default context: exten = s,1,hangup and you also would get a 200 ok for the keep alive package. IMHO a stale nonce would only occur when a user tries to register faster than 3600s cause of the register timeout used in asterisk. Maybe you should also try to set a higher register timeout on your phones. but i dont have an 1.4 system running, only around 2k of linksys phones on a 1.2.40 and 300 on 1.6.1.18 and i dont see this problem there. I'm not sure how this works. The OPTIONS message fails chan_sip.c:parse_request() so the OPTIONS message never gets processed. The options message I receive from a Linksys942 6.1.3(a) looks like this: --- SIP read from xxx.xxx.xxx.xxx:8037 --- OPTIONS - -- James -- I had the same result when using $OPTIONS on a SPA941 phone with firmware 5.1.8. I am running Asterisk 1.6.2.9 that has the keep-alive support, however I still see Asterisk sending a 489 Bad Event. I just reopened the issue and provided the necessary debug log at https://issues.asterisk.org/bug_view_page.php?bug_id=17379 Ryan, This is most likely because the packet never makes it to handle_request_notify. I haven't looked at the code for 1.6.2.9 yet, but in 1.4.32 without my patch, the NOTIFY request would never make it out of find_call() and return early with a 489 Bad Event response. Were you getting any response at 1.6.2.9 with the OPTIONS message? -- James -- _ -- 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] Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)
On Sun, Jun 20, 2010 at 5:42 AM, Ryan Wagoner rswago...@gmail.com wrote: On Sat, Jun 19, 2010 at 12:00 PM, James Lamanna jlama...@gmail.com wrote: On Fri, Jun 18, 2010 at 10:51 PM, Stefan Schmidt s...@sil.at wrote: James Lamanna schrieb: It appears as though the 489 Bad Event response to the NAT keep alive event responds to the local address, instead of responding to the NATted address. This causes Linksys phones to go amber (no registration) after a short amount of time after placing calls. Turning the Linksys NAT keep alive off is a workound, but non-ideal in may situations. Apparently the asterisk devs don't even think this is a bug: https://issues.asterisk.org/view.php?id=17532 Has anyone dealt with this at all? Thanks. -- James Hello james, in the SPA config webpage on EXT 1 in the Nat Keep Alive MSG you should set $OPTIONS instead of $NOTIFY. then in your asterisk extension default context just set this: exten = s,1,Hangup then the phone will send a options packet and you will get a 200 OK instead of 489 Bad event. this should help. best regards Thanks Steve, I'll give that a try. I think I'll also look into why responses to NOTIFYs don't do the right thing in terms of NAT either. steve -- James I have created an issue report on this a few weeks on with Asterisk 1.6.2.8-rc1. This was happening on a client site, which I didn't have a chance to stop back by, so they closed the issue. https://issues.asterisk.org/bug_view_page.php?bug_id=17379 It looked to me like Asterisk was rejecting the NOTIFY message due to no callid, which is in the message. I couldn't figure out what was going and there is code in 1.6.2.x to return a 200 OK to a NOTIFY message. Ryan Interesting. I'm still on the 1.4.x series (and I don't plan on upgrading until 1.8.x is out), but my issue, without the workaround that Steve suggested above, is that the NOTIFY Bad Event reply does not seem to respect NAT for some reason. Whether it doesn't look up the peer properties or what I'm not sure, but I plan on doing a thorough investigation with 1.4.32 this week to see what is indeed going on. Problems like this, and some other issues I've reported (where a channel can get stuck Up if a phone goes Unavailable while in Ringing), makes me lean more and more to moving to OpenSIPs for handling device registrations. -- James -- _ -- 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] Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)
On Fri, Jun 18, 2010 at 10:51 PM, Stefan Schmidt s...@sil.at wrote: James Lamanna schrieb: It appears as though the 489 Bad Event response to the NAT keep alive event responds to the local address, instead of responding to the NATted address. This causes Linksys phones to go amber (no registration) after a short amount of time after placing calls. Turning the Linksys NAT keep alive off is a workound, but non-ideal in may situations. Apparently the asterisk devs don't even think this is a bug: https://issues.asterisk.org/view.php?id=17532 Has anyone dealt with this at all? Thanks. -- James Hello james, in the SPA config webpage on EXT 1 in the Nat Keep Alive MSG you should set $OPTIONS instead of $NOTIFY. then in your asterisk extension default context just set this: exten = s,1,Hangup then the phone will send a options packet and you will get a 200 OK instead of 489 Bad event. this should help. best regards Thanks Steve, I'll give that a try. I think I'll also look into why responses to NOTIFYs don't do the right thing in terms of NAT either. steve -- James -- _ -- 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] Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)
It appears as though the 489 Bad Event response to the NAT keep alive event responds to the local address, instead of responding to the NATted address. This causes Linksys phones to go amber (no registration) after a short amount of time after placing calls. Turning the Linksys NAT keep alive off is a workound, but non-ideal in may situations. Apparently the asterisk devs don't even think this is a bug: https://issues.asterisk.org/view.php?id=17532 Has anyone dealt with this at all? Thanks. -- James SIP trace: U external.ip:9375 - asterisk.ip:5060 NOTIFY sip:asterisk.ip SIP/2.0..Via: SIP/2.0/UDP 10.10.30.65:9375;branch=z9hG4bK-8ebce8bc..From: xxx-xxx- sip:9497197...@asterisk.ip;tag=3a6a735864619b8bo0..To: sip:asterisk.ip..Call-ID: 19a0bd7 c-3cb13...@10.10.30.65..cseq: 395 NOTIFY..Max-Forwards: 70..Contact: xxx-xxx- sip:xxx...@10.10.30.65:9375..Event: keep-alive..User-Agent: Linksys/SPA942-6.1.3(a)-000e08d87445..Content-Length: 0 # U asterisk.ip:5060 - 10.10.30.65:9375 SIP/2.0 489 Bad event..Via: SIP/2.0/UDP 10.10.30.65:9375;branch=z9hG4bK-8ebce8bc;received=external.ip..From: xxx-xxx- sip:9497197...@asterisk.ip;tag=3a6a735864619b8bo0..To: sip:asterisk.ip;tag=as4a 4466b0..Call-ID: 19a0bd7c-3cb13...@10.10.30.65..cseq: 395 NOTIFY..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Content-Length: 0 -- _ -- 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] Small VoIP company looking for Asterisk Scalability and Maintenance Engineer
Hi, I work for a small VoIP provider in Southern California. We are looking for someone very knowledgeable in Asterisk to help work on the following: - Maintenance of current Asterisk servers, updating Asterisk, monitoring load, and other sysadmin tasks - Devise and implement high-availability strategies for the current servers including fail-over procedures - Devise and implement scalability strategies so that adding additional capacity is easy and does not compromise anything about the current system - Troubleshooting call routing issues through Asterisk by examining log files or other means Candidates should have the following experience: - Minimum 3 years working with Asterisk - Have worked in an environment with a significant number of phones (500) - Have already implemented strategies for scalability and high-availability in other environments. We are a small company, so you will have latitude as to what strategies you are able to implement, however any strategy must be written up and agreed upon by the team before any implementation starts. This is a part-time contractor position. We are located in Southern California, and while having someone local would be ideal, telecommuting is an option. Hourly rate DOE. Please email all resumes directly to me at jlama...@gmail.com Thank you. -- James -- _ -- 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] Extension state can get stuck in 'Ringing' state
Hi, I've noticed that if a phone goes UNREACHABLE while it is Ringing, when the phone comes back, Asterisk will not clear the channel that was created, so it still thinks it is in the Ringing state. The only way to clear this is to do a soft hangup on the SIP channel or to restart Asterisk. Unfortunately these issues are very hard to automatically track down and clear and it seems like if a phone goes UNREACHABLE, Asterisk should clear the channel anyways. This is at 1.4.26.2. I'm planning to upgrade to 1.4.31 shortly. I will see if I can replicate the problem in that version as well. Has anyone else noticed this? Thanks. -- James -- _ -- 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] OT: NAT in SPA922
On Thu, May 6, 2010 at 8:14 PM, Vineet Bhojnagarwala vbho...@gmail.com wrote: Alternatively, if using normal vlans, this can also be achieved by enabling access list on the switch and restrict traffic flows. Generally this is done on a layer 3 switch, don't think it will support on your switch model. That is correct. In order to do this on a 2950, you will need a router behind this to be the gateway for each vlan. (On Cisco equipment you'd need to create a subinterface for each vlan (i.e. FastEthernet 0.xxx) where xxx is your vlan number. Then you can set each port up to be a trunk port on the 2950, but specify the native vlan on the port as the PC vlan # and allow the Vlan # for the phone vlan. So something like: switchport mode trunk switchport trunk native vlan [pc vlan #] switchport trunk allowed vlan [pc vlan #],[phone vlan #] Then you will have to create access-lists on the router to block intra-VLAN traffic. This can also be all done on a Layer 3 switch (like the Cisco 3550), by defining each VLAN as an interface: interface VLAN 100 description Phone VLAN ip address 192.168.100.1 255.255.255.0 ! interface VLAN 101 description Customer 1 VLAN ip address 192.168.101.1 255.255.255.0 ! etc.. then your ports will look like: interface FastEthernet 0/2 description customer 1 port switchport mode trunk switchport trunk encapsulation dot1q switchport trunk native vlan 101 switchport trunk allowed vlan 100,101 ! Then you'll need access lists to prevent the intra-vlan traffic.. -- James Rgds, Vineet Bhojnagarwala RCDD, NTS, OSP Spear Networks Pvt Ltd Integration Consultancy +91-9831436607 On May 7, 2010, at 8:39 AM, Vineet Bhojnagarwala vbho...@gmail.com wrote: I think this is a motel kind of situation and a PVLAN serves the situation right. Put all the ipphones in the voice vlan as suggested, make a seperate isolated vlan for the PCs, this will restrict traffic between the clients. Rgds, Vineet Bhojnagarwala RCDD, NTS, OSP Spear Networks Pvt Ltd Integration Consultancy +91-9831436607 On May 6, 2010, at 11:30 PM, David White david.wh...@watchguard.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Noah Miller Sent: Thu 5/6/2010 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: NAT in SPA922 It is a building, with 24 separated rooms, each room will have a PC and a IP Phone. Every room connected to a switch Cisco 2950. I want keeping all PCs isolated behind a NAT (no access to neighbour's PC), and still keep communication in same LAN between all IP Phones. Should I take another approach on that? Put each PC in its own VLAN. Keep all the phones in one VLAN. Although having a $30 router in each room hanging off the phone would accomplish what you want also. Take j's suggestion to use VLANs. This is not a good situation for NAT. Cisco 2950's can do VLANs. to be clear, the only way this will work with the PCs is if each PC vlan is *also* a unique ip subnet (else how do all the vlans access a common default gw?) place the phones in a voice vlan, and the phone problem is solved. as for the PC isolation, you might get better feedback on a cisco or other networking forum. -david -- _ -- 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] OT: NAT in SPA922
On May 7, 2010, at 8:03, James Lamanna jlama...@gmail.com wrote: On Thu, May 6, 2010 at 8:14 PM, Vineet Bhojnagarwala vbho...@gmail.com wrote: Alternatively, if using normal vlans, this can also be achieved by enabling access list on the switch and restrict traffic flows. Generally this is done on a layer 3 switch, don't think it will support on your switch model. That is correct. In order to do this on a 2950, you will need a router behind this to be the gateway for each vlan. (On Cisco equipment you'd need to create a subinterface for each vlan (i.e. FastEthernet 0.xxx) where xxx is your vlan number. Then you can set each port up to be a trunk port on the 2950, but specify the native vlan on the port as the PC vlan # and allow the Vlan # for the phone vlan. So something like: switchport mode trunk switchport trunk native vlan [pc vlan #] switchport trunk allowed vlan [pc vlan #],[phone vlan #] Then you will have to create access-lists on the router to block intra-VLAN traffic. This can also be all done on a Layer 3 switch (like the Cisco 3550), by defining each VLAN as an interface: interface VLAN 100 description Phone VLAN ip address 192.168.100.1 255.255.255.0 ! interface VLAN 101 description Customer 1 VLAN ip address 192.168.101.1 255.255.255.0 ! etc.. then your ports will look like: interface FastEthernet 0/2 description customer 1 port switchport mode trunk switchport trunk encapsulation dot1q switchport trunk native vlan 101 switchport trunk allowed vlan 100,101 ! Then you'll need access lists to prevent the intra-vlan traffic.. I lied. You don't need access-lists in this case with the allowed vlan statement. -- James Rgds, Vineet Bhojnagarwala RCDD, NTS, OSP Spear Networks Pvt Ltd Integration Consultancy +91-9831436607 On May 7, 2010, at 8:39 AM, Vineet Bhojnagarwala vbho...@gmail.com wrote: I think this is a motel kind of situation and a PVLAN serves the situation right. Put all the ipphones in the voice vlan as suggested, make a seperate isolated vlan for the PCs, this will restrict traffic between the clients. Rgds, Vineet Bhojnagarwala RCDD, NTS, OSP Spear Networks Pvt Ltd Integration Consultancy +91-9831436607 On May 6, 2010, at 11:30 PM, David White david.wh...@watchguard.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Noah Miller Sent: Thu 5/6/2010 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: NAT in SPA922 It is a building, with 24 separated rooms, each room will have a PC and a IP Phone. Every room connected to a switch Cisco 2950. I want keeping all PCs isolated behind a NAT (no access to neighbour's PC), and still keep communication in same LAN between all IP Phones. Should I take another approach on that? Put each PC in its own VLAN. Keep all the phones in one VLAN. Although having a $30 router in each room hanging off the phone would accomplish what you want also. Take j's suggestion to use VLANs. This is not a good situation for NAT. Cisco 2950's can do VLANs. to be clear, the only way this will work with the PCs is if each PC vlan is *also* a unique ip subnet (else how do all the vlans access a common default gw?) place the phones in a voice vlan, and the phone problem is solved. as for the PC isolation, you might get better feedback on a cisco or other networking forum. -david -- _ -- 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
[asterisk-users] Registering a Cisco 7965 on 1.4.26
Hi, I'm having a problem trying to get a Cisco 7965 phone registered on Asterisk 1.4.26. As we know, Cisco now, for security reasons, has made the phone ports non-symmetric, in that it sends out UDP requests on a high port and receives them on a different port. It seems that, even with 'nat' set to 'no', that Asterisk is not honoring the Contact header and keeps attempting to send requests back to the high port number. I tried this on 1.6.0.9 with nat=no and everything works fine. Is this a problem with 1.4.26? Is there a 1.4.x version that works? Or did I manage to screw something up? Thanks. -- James -- _ -- 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] Registering a Cisco 7965 on 1.4.26
On Wed, May 5, 2010 at 10:16 AM, Danny Nicholas da...@debsinc.com wrote: Maybe a rtp.conf problem - normal values are 1-2. I haven't even gotten to the RTP stage, it won't even register on the SIP side because responses are being sent back to the wrong SIP signaling port. -- James -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Lamanna Sent: Wednesday, May 05, 2010 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Registering a Cisco 7965 on 1.4.26 Hi, I'm having a problem trying to get a Cisco 7965 phone registered on Asterisk 1.4.26. As we know, Cisco now, for security reasons, has made the phone ports non-symmetric, in that it sends out UDP requests on a high port and receives them on a different port. It seems that, even with 'nat' set to 'no', that Asterisk is not honoring the Contact header and keeps attempting to send requests back to the high port number. I tried this on 1.6.0.9 with nat=no and everything works fine. Is this a problem with 1.4.26? Is there a 1.4.x version that works? Or did I manage to screw something up? Thanks. -- James -- _ -- 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] ATA shootout: PAP2T versus Grandstream Handytone 286
It seems that the PAP2T does support TFTP and an XML-based config for deployments... I've used both the Grandstream 286 and the Linksys PAP2T. I have been able to get some limited faxing to work using T30 with a PAP2T. Configuration and provisioning of the Linksys is very easy through either the web GUI or XML configuration files, which can be transferred through TFTP or HTTP. I can only hope that Cisco will update the firmware of the PAP2T to support T38 one day... -- James -- _ -- 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] Duplicated DTMF with bridged IAX channels maybe?
On Wed, Apr 28, 2010 at 6:57 PM, James Lamanna jlama...@gmail.com wrote: Hi, I have a duplicated DTMF issue with, it appears, bridged IAX channels. I have the following setup: PRI IAX * PSTN ---* Dialplan I've configured a number on the dialplan server to make and outbound call to the pstn. This call then comes back into the dialplan server to SayDigits(). I'm seeing that a few of my digits are being duplicated every so often. I've attached an IAX trace from the PSTN server to this message where you can see the duplication (digits 9 3). The digits entered were 258963. [snip] Testing the following scenario: Call --(from pstn PRI)-- (PSTN box) --IAX-- (PBX box) --IAX-- (PSTN box) --(to/from pstn PRI)-- (PSTN box) Results in Duplication. Here are 2 traces from the PSTN box's DTMF log: '8' was not duplicated, '9' was. Zap/42-1 is the first inbound leg from the PSTN IAX2/w2bpstn-8399 is the leg from the PBX box to the PSTN box Zap/53-1 is the inbound leg from the PSTN [Apr 29 17:02:15] DTMF[16062] channel.c: DTMF begin '8' received on Zap/42-1 [Apr 29 17:02:15] DTMF[16062] channel.c: DTMF begin passthrough '8' on Zap/42-1 [Apr 29 17:02:15] DTMF[16065] channel.c: DTMF begin '8' received on IAX2/w2bpstn-8399 [Apr 29 17:02:15] DTMF[16065] channel.c: DTMF begin passthrough '8' on IAX2/w2bpstn-8399 [Apr 29 17:02:15] DTMF[16071] channel.c: DTMF begin '8' received on Zap/53-1 [Apr 29 17:02:15] DTMF[16071] channel.c: DTMF begin ignored '8' on Zap/53-1 [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF end '8' received on Zap/42-1, duration 63 ms [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF end accepted with begin '8' on Zap/42-1 [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF end '8' has duration 63 but want minimum 80, emulating on Zap/42-1 [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF end emulation of '8' queued on Zap/42-1 [Apr 29 17:02:16] DTMF[16065] channel.c: DTMF end '8' received on IAX2/w2bpstn-8399, duration 0 ms [Apr 29 17:02:16] DTMF[16065] channel.c: DTMF end accepted with begin '8' on IAX2/w2bpstn-8399 [Apr 29 17:02:16] DTMF[16065] channel.c: DTMF end passthrough '8' on IAX2/w2bpstn-8399 [Apr 29 17:02:16] DTMF[16071] channel.c: DTMF end '8' received on Zap/53-1, duration 223 ms [Apr 29 17:02:16] DTMF[16071] channel.c: DTMF end passthrough '8' on Zap/53-1 [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF begin '9' received on Zap/42-1 [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF begin passthrough '9' on Zap/42-1 [Apr 29 17:02:16] DTMF[16065] channel.c: DTMF begin '9' received on IAX2/w2bpstn-8399 [Apr 29 17:02:16] DTMF[16065] channel.c: DTMF begin passthrough '9' on IAX2/w2bpstn-8399 [Apr 29 17:02:16] DTMF[16071] channel.c: DTMF begin '9' received on Zap/53-1 [Apr 29 17:02:16] DTMF[16071] channel.c: DTMF begin ignored '9' on Zap/53-1 [Apr 29 17:02:16] DTMF[16071] channel.c: DTMF end '9' received on Zap/53-1, duration 223 ms [Apr 29 17:02:16] DTMF[16071] channel.c: DTMF end passthrough '9' on Zap/53-1 [Apr 29 17:02:16] DTMF[16071] channel.c: DTMF begin '9' received on Zap/53-1 [Apr 29 17:02:16] DTMF[16071] channel.c: DTMF begin ignored '9' on Zap/53-1 [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF end '9' received on Zap/42-1, duration 63 ms [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF end accepted with begin '9' on Zap/42-1 [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF end '9' has duration 63 but want minimum 80, emulating on Zap/42-1 [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF end emulation of '9' queued on Zap/42-1 [Apr 29 17:02:16] DTMF[16065] channel.c: DTMF end '9' received on IAX2/w2bpstn-8399, duration 0 ms [Apr 29 17:02:16] DTMF[16065] channel.c: DTMF end accepted with begin '9' on IAX2/w2bpstn-8399 [Apr 29 17:02:16] DTMF[16065] channel.c: DTMF end passthrough '9' on IAX2/w2bpstn-8399 [Apr 29 17:02:17] DTMF[16071] channel.c: DTMF end '9' received on Zap/53-1, duration 223 ms [Apr 29 17:02:17] DTMF[16071] channel.c: DTMF end passthrough '9' on Zap/53-1 -- James -- _ -- 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] Duplicated DTMF with bridged IAX channels maybe?
Hi, I have a duplicated DTMF issue with, it appears, bridged IAX channels. I have the following setup: PRI IAX * PSTN ---* Dialplan I've configured a number on the dialplan server to make and outbound call to the pstn. This call then comes back into the dialplan server to SayDigits(). I'm seeing that a few of my digits are being duplicated every so often. I've attached an IAX trace from the PSTN server to this message where you can see the duplication (digits 9 3). The digits entered were 258963. Thank you. -- James [Apr 28 18:43:42] VERBOSE[2806] logger.c: Tx-Frame Retry[000] -- OSeqno: 005 ISeqno: 004 Type: DTMF_B Subclass: 2 [Apr 28 18:43:42] VERBOSE[2806] logger.c:Timestamp: 14504ms SCall: 12052 DCall: 04642 [208.90.184.4:4569] [Apr 28 18:43:42] VERBOSE[2799] logger.c: Rx-Frame Retry[ No] -- OSeqno: 005 ISeqno: 004 Type: DTMF_B Subclass: 2 [Apr 28 18:43:42] VERBOSE[2799] logger.c:Timestamp: 14504ms SCall: 12052 DCall: 04642 [208.90.184.4:4569] -- [Apr 28 18:43:42] VERBOSE[2806] logger.c: Tx-Frame Retry[000] -- OSeqno: 008 ISeqno: 007 Type: DTMF_B Subclass: 2 [Apr 28 18:43:42] VERBOSE[2806] logger.c:Timestamp: 13363ms SCall: 09503 DCall: 09749 [208.90.184.3:4569] -- [Apr 28 18:43:42] VERBOSE[2806] logger.c: Tx-Frame Retry[000] -- OSeqno: 006 ISeqno: 004 Type: DTMF_E Subclass: 2 [Apr 28 18:43:42] VERBOSE[2806] logger.c:Timestamp: 14828ms SCall: 12052 DCall: 04642 [208.90.184.4:4569] [Apr 28 18:43:42] VERBOSE[2799] logger.c: Rx-Frame Retry[ No] -- OSeqno: 006 ISeqno: 004 Type: DTMF_E Subclass: 2 [Apr 28 18:43:42] VERBOSE[2799] logger.c:Timestamp: 14828ms SCall: 12052 DCall: 04642 [208.90.184.4:4569] -- [Apr 28 18:43:42] VERBOSE[2806] logger.c: Tx-Frame Retry[000] -- OSeqno: 009 ISeqno: 007 Type: DTMF_E Subclass: 2 [Apr 28 18:43:42] VERBOSE[2806] logger.c:Timestamp: 13700ms SCall: 09503 DCall: 09749 [208.90.184.3:4569] -- [Apr 28 18:43:43] VERBOSE[2806] logger.c: Tx-Frame Retry[000] -- OSeqno: 007 ISeqno: 004 Type: DTMF_B Subclass: 5 [Apr 28 18:43:43] VERBOSE[2806] logger.c:Timestamp: 15263ms SCall: 12052 DCall: 04642 [208.90.184.4:4569] [Apr 28 18:43:43] VERBOSE[2797] logger.c: Rx-Frame Retry[ No] -- OSeqno: 007 ISeqno: 004 Type: DTMF_B Subclass: 5 [Apr 28 18:43:43] VERBOSE[2797] logger.c:Timestamp: 15263ms SCall: 12052 DCall: 04642 [208.90.184.4:4569] -- [Apr 28 18:43:43] VERBOSE[2806] logger.c: Tx-Frame Retry[000] -- OSeqno: 010 ISeqno: 007 Type: DTMF_B Subclass: 5 [Apr 28 18:43:43] VERBOSE[2806] logger.c:Timestamp: 14103ms SCall: 09503 DCall: 09749 [208.90.184.3:4569] -- [Apr 28 18:43:43] VERBOSE[2806] logger.c: Tx-Frame Retry[000] -- OSeqno: 008 ISeqno: 004 Type: DTMF_E Subclass: 5 [Apr 28 18:43:43] VERBOSE[2806] logger.c:Timestamp: 15613ms SCall: 12052 DCall: 04642 [208.90.184.4:4569] [Apr 28 18:43:43] VERBOSE[2798] logger.c: Rx-Frame Retry[ No] -- OSeqno: 008 ISeqno: 004 Type: DTMF_E Subclass: 5 [Apr 28 18:43:43] VERBOSE[2798] logger.c:Timestamp: 15613ms SCall: 12052 DCall: 04642 [208.90.184.4:4569] -- [Apr 28 18:43:43] VERBOSE[2806] logger.c: Tx-Frame Retry[000] -- OSeqno: 011 ISeqno: 007 Type: DTMF_E Subclass: 5 [Apr 28 18:43:43] VERBOSE[2806] logger.c:Timestamp: 14460ms SCall: 09503 DCall: 09749 [208.90.184.3:4569] -- [Apr 28 18:43:43] VERBOSE[2806] logger.c: Tx-Frame Retry[000] -- OSeqno: 009 ISeqno: 004 Type: DTMF_B Subclass: 8 [Apr 28 18:43:43] VERBOSE[2806] logger.c:Timestamp: 15983ms SCall: 12052 DCall: 04642 [208.90.184.4:4569] [Apr 28 18:43:43] VERBOSE[2799] logger.c: Rx-Frame Retry[ No] -- OSeqno: 009 ISeqno: 004 Type: DTMF_B Subclass: 8 [Apr 28 18:43:43] VERBOSE[2799] logger.c:Timestamp: 15983ms SCall: 12052 DCall: 04642 [208.90.184.4:4569] -- [Apr 28 18:43:43] VERBOSE[2806] logger.c: Tx-Frame Retry[000] -- OSeqno: 012 ISeqno: 007 Type: DTMF_B Subclass: 8 [Apr 28 18:43:43] VERBOSE[2806] logger.c:Timestamp: 14823ms SCall: 09503 DCall: 09749 [208.90.184.3:4569] -- [Apr 28 18:43:44] VERBOSE[2806] logger.c: Tx-Frame Retry[000] -- OSeqno: 010 ISeqno: 004 Type: DTMF_E Subclass: 8 [Apr 28 18:43:44] VERBOSE[2806] logger.c:Timestamp: 16351ms SCall: 12052 DCall: 04642 [208.90.184.4:4569] [Apr 28 18:43:44] VERBOSE[2804] logger.c: Rx-Frame Retry[ No] -- OSeqno: 010 ISeqno: 004 Type: DTMF_E Subclass: 8 [Apr 28 18:43:44] VERBOSE[2804] logger.c:Timestamp: 16351ms SCall: 12052 DCall: 04642 [208.90.184.4:4569] -- [Apr 28 18:43:44] VERBOSE[2806] logger.c: Tx-Frame Retry[000] -- OSeqno: 013 ISeqno: 007 Type: DTMF_E Subclass: 8 [Apr 28 18:43:44] VERBOSE[2806] logger.c:Timestamp: 15200ms SCall: 09503 DCall: 09749 [208.90.184.3:4569] -- [Apr 28 18:43:44] VERBOSE[2806] logger.c: Tx-Frame Retry[000] -- OSeqno: 011 ISeqno: 004 Type: DTMF_B Subclass: 9 [Apr 28 18:43:44] VERBOSE[2806] logger.c:Timestamp: 16763ms SCall: 12052 DCall: 04642 [208.90.184.4:4569] [Apr 28 18:43:44] VERBOSE[2801] logger.c: Rx-Frame Retry[ No] -- OSeqno: 011 ISeqno: 004 Type: DTMF_B
[asterisk-users] [PATCH] Make Queue announcements more consistent (1.4.26.2)
Hi,
After playing around with queues a bunch on 1.4.26.2, I noticed a few things,
which the patch below addresses. It addresses:
- Callers in position 0 will hear periodic/position announcements at a
very different rate than all other callers.
-- Announcements while in position 0 could be delayed up to
timeout+retry seconds.
-- This patch reduces that possible delay to only timeout seconds
- The say_position and periodic_announcement times are in elapsed time
that _includes_ the
time of the announcement.
-- This patch changes those times to be the time _between_ playing
of those announcements
Thanks.
-- James
--- asterisk-1.4.26.2/apps/app_queue.c 2009-08-10 13:14:34.0 -0700
+++ asterisk-1.4.26.2.new/apps/app_queue.c 2010-04-25 22:25:08.0
-0700
@@ -345,6 +345,7 @@
time_t last_periodic_announce_time; /*! The last time we played a
periodic announcement */
int last_periodic_announce_sound; /*! The last periodic
announcement we made */
time_t last_pos;/*! Last time we told the user
their position */
+ time_t last_ring_time; /*! Last time we tried to ring
the agents */
int opos; /*! Where we started in the queue
*/
int handled;/*! Whether our call was handled */
int pending;/*! Non-zero if we are
attempting to call a member */
@@ -1653,6 +1654,7 @@
res = 0;
/* Set our last_pos indicators */
+ time(now);
qe-last_pos = now;
qe-last_pos_said = qe-pos;
@@ -2131,6 +2133,8 @@
if (!res)
ast_moh_start(qe-chan, qe-moh, NULL);
+ /* Refresh now so that frequency is time _between_ recordings */
+ time(now);
/* update last_periodic_announce_time */
qe-last_periodic_announce_time = now;
@@ -3292,7 +3296,8 @@
static int wait_a_bit(struct queue_ent *qe)
{
/* Don't need to hold the lock while we setup the outgoing calls */
- int retrywait = qe-parent-retry * 1000;
+ //int retrywait = qe-parent-retry * 1000;
+ int retrywait = RECHECK * 1000;
int res = ast_waitfordigit(qe-chan, retrywait);
if (res 0 !valid_exit(qe, res))
@@ -4003,6 +4008,7 @@
qe.max_penalty = max_penalty;
qe.last_pos_said = 0;
qe.last_pos = 0;
+ qe.last_ring_time = 0;
qe.last_periodic_announce_time = time(NULL);
qe.last_periodic_announce_sound = 0;
qe.valid_digits = 0;
@@ -4074,9 +4080,12 @@
break;
}
/* Try calling all queue members for 'timeout' seconds
*/
- res = try_calling(qe, args.options,
args.announceoverride,
args.url, tries, noption, args.agi);
- if (res)
- goto stop;
+ if ((time(NULL) - qe.last_ring_time)
qe.parent-retry) {
+ res = try_calling(qe, args.options,
args.announceoverride,
args.url, tries, noption, args.agi);
+ qe.last_ring_time = time(NULL);
+ if (res)
+ goto stop;
+ }
stat = get_member_status(qe.parent, qe.max_penalty);
@@ -4125,7 +4134,7 @@
/* If using dynamic realtime members, we should
regenerate the
member list for this queue */
update_realtime_members(qe.parent);
- /* OK, we didn't get anybody; wait for 'retry' seconds;
may get a
digit to exit with */
+ /* OK, we didn't get anybody; poll our retry */
res = wait_a_bit(qe);
if (res)
goto stop;
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Repeated: Got SIP response 489 Bad event back from
On Sat, Apr 10, 2010 at 6:35 AM, Adrian Marsh adrian.ma...@ubiquisys.com wrote: Hi All, I’ve two asterisk servers on the same LAN, both 1.4, and I keep getting “Got SIP response 489 Bad event back from 192.168.3.10” No idea whats causing it. The only references I can find mentions NATing issues, but these are on the same LAN so NAT shouldn’t be an issue. 3.10 does authenticate into the server logging the error. The error appears in the log every 1m20s (ish) Is 3.10 on a SIP trunk to the other asterisk box? Is qualify=yes on this SIP trunk? I think you'll find that if you run an ngrep/tcpdump on port 5060 on the box receiving the error it will send out an OPTIONS or NOTIFY (I can't remember which) and then you'll see the 489 Bad Event. Grab a trace of the SIP traffic and post it, its the only way to know for sure though. -- James Any ideas? Thanks, Adrian -- _ -- 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] Sending RTP media to a different server than SIP Signaling
On Sat, Apr 10, 2010 at 8:50 AM, Tarek Sawah tareksa...@hotmail.com wrote: Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 I don't believe this can be done in asterisk by itself, but you may be able to use the Linux conntrack stuff (http://netfilter.org/) to rewrite the SDP host information... However, if you want to dive into the world of OpenSIPS, I know you can do this with an OpenSIPS/MediaProxy setup between your asterisk box and your provider. -- James _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 -- _ -- 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] tones detection
Hi Jerry, On Thu, Apr 8, 2010 at 6:54 PM, Jerry Geis ge...@pagestation.com wrote: I am looking for something in asterisk that will let me record a wav file in asterisk (which I know how to do) then some other command (external or dialplan) that would read the wave file and tell me if a certain tone or frequency is present. Is this in asterisk already - any way to do it? Thanks You might want to look into the PipeWave tools: http://www.cardiff.ac.uk/psych/home2/CullingJ/pipewave.html The tools can generate a FFT (fast-fourier transform) of a wav file which converts the data into the frequency domain, which should allow you to tell if a certain frequency is present. -- James Jerry -- _ -- 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] Asterisk + DRBD Performance
Hi, Has anyone had any experience using DRBD to mirror an entire asterisk machine? If so, is there a performance issue at all when people are recording voicemails and the like? It seems like that could generate quite a bit of traffic. Also, do you bother to mirror the log files as well? Thanks. -- James -- _ -- 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] Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCodes
On Fri, Apr 9, 2010 at 5:17 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello my friends, I want to make fax work in the following scenario: My versions are: Asterisk 1.4.21.2 WANPIPE Release: 3.4.7 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P The E1 pri is connected to our Sangoma A102DE, we also have a SIP Mediant Audiocodes 1000 where we have some fax machines connected to fxs ports, what we need is to make fax machines through mediant send faxes to the pstn (through E1 PRI) and viceversa... What should we do to make this work properly? what parameters in zapata? mediant 1000? Thanks in advance for all your help! I've had fairly good success with faxing using Asterisk + Hylafax. I haven't tried any of the built-in Asterisk faxing programs yet because I designed this setup before the newest revisions, when Asterisk + built-in faxing was not working well. What I do is run Hylafax on the same machine as Asterisk, and then run IAXModem to do the communication between the 2. There's a lot of documentation online about how to set this up. -- James -- _ -- 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
[asterisk-users] Busy(20) returns non-zero and exits immediately on IAX channel
Hi, I'm running Asterisk 1.4.26.3 and I've noticed an interesting problem when trying to play a Busy tone over a IAX trunk from the PSTN. It seems as though Busy(20) returns non-zero immediately (it does not wait 20s), so the caller never hears the busy tone, but the call just appears to hang up. I don't believe this happens when trying to play a Busy on a SIP trunk. The busy part of the dialplan looks like this, exten = s-BUSY,1,Noop(Dial failed due to trunk reporting BUSY - giving up) exten = s-BUSY,n,Playtones(busy) exten = s-BUSY,n,Busy(20) The only way to remedy this is to put a Wait(20) between the Playtones() and Busy(). Any ideas on why this only fails on IAX and not SIP? Thank you. -- James -- _ -- 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] Problem with Sangoma A104 and euroisdn pri
On Thu, Apr 1, 2010 at 6:15 AM, Jaap Winius jwin...@umrk.to wrote: [snip] Besides the above error, I also noticed this: CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 The status needs to be Provisioned, Up, Active. Following Sangoma's instructions for debugging an Asterisk PRI span, I can confirm that there are only outgoing frames and that the D-channel messages in Asterisk are the same as what the Wanpipe drivers are seeing. So, assuming that my local telco (KPN Telecom) has activated the D-channel, what else could possibly be causing this problem? I would call KPN Telecom and ask them for help as well. They will have much more sophisticated tools for debugging PRIs and also will be able to check on their end if they see the D-Channel as up. -- James -- _ -- 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] Exceptionally long voice queue length errors...
Hi, I'm seeing a lot of Exceptionally long voice queue length errors in my logs, and then I seem to have a problem where Asterisk will drop the registration for a significant number of phones (they go UNREACHABLE), but then they come back approximately a minute later. Is this some sort of load problem? Or something else? Thank you. -- James -- _ -- 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] How are your PRI interrupts balanced? (+ Soft lockup BUG)
Hi, I'm trying to figure out the cause of a soft lockup I experienced: Mar 29 09:38:24 pstn1 kernel: BUG: soft lockup - CPU#0 stuck for 10s! [asterisk:32029] Mar 29 09:38:24 pstn1 kernel: Pid: 32029, comm: asterisk Mar 29 09:38:24 pstn1 kernel: EIP: 0060:[c046e7fe] CPU: 0 Mar 29 09:38:24 pstn1 kernel: EIP is at kfree+0x68/0x6c Mar 29 09:38:24 pstn1 kernel: EFLAGS: 0286Tainted: GF (2.6.18-128.1.10.el5 #1) Mar 29 09:38:24 pstn1 kernel: EAX: 0029 EBX: f7ff9380 ECX: f7fff880 EDX: c11ff9a0 Mar 29 09:38:24 pstn1 kernel: ESI: 0286 EDI: cffcda00 EBP: e5e10c80 DS: 007b ES: 007b Mar 29 09:38:24 pstn1 kernel: CR0: 80050033 CR2: b7ce39e0 CR3: 0f911000 CR4: 06d0 Mar 29 09:38:24 pstn1 kernel: [c05b067c] kfree_skbmem+0x8/0x61 Mar 29 09:38:24 pstn1 kernel: [c05e9aaf] __udp_queue_rcv_skb+0x4a/0x51 Mar 29 09:38:24 pstn1 kernel: [c05ad993] release_sock+0x44/0x91 Mar 29 09:38:24 pstn1 kernel: [c05ea939] udp_sendmsg+0x44e/0x514 Mar 29 09:38:24 pstn1 kernel: [c05efdec] inet_sendmsg+0x35/0x3f Mar 29 09:38:24 pstn1 kernel: [c05ab30c] sock_sendmsg+0xce/0xe8 Mar 29 09:38:24 pstn1 kernel: [c043464f] autoremove_wake_function+0x0/0x2d Mar 29 09:38:24 pstn1 kernel: [c04ea17b] copy_from_user+0x17/0x5d Mar 29 09:38:24 pstn1 kernel: [c04ea3a1] copy_to_user+0x31/0x48 Mar 29 09:38:24 pstn1 kernel: [f89ab141] zt_chan_read+0x1e0/0x20b [zaptel] Mar 29 09:38:24 pstn1 kernel: [c04ea195] copy_from_user+0x31/0x5d Mar 29 09:38:24 pstn1 kernel: [c05ac4c4] sys_sendto+0x116/0x140 Mar 29 09:38:24 pstn1 kernel: [c0415d4f] flush_tlb_page+0x74/0x77 Mar 29 09:38:24 pstn1 kernel: [c0461331] do_wp_page+0x3bf/0x40a Mar 29 09:38:24 pstn1 kernel: [c04284f1] current_fs_time+0x4a/0x55 Mar 29 09:38:24 pstn1 kernel: [c0488f9b] touch_atime+0x60/0x91 Mar 29 09:38:24 pstn1 kernel: [c047d9d0] pipe_readv+0x315/0x321 Mar 29 09:38:24 pstn1 kernel: [c05acde4] sys_socketcall+0x106/0x19e Mar 29 09:38:24 pstn1 kernel: [c0404f17] syscall_call+0x7/0xb Mar 29 09:38:24 pstn1 kernel: === This occurred during a high load period (52 calls across 3 PRI spans). A couple days ago I moved the interrupts for my PRI card to CPU0 from CPU3, because CPU3 was handling everything else: CPU0 CPU1 CPU2 CPU3 0:306 0 0 3684057379IO-APIC-edge timer 1: 0 0 0 13468IO-APIC-edge i8042 8: 0 0 0 3IO-APIC-edge rtc 9: 0 0 0 0 IO-APIC-level acpi 12: 0 0 0 4IO-APIC-edge i8042 169: 0 0 0 0 IO-APIC-level uhci_hcd:usb2 177: 0 0 0 18392593 IO-APIC-level ata_piix 185: 0 0 0 1 IO-APIC-level ehci_hcd:usb1 193: 0 0 0 0 IO-APIC-level uhci_hcd:usb3 201: 0 0 0 2090021759 IO-APIC-level eth0 209: 149621223 0 0 3534419461 IO-APIC-level wct4xxp (The CPU3 number for wct4xxp is not increasing any more). What is the interrupt distribution of other people's systems? Before I made this change I was having a problem with D-channels dropping occasionally, so I thought it might be an interrupt/load issue. Thank you. -- James -- _ -- 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] How are your PRI interrupts balanced? (+ Soft lockup BUG)
On Mon, Mar 29, 2010 at 8:38 PM, Matt Watson m...@mattgwatson.ca wrote: Dell server by any chance? I have a similar problem with a TE220B in a Dell 1950 III server - i've seen several other people having issues with digium cards in dell servers as well. I've actually done something similar to what you have done - isolated the TE220B onto its own IRQ and set processor affinity for all the IRQs to particular cores... so far I haven't had kernel pancs since doing this, but its still a little too early to say if it has fixed the issue 100% or not. Interesting. It is actually a Dell SC1425 - Dual, dual-core Xeon Processors. I'm hopefully going to be able to stress test this machine to see if I can make it panic again with the PRI card IRQ isolated to CPU0. If so, I'll see if it does the same thing on the other cores... -- James -- Matt On Mon, Mar 29, 2010 at 8:30 PM, James Lamanna jlama...@gmail.com wrote: Hi, I'm trying to figure out the cause of a soft lockup I experienced: Mar 29 09:38:24 pstn1 kernel: BUG: soft lockup - CPU#0 stuck for 10s! [asterisk:32029] Mar 29 09:38:24 pstn1 kernel: Pid: 32029, comm: asterisk Mar 29 09:38:24 pstn1 kernel: EIP: 0060:[c046e7fe] CPU: 0 Mar 29 09:38:24 pstn1 kernel: EIP is at kfree+0x68/0x6c Mar 29 09:38:24 pstn1 kernel: EFLAGS: 0286 Tainted: GF (2.6.18-128.1.10.el5 #1) Mar 29 09:38:24 pstn1 kernel: EAX: 0029 EBX: f7ff9380 ECX: f7fff880 EDX: c11ff9a0 Mar 29 09:38:24 pstn1 kernel: ESI: 0286 EDI: cffcda00 EBP: e5e10c80 DS: 007b ES: 007b Mar 29 09:38:24 pstn1 kernel: CR0: 80050033 CR2: b7ce39e0 CR3: 0f911000 CR4: 06d0 Mar 29 09:38:24 pstn1 kernel: [c05b067c] kfree_skbmem+0x8/0x61 Mar 29 09:38:24 pstn1 kernel: [c05e9aaf] __udp_queue_rcv_skb+0x4a/0x51 Mar 29 09:38:24 pstn1 kernel: [c05ad993] release_sock+0x44/0x91 Mar 29 09:38:24 pstn1 kernel: [c05ea939] udp_sendmsg+0x44e/0x514 Mar 29 09:38:24 pstn1 kernel: [c05efdec] inet_sendmsg+0x35/0x3f Mar 29 09:38:24 pstn1 kernel: [c05ab30c] sock_sendmsg+0xce/0xe8 Mar 29 09:38:24 pstn1 kernel: [c043464f] autoremove_wake_function+0x0/0x2d Mar 29 09:38:24 pstn1 kernel: [c04ea17b] copy_from_user+0x17/0x5d Mar 29 09:38:24 pstn1 kernel: [c04ea3a1] copy_to_user+0x31/0x48 Mar 29 09:38:24 pstn1 kernel: [f89ab141] zt_chan_read+0x1e0/0x20b [zaptel] Mar 29 09:38:24 pstn1 kernel: [c04ea195] copy_from_user+0x31/0x5d Mar 29 09:38:24 pstn1 kernel: [c05ac4c4] sys_sendto+0x116/0x140 Mar 29 09:38:24 pstn1 kernel: [c0415d4f] flush_tlb_page+0x74/0x77 Mar 29 09:38:24 pstn1 kernel: [c0461331] do_wp_page+0x3bf/0x40a Mar 29 09:38:24 pstn1 kernel: [c04284f1] current_fs_time+0x4a/0x55 Mar 29 09:38:24 pstn1 kernel: [c0488f9b] touch_atime+0x60/0x91 Mar 29 09:38:24 pstn1 kernel: [c047d9d0] pipe_readv+0x315/0x321 Mar 29 09:38:24 pstn1 kernel: [c05acde4] sys_socketcall+0x106/0x19e Mar 29 09:38:24 pstn1 kernel: [c0404f17] syscall_call+0x7/0xb Mar 29 09:38:24 pstn1 kernel: === This occurred during a high load period (52 calls across 3 PRI spans). A couple days ago I moved the interrupts for my PRI card to CPU0 from CPU3, because CPU3 was handling everything else: CPU0 CPU1 CPU2 CPU3 0: 306 0 0 3684057379 IO-APIC-edge timer 1: 0 0 0 13468 IO-APIC-edge i8042 8: 0 0 0 3 IO-APIC-edge rtc 9: 0 0 0 0 IO-APIC-level acpi 12: 0 0 0 4 IO-APIC-edge i8042 169: 0 0 0 0 IO-APIC-level uhci_hcd:usb2 177: 0 0 0 18392593 IO-APIC-level ata_piix 185: 0 0 0 1 IO-APIC-level ehci_hcd:usb1 193: 0 0 0 0 IO-APIC-level uhci_hcd:usb3 201: 0 0 0 2090021759 IO-APIC-level eth0 209: 149621223 0 0 3534419461 IO-APIC-level wct4xxp (The CPU3 number for wct4xxp is not increasing any more). What is the interrupt distribution of other people's systems? Before I made this change I was having a problem with D-channels dropping occasionally, so I thought it might be an interrupt/load issue. Thank you. -- James -- _ -- 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
Re: [asterisk-users] How are your PRI interrupts balanced? (+ Soft lockup BUG)
On Mon, Mar 29, 2010 at 9:23 PM, James Lamanna jlama...@gmail.com wrote: On Mon, Mar 29, 2010 at 8:38 PM, Matt Watson m...@mattgwatson.ca wrote: Dell server by any chance? I have a similar problem with a TE220B in a Dell 1950 III server - i've seen several other people having issues with digium cards in dell servers as well. I've actually done something similar to what you have done - isolated the TE220B onto its own IRQ and set processor affinity for all the IRQs to particular cores... so far I haven't had kernel pancs since doing this, but its still a little too early to say if it has fixed the issue 100% or not. Interesting. It is actually a Dell SC1425 - Dual, dual-core Xeon Processors. I'm hopefully going to be able to stress test this machine to see if I can make it panic again with the PRI card IRQ isolated to CPU0. If so, I'll see if it does the same thing on the other cores... As a data point, I tried stress testing this box this evening. Moving the interrupt to each core, the results did not change. The test was as follows: Originate() a call that goes out to the PSTN and comes back in. Both sides used Milliwatt() to make sure audio flowed both ways. I generated 30 calls this way (to use 60 PRI channels), however, I was never able to simultaneously keep 60 channels alive. During the test, there would always be a D-Channel down/up, which would drop all calls on that PRI span. I do not know if this is a Zaptel issue (1.4.12), PRI card issue (TE401P first-gen), or something more subtle... Any help would be appricated! Thanks. -- James -- James -- Matt On Mon, Mar 29, 2010 at 8:30 PM, James Lamanna jlama...@gmail.com wrote: Hi, I'm trying to figure out the cause of a soft lockup I experienced: Mar 29 09:38:24 pstn1 kernel: BUG: soft lockup - CPU#0 stuck for 10s! [asterisk:32029] Mar 29 09:38:24 pstn1 kernel: Pid: 32029, comm: asterisk Mar 29 09:38:24 pstn1 kernel: EIP: 0060:[c046e7fe] CPU: 0 Mar 29 09:38:24 pstn1 kernel: EIP is at kfree+0x68/0x6c Mar 29 09:38:24 pstn1 kernel: EFLAGS: 0286 Tainted: GF (2.6.18-128.1.10.el5 #1) Mar 29 09:38:24 pstn1 kernel: EAX: 0029 EBX: f7ff9380 ECX: f7fff880 EDX: c11ff9a0 Mar 29 09:38:24 pstn1 kernel: ESI: 0286 EDI: cffcda00 EBP: e5e10c80 DS: 007b ES: 007b Mar 29 09:38:24 pstn1 kernel: CR0: 80050033 CR2: b7ce39e0 CR3: 0f911000 CR4: 06d0 Mar 29 09:38:24 pstn1 kernel: [c05b067c] kfree_skbmem+0x8/0x61 Mar 29 09:38:24 pstn1 kernel: [c05e9aaf] __udp_queue_rcv_skb+0x4a/0x51 Mar 29 09:38:24 pstn1 kernel: [c05ad993] release_sock+0x44/0x91 Mar 29 09:38:24 pstn1 kernel: [c05ea939] udp_sendmsg+0x44e/0x514 Mar 29 09:38:24 pstn1 kernel: [c05efdec] inet_sendmsg+0x35/0x3f Mar 29 09:38:24 pstn1 kernel: [c05ab30c] sock_sendmsg+0xce/0xe8 Mar 29 09:38:24 pstn1 kernel: [c043464f] autoremove_wake_function+0x0/0x2d Mar 29 09:38:24 pstn1 kernel: [c04ea17b] copy_from_user+0x17/0x5d Mar 29 09:38:24 pstn1 kernel: [c04ea3a1] copy_to_user+0x31/0x48 Mar 29 09:38:24 pstn1 kernel: [f89ab141] zt_chan_read+0x1e0/0x20b [zaptel] Mar 29 09:38:24 pstn1 kernel: [c04ea195] copy_from_user+0x31/0x5d Mar 29 09:38:24 pstn1 kernel: [c05ac4c4] sys_sendto+0x116/0x140 Mar 29 09:38:24 pstn1 kernel: [c0415d4f] flush_tlb_page+0x74/0x77 Mar 29 09:38:24 pstn1 kernel: [c0461331] do_wp_page+0x3bf/0x40a Mar 29 09:38:24 pstn1 kernel: [c04284f1] current_fs_time+0x4a/0x55 Mar 29 09:38:24 pstn1 kernel: [c0488f9b] touch_atime+0x60/0x91 Mar 29 09:38:24 pstn1 kernel: [c047d9d0] pipe_readv+0x315/0x321 Mar 29 09:38:24 pstn1 kernel: [c05acde4] sys_socketcall+0x106/0x19e Mar 29 09:38:24 pstn1 kernel: [c0404f17] syscall_call+0x7/0xb Mar 29 09:38:24 pstn1 kernel: === This occurred during a high load period (52 calls across 3 PRI spans). A couple days ago I moved the interrupts for my PRI card to CPU0 from CPU3, because CPU3 was handling everything else: CPU0 CPU1 CPU2 CPU3 0: 306 0 0 3684057379 IO-APIC-edge timer 1: 0 0 0 13468 IO-APIC-edge i8042 8: 0 0 0 3 IO-APIC-edge rtc 9: 0 0 0 0 IO-APIC-level acpi 12: 0 0 0 4 IO-APIC-edge i8042 169: 0 0 0 0 IO-APIC-level uhci_hcd:usb2 177: 0 0 0 18392593 IO-APIC-level ata_piix 185: 0 0 0 1 IO-APIC-level ehci_hcd:usb1 193: 0 0 0 0 IO-APIC-level uhci_hcd:usb3 201: 0 0 0 2090021759 IO-APIC-level eth0 209: 149621223 0 0 3534419461 IO-APIC-level wct4xxp (The CPU3 number for wct4xxp is not increasing any more). What is the interrupt distribution of other people's systems? Before I made this change I was having a problem with D
Re: [asterisk-users] Cisco 7960 become UNREACHABLE behind pix firewall
Alyed wrote: From: http://www.voip-info.org/wiki/view/Asterisk+sip+qualify If you turn on *qualify* in the configuration of a SIP device in sip.confhttp://www.voip-info.org/wiki/view/Asterisk+config+sip.conf, asterisk will send a SIP OPTIONShttp://www.voip-info.org/wiki/view/SIP+method+optionscommand regularly to check that the device is still online. If the device does not answer within the configured (or default) period (in ms) Asterisk considers the device off-line for future calls. This status can be checked by the SIPPEER functionhttp://www.voip-info.org/wiki/view/Asterisk+func+sippeer, and inversely this function will only provide status information for peers which have *qualify=yes*. My guess is that your Nat/firewall is closing the connection after some time the phone is idle, so this way Asterisk will make sure to always have communication going trhough that connection so your NAT/firewall won't just close it. Sorry, should have mentioned that all these phones have qualify=yes and nat=yes in sip.conf. Thanks. -- James On Sat, Mar 27, 2010 at 8:17 AM, James Lamanna jlama...@gmail.com wrote: Hi, I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall. After some period of time, asterisk says that some of them are unreachable, and the phones lose their registration. The only way to make the phones recover is to clear the NAT translation tables for the phones on the PIX (clear xlate...) Does anyone know how to fix this? As you can imagine, it is quite annoying. And it does not happen to all the phones either. sip fixup is enabled on the PIX phone config parts: nat_enable : 1 nat_received_processing : 0 nat_address: [public ip of PIX] -- _ -- 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] 24 FXS Port Voip Gateway and Asterisk
On Sun, Mar 28, 2010 at 8:28 PM, Steve Edwards asterisk@sedwards.com wrote: On Sun, 28 Mar 2010, Joseph Begumisa wrote: Can anyone recommend a 24 fxs port voip gateway that has worked well with asterisk? I have a couple of analog handsets that I want to hookup to my asterisk server? Any tested and tried product recommendations are welcome. Thanks. Adtran channel banks are a great trailing edge technology. You can get them off Ebay for pennies on the original dollar and they are built like a tank. (voip gateway is not very specific. If you meant SIP or IAX, you might want to specify which.) I've actually had decent success with the GXW-4024 (FXS - SIP) from Grandstream which is probably one of the cheapest 24 FXS port boxes you'll find out there. -- James -- _ -- 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] Cisco 7960 become UNREACHABLE behind pix firewall
Hi, I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall. After some period of time, asterisk says that some of them are unreachable, and the phones lose their registration. The only way to make the phones recover is to clear the NAT translation tables for the phones on the PIX (clear xlate...) Does anyone know how to fix this? As you can imagine, it is quite annoying. And it does not happen to all the phones either. sip fixup is enabled on the PIX phone config parts: nat_enable : 1 nat_received_processing : 0 nat_address: [public ip of PIX] Thank you. -- James (Please CC me on all replies) -- _ -- 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] Maximum number of PRI calls on 1 asterisk box (no HW echo)
Zeeshan A Zakaria wrote: On Wed, Mar 24, 2010 at 5:42 PM, James Lamanna jlama...@gmail.com wrote: [snip] The specs of the machine are, Dual Xeon 2.80Ghz (both single core but w/HT) 4GB memory. Running asterisk 1.4.26.3 (32-bit) with libpri-1.4.7 and zaptel-1.4.12.9 So I think it is not your T1 card but some software/driver issue. Did you upgrade anything recently on this server? I actually downgraded this server from 1.6.0.12 to the above configuration. I was having issues with spans locking all their channels with 1.6.0.12. -- James ** Please CC me directly on all responses - I am subscribed in digest mode ** -- _ -- 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] Maximum number of PRI calls on 1 asterisk box (no HW echo)
Hi, Does anyone have any good empirical data suggesting what the maximum number of PRI calls (incoming and outgoing) without hardware echo cancellation can be handled on a single box is? I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of D-Channels going down and then coming back up (See below). I've looked at the number of simultaneous calls at each of these points, and each time the span seems to have around 21-23 calls, and the total number of calls ranges between 47 and 53. I'm trying to figure out if this is a load issue or an issue on the provider side, though my provider says they do not see any errors on any of the T1s. Could this be some sort of hardware interrupt problem? If so, how can I check? The specs of the machine are, Dual Xeon 2.80Ghz (both single core but w/HT) 4GB memory. Running asterisk 1.4.26.3 (32-bit) with libpri-1.4.7 and zaptel-1.4.12.9 Thanks. -- James Please CC me on responses. [Mar 22 09:45:00] VERBOSE[8887] logger.c: == Primary D-Channel on span 2 down [Mar 22 09:45:00] WARNING[8887] chan_dahdi.c: No D-channels available! Using Primary channel 48 as D-channel anyway! [Mar 22 09:45:00] VERBOSE[8887] logger.c: == Primary D-Channel on span 2 up [Mar 22 09:59:23] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 09:59:23] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 09:59:23] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 09:59:23] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 09:59:23] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 09:59:23] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 10:36:11] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 10:36:11] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 10:36:11] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 10:36:11] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 10:36:11] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 10:36:11] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 10:44:36] NOTICE[] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 3 [Mar 22 10:45:44] NOTICE[8886] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 [Mar 22 10:59:33] NOTICE[8887] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 [Mar 22 11:30:53] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 11:30:53] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 11:30:53] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 15:34:28] VERBOSE[8887] logger.c: == Primary D-Channel on span 2 down [Mar 22 15:34:28] WARNING[8887] chan_dahdi.c: No D-channels available! Using Primary channel 48 as D-channel anyway! [Mar 22 15:34:28] VERBOSE[8887] logger.c: == Primary D-Channel on span 2 up -- _ -- 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] Hung channel problem with 1.4.26.2
Hi, I have a case where SIP channels will not be destroyed, resulting in further calls to ChanIsAvail() to fail. The process (I believe) to replicate this is the following: - Make a call to another SIP phone that is an intercom call (Auto-Answer) - For whatever reason, the phone happens to go UNREACHABLE during this call - Phone comes back REACHABLE, but channel still exists in core show channels As an example, here's 3 stuck calls from today: r...@hades:~# asterisk -rx core show channels Channel Location State Application(Data) SIP/6296-a2298 (None) Up AppDial((Outgoing Line)) SIP/6315-a0906 *806...@ext-in Up Dial(SIP/6296|5|A(beep)) SIP/6333-a131e (None) Up AppDial((Outgoing Line)) SIP/6294-a24fc *806...@ext-in Up Dial(SIP/6333|5|A(beep)) SIP/6297-a1cb7 (None) Up AppDial((Outgoing Line)) SIP/6315-adc5d *806...@ext-in Up Dial(SIP/6297|5|A(beep)) I don't know if this has been fixed in a later 1.4.x version, though after reading some of the DTMF relaying problems with 1.4.27 and beyond, I don't think I would want to upgrade... Thanks. -- James -- _ -- 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
Re: [asterisk-users] PRI Problems with 1.6.0.10
On Fri, Feb 12, 2010 at 12:54 PM, James Lamanna jlama...@gmail.com wrote: Hi, I have a PRI problem where it appears that my system is not responding to SETUP messages on a channel. It seems to be retransmitting a significant number of RELEASE messages to clear a call that is most likely to be long gone. This causes a huge issue because I get a bunch of hangup cause 102s (timeout). I'm using a TE410P (1st Gen) as my PRI card. Has anyone seen this at all? So I'm guessing no one has seen this... I am going to try and go back to 1.4.26.3 built against Zaptel, not Dahdi, to see if that works better. Unfortunately I don't have ISDN traces from both sides because getting the actual traces from my provider has proven difficult. But there were definitely traces of my asterisk box sending a SETUP message, though my provider never received it. -- James -- _ -- 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
[asterisk-users] PRI Problems with 1.6.0.10
Hi, I have a PRI problem where it appears that my system is not responding to SETUP messages on a channel. It seems to be retransmitting a significant number of RELEASE messages to clear a call that is most likely to be long gone. This causes a huge issue because I get a bunch of hangup cause 102s (timeout). I'm using a TE410P (1st Gen) as my PRI card. Has anyone seen this at all? Thanks. --James -- _ -- 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
[asterisk-users] Terminate T.38 to PSTN
Hi, Has terminating T.38 to PSTN found its way into the asterisk 1.6 mainline yet? I remember seeing an app_gateway floating around at some point a while ago, but I never had any luck with it. Thanks. -- James ___ -- 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
[asterisk-users] MeetMe thinks DAHDI is missing 1.6.0.10
Hi, I've noticed that my MeetMe install seems to think chan_dahdi is missing: app_meetme.c: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) However, it definitely is since I have 3 PRIs functioning normally :) Is there anything I should check before I restart asterisk this evening to see if that fixes it? Thanks. -- James ** Please CC me on all responses. Thanks! ___ -- 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
[asterisk-users] dahdi dies with No more room in scheduler
Hi, I noticed that Dahdi starting producing these error messages: ERROR[29250] chan_dahdi.c: No more room in scheduler ERROR[29250] chan_dahdi.c: Asked to delete sched id -1??? during which time I could not send any calls or receive calls on at least one of my Dahdi spans. The only way to clear the problem seemed to be to restart Asterisk. It appears to start after the following message ERROR[29250] chan_dahdi.c: T200 counter expired, nothing to send... This is with dahdi 2.2.0 and asterisk 1.6.0.10. Any ideas on this issue? Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Busy() returns immediately on IAX trunk
Hi, I have two asterisk boxes AB connected together via IAX. Phones register to Asterisk box A, and Asterisk box B is the PSTN connection. When dialing a number from a phone registered to A that DAHDI returns as BUSY, the Busy(20) application returns immediately instead of playing the busy tone. The user, instead of hearing the busy tone, gets an immediate hangup. Why is this? Busy(20) seems to work fine if there is a SIP trunk between the boxes, but for other reasons I cannot use a SIP trunk. Thank you. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configuring Parallel SIP Trunks
Hi, I'm trying to configure 2 parallel sip trunks between 2 boxes. However I seem to have the problem that when making a call from Box 2 to Box 1, it sometimes says authentication failed because it is using the username of the other trunk. Here's my configuration: Box 1: [dp-dp2] type=peer username=dp-dp2 secret=mysecret qualify=yes host=box.2.ip.address context=from-internal [e911-dp2] context=from-pstn host=box.2.ip.address qualify=yes secret=mysecret2 type=peer username=e911-dp2 Box 2: [dp-dp2] host=box.1.ip.address qualify=yes type=peer username=dp-dp2 secret=mysecret context=from-pstn [e911-dp2] context=from-internal host=box.1.ip.address qualify=yes secret=mysecret2 type=peer username=e911-dp2 If I have both trunks up, I'll see in the log on Box 1, when calling from Box 2 - Box 1: username mismatch, have e911-dp2, digest has dp-dp2 How can I get both to co-exist? Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Company in Los Angeles looking for Asterisk Network Administration/Maintenance Engineer
Hi, I work for a small VoIP/Internet service provider here in Southern California and we are currently looking for a Network Administrator who also knows Asterisk to add to our support staff. Some of your duties would include, - Maintenance of current Linux servers - Maintenance of current Asterisk servers - Troubleshooting of Asterisk/server related issues - Call routing provisioning We are a very small company, so you will be given a significant amount of latitude when it comes to administration tasks, and also be given the opportunity to grow with the company. Being that we are very small company, you should be a self-starter and also be able to contribute to enhancing the network as well, through implementation of more proactive monitoring techniques and whatever else you may see fit. Ideally this candidate will have had at least 5 years experience of Linux server administration along with at least 1 year of Asterisk administration and troubleshooting. Cisco router/switch configuration/administration knowledge is a HUGE plus. We are located in Pasadena, California, so ideally you would reside in the Southern California area, but telecommuting options can be discussed. Please email me directly for more details or any questions about the position. Thanks. James Lamanna Warp2Biz, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk in VMWare, how does it perform and what is the limit?
Hi, I'm coming up with ideas about building a cluster of asterisk servers, and am exploring the virtualization option. I'm curious to know some real-world data about how many extensions a VMWare install on good hardware could support. I've seen stories about how the hypervisor timeslicing can wreak havoc on call quality at some point. Is this really the case? If so, what's a feasible extension limit? 20? 50? 100? Any information would be great. Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in VMWare, how does it perform and what is the limit?
On Fri, Aug 7, 2009 at 11:47 AM, James Lamannajlama...@gmail.com wrote: Hi, I'm coming up with ideas about building a cluster of asterisk servers, and am exploring the virtualization option. I'm curious to know some real-world data about how many extensions a VMWare install on good hardware could support. I've seen stories about how the hypervisor timeslicing can wreak havoc on call quality at some point. Is this really the case? If so, what's a feasible extension limit? 20? 50? 100? Any information would be great. So VMWare messes around with clock timing. This is a Bad Thing if you're trying to do things that rely on faithful timing, such as audio mixing for a MeetMe conference room. If you're only doing very simple things like playing messages or ordinary bridged two-way phone calls it probably wouldn't be as bad. If call quality matters, at all, I wouldn't go that route. If managing a real server with asterisk is too hard for your data center, may I humbly suggest an asterisk appliance? Managing a server isn't the problem, I'm just looking to explore all solutions. If the call quality issues are that bad on vmware, then it is a non-starter in my book, especially trying to support the number of extensions I have now (I have 500 at the moment). Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 and RFC4235
Does Asterisk 1.6 fully support RFC4235? Or is it the same implementation as 1.4? Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Weird audio problem with remote IVRs + DMTF
Hi, Some users have been reporting a peculiar problem. The are having an issue when they dial out to some multi-level IVRs where you make 2 or 3 touchtone choices and then are connected to a live operator. When the live operator connects the operator cannot hear them or sometimes it results in dead air. With the one-way audio issue, is it possible that something has locked the channel into some mode where all audio being sent is muted? (As a result of DTMF?) I'm really perplexed by this one. Thanks. -- James ___ -- 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
Re: [asterisk-users] Phones dropping registration, but asterisk thinks phones are still registered
On Thu, Jun 4, 2009 at 11:08 AM, James Lamannajlama...@gmail.com wrote: Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is to completely restart Asterisk. Has anyone experienced this? This is a serious problem. I've poured over the logs while and after this happens and there is nothing in the logs that would suggest there is a problem. This is a production server, so I can't just upgrade Asterisk to the latest 1.4 version. I know people have suggested upgrading the server, but I'm not in a position to do that right now. However, I believe there is a symptom. When I do a sip show peer on an affected phone, the expire time is NEGATIVE. I think this might be contributing to the problem, and why Asterisk thinks the phone is still registered. Thanks. -- James ___ -- 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
Re: [asterisk-users] Phones dropping registration, but asterisk thinks phones are still registered
Why can't you just do a daily/weekly cron to restart when convenient in off/slow hours for local time. Is your business constantly on-line 24/7? I have tried that. Unfortunately restart when convenient doesn't always seem to actually restart asterisk, presumably because there are stuck calls or something. Very annoying as well. -- James On Thu, Jun 4, 2009 at 11:08 AM, James Lamannajlama...@gmail.com wrote: Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is to completely restart Asterisk. Has anyone experienced this? This is a serious problem. I've poured over the logs while and after this happens and there is nothing in the logs that would suggest there is a problem. This is a production server, so I can't just upgrade Asterisk to the latest 1.4 version. I know people have suggested upgrading the server, but I'm not in a position to do that right now. However, I believe there is a symptom. When I do a sip show peer on an affected phone, the expire time is NEGATIVE. I think this might be contributing to the problem, and why Asterisk thinks the phone is still registered. ___ -- 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
[asterisk-users] T38 Fax Gateway for Asterisk 1.6
Hi, I remember seeing a T38 Gateway application for Asterisk 1.6 floating around, but I can't seem to find it again. Does anyone have any pointers to it? I really want to be able to send an incoming T38 connection directly to the PSTN. Thanks. -- James ___ -- 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
Re: [asterisk-users] T38 Fax Gateway for Asterisk 1.6
On Fri, Jun 26, 2009 at 11:10 AM, James Lamannajlama...@gmail.com wrote: Hi, I remember seeing a T38 Gateway application for Asterisk 1.6 floating around, but I can't seem to find it again. Does anyone have any pointers to it? I really want to be able to send an incoming T38 connection directly to the PSTN. I've seen a number of similar requests. Can somebody explain the use-case to me? The use case is that a customer has a fax machine attached to an ATA. The ATA sends T38 to Asterisk over SIP, then I need to forward that out the PSTN. -- James ___ -- 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
Re: [asterisk-users] T38 Fax Gateway for Asterisk 1.6
On Fri, Jun 26, 2009 at 1:31 PM, James Lamannajlama...@gmail.com wrote: The use case is that a customer has a fax machine attached to an ATA. The ATA sends T38 to Asterisk over SIP, then I need to forward that out the PSTN. Got it. I'm saying why not skip the ATA and asterisk, and plug the fax into the PSTN? When you run a hosted service where a customer is connected to your remote datacenter over dedicated voice/data T1s, they are a little far from the PSTN connection :) -- James ___ -- 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
Re: [asterisk-users] T38 Fax Gateway for Asterisk 1.6
On Fri, Jun 26, 2009 at 11:10 AM, James Lamannajlama...@gmail.com wrote: Hi, I remember seeing a T38 Gateway application for Asterisk 1.6 floating around, but I can't seem to find it again. Does anyone have any pointers to it? I really want to be able to send an incoming T38 connection directly to the PSTN. There's always store and forward, also known as email, but with far less resolution. ReceiveFax(), set the file to a directory that you scan for outgoing fax, then SendFax(). That would require putting a * box at the customer's premises, which we do not do (see hosted model from earlier). I'm trying to avoid sending T.30 at all, even over dedicated links. -- James ___ -- 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
Re: [asterisk-users] Phones dropping registration, but asterisk thinks phones are still registered
Oliver wrote: How many phones are concerned ? The box currently has about 380 active phone registrations. Thanks. Please CC me directly as well because I'm on digest mode. -- James ___ -- 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
[asterisk-users] Phones dropping registration, but asterisk thinks phones are still registered
Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is to completely restart Asterisk. Has anyone experienced this? This is a serious problem. I've poured over the logs while and after this happens and there is nothing in the logs that would suggest there is a problem. This is a production server, so I can't just upgrade Asterisk to the latest 1.4 version. Thanks. -- James ___ -- 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
[asterisk-users] No response to our critical packet problem
Hi, I have a strange problem. At a site where there are 20+ phones, there is one phone that cannot make outbound (to PSTN) calls. Each call is dropped after 20s with no response to our critical packet. Calls to voicemail and internal extensions work fine. I understand that everything points to a NAT problem, but I don't understand how it could be because: 1) It does not affect calls to internal office extensions (which still go through asterisk) OR voicemail 2) The other 20+ phones in the same office on the same network have 0 problems. Here's a SIP trace of the problem. yyy.yyy.yyy.yyy is the outside NAT IP xxx.xxx.xxx.xxx is the IP of my PBX dd is the dialed phone number sss is the source phone number The peculiar thing is that asterisk sends an OK in response to an INVITE, then the phone sends back an ACK, which asterisk seems to ignore because it retransmits the OK message again Then eventually the phone gives up and sends a BYE message. -- James --- SIP read from yyy.yyy.yyy.yyy:24050 --- INVITE sip:ddd...@xxx.xxx.xxx.xxx SIP/2.0^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-6e730c81^M From: sss-sss- ;tag=bdfe4214c494d109o0^M To: ^M Call-ID: c4560330-de7ca...@10.1.24.145^m CSeq: 101 INVITE^M Max-Forwards: 70^M Contact: sss-sss- ^M Expires: 240^M User-Agent: Linksys/SPA942-6.1.3(a)^M Content-Length: 395^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: replaces^M Content-Type: application/sdp^M ^M v=0^M o=- 6363534 6363534 IN IP4 10.1.24.145^M s=-^M c=IN IP4 10.1.24.145^M t=0 0^M m=audio 16458 RTP/AVP 0 2 4 8 18 96 97 98 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:2 G726-32/8000^M a=rtpmap:4 G723/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:18 G729a/8000^M a=rtpmap:96 G726-40/8000^M a=rtpmap:97 G726-24/8000^M a=rtpmap:98 G726-16/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-15^M a=ptime:20^M a=sendrecv^M - --- Reliably Transmitting (NAT) to yyy.yyy.yyy.yyy:24050 --- SIP/2.0 407 Proxy Authentication Required^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-6e730c81;received=yyy.yyy.yyy.yyy^M From: sss-sss- ;tag=bdfe4214c494d109o0^M To: ;tag=as70a8455c^M Call-ID: c4560330-de7ca...@10.1.24.145^m CSeq: 101 INVITE^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Supported: replaces^M Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=6d2db4b7^M Content-Length: 0^M --- SIP read from yyy.yyy.yyy.yyy:24050 --- ACK sip:ddd...@xxx.xxx.xxx.xxx SIP/2.0^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-6e730c81^M From: sss-sss- ;tag=bdfe4214c494d109o0^M To: ;tag=as70a8455c^M Call-ID: c4560330-de7ca...@10.1.24.145^m CSeq: 101 ACK^M Max-Forwards: 70^M Contact: sss-sss- ^M User-Agent: Linksys/SPA942-6.1.3(a)^M Content-Length: 0^G ^M --- SIP read from yyy.yyy.yyy.yyy:24050 --- INVITE sip:ddd...@xxx.xxx.xxx.xxx SIP/2.0^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-3d87585d^M From: sss-sss- ;tag=bdfe4214c494d109o0^M To: ^M Call-ID: c4560330-de7ca...@10.1.24.145^m CSeq: 102 INVITE^M Max-Forwards: 70^M Proxy-Authorization: Digest username=ss,realm=asterisk,nonce=6d2db4b7,uri=sip:ddd...@xxx.xxx.xxx.xxx,algorithm=MD5,response= Contact: sss-sss- ^M Expires: 240^M User-Agent: Linksys/SPA942-6.1.3(a)^M Content-Length: 395^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: replaces^M Content-Type: application/sdp^M ^M v=0^M o=- 6363534 6363534 IN IP4 10.1.24.145^M s=-^M c=IN IP4 10.1.24.145^M t=0 0^M m=audio 16458 RTP/AVP 0 2 4 8 18 96 97 98 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:2 G726-32/8000^M a=rtpmap:4 G723/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:18 G729a/8000^M a=rtpmap:96 G726-40/8000^M a=rtpmap:97 G726-24/8000^M a=rtpmap:98 G726-16/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-15^M a=ptime:20^M a=sendrecv^M - --- Transmitting (NAT) to yyy.yyy.yyy.yyy:24050 --- SIP/2.0 100 Trying^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-3d87585d;received=yyy.yyy.yyy.yyy^M From: sss-sss- ;tag=bdfe4214c494d109o0^M To: ^M Call-ID: c4560330-de7ca...@10.1.24.145^m CSeq: 102 INVITE^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Supported: replaces^M Contact: ^M Content-Length: 0^M ^M --- Transmitting (NAT) to yyy.yyy.yyy.yyy:24050 --- SIP/2.0 183 Session Progress^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-3d87585d;received=yyy.yyy.yyy.yyy^M From: sss-sss- ;tag=bdfe4214c494d109o0^M To: ;tag=as30846812^M Call-ID: c4560330-de7ca...@10.1.24.145^m CSeq: 102 INVITE^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Supported: replaces^M Contact: ^M Content-Type: application/sdp^M Content-Length: 264^M ^M v=0^M o=root 32147 32147 IN IP4 xxx.xxx.xxx.xxx^M s=session^M c=IN IP4 xxx.xxx.xxx.xxx^M t=0 0^M m=audio 19536 RTP/AVP 0 8 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:8 PCMA/8000^M
Re: [asterisk-users] No response to our critical packet problem
Hi Guys, I just wanted to let you all know that you were indeed correct, it was the SIP INFO '#' that was causing the problem. You'll pardon me, but I find this problem _utterly ridiculous_. I am running asterisk v1.4.18. Are there any asterisk versions that this is fixed on? Thanks. (Oh and please CC me, I'm reading in digest mode..) -- James On Fri, May 22, 2009 at 10:36 AM, James Lamanna jlama...@gmail.com wrote: Hi, I have a strange problem. At a site where there are 20+ phones, there is one phone that cannot make outbound (to PSTN) calls. Each call is dropped after 20s with no response to our critical packet. Calls to voicemail and internal extensions work fine. I understand that everything points to a NAT problem, but I don't understand how it could be because: 1) It does not affect calls to internal office extensions (which still go through asterisk) OR voicemail 2) The other 20+ phones in the same office on the same network have 0 problems. Here's a SIP trace of the problem. yyy.yyy.yyy.yyy is the outside NAT IP xxx.xxx.xxx.xxx is the IP of my PBX dd is the dialed phone number sss is the source phone number The peculiar thing is that asterisk sends an OK in response to an INVITE, then the phone sends back an ACK, which asterisk seems to ignore because it retransmits the OK message again Then eventually the phone gives up and sends a BYE message. -- James --- SIP read from yyy.yyy.yyy.yyy:24050 --- INVITE sip:ddd...@xxx.xxx.xxx.xxx SIP/2.0^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-6e730c81^M From: sss-sss- ;tag=bdfe4214c494d109o0^M To: ^M Call-ID: c4560330-de7ca...@10.1.24.145^m CSeq: 101 INVITE^M Max-Forwards: 70^M Contact: sss-sss- ^M Expires: 240^M User-Agent: Linksys/SPA942-6.1.3(a)^M Content-Length: 395^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: replaces^M Content-Type: application/sdp^M ^M v=0^M o=- 6363534 6363534 IN IP4 10.1.24.145^M s=-^M c=IN IP4 10.1.24.145^M t=0 0^M m=audio 16458 RTP/AVP 0 2 4 8 18 96 97 98 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:2 G726-32/8000^M a=rtpmap:4 G723/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:18 G729a/8000^M a=rtpmap:96 G726-40/8000^M a=rtpmap:97 G726-24/8000^M a=rtpmap:98 G726-16/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-15^M a=ptime:20^M a=sendrecv^M - --- Reliably Transmitting (NAT) to yyy.yyy.yyy.yyy:24050 --- SIP/2.0 407 Proxy Authentication Required^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-6e730c81;received=yyy.yyy.yyy.yyy^M From: sss-sss- ;tag=bdfe4214c494d109o0^M To: ;tag=as70a8455c^M Call-ID: c4560330-de7ca...@10.1.24.145^m CSeq: 101 INVITE^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Supported: replaces^M Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=6d2db4b7^M Content-Length: 0^M --- SIP read from yyy.yyy.yyy.yyy:24050 --- ACK sip:ddd...@xxx.xxx.xxx.xxx SIP/2.0^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-6e730c81^M From: sss-sss- ;tag=bdfe4214c494d109o0^M To: ;tag=as70a8455c^M Call-ID: c4560330-de7ca...@10.1.24.145^m CSeq: 101 ACK^M Max-Forwards: 70^M Contact: sss-sss- ^M User-Agent: Linksys/SPA942-6.1.3(a)^M Content-Length: 0^G ^M --- SIP read from yyy.yyy.yyy.yyy:24050 --- INVITE sip:ddd...@xxx.xxx.xxx.xxx SIP/2.0^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-3d87585d^M From: sss-sss- ;tag=bdfe4214c494d109o0^M To: ^M Call-ID: c4560330-de7ca...@10.1.24.145^m CSeq: 102 INVITE^M Max-Forwards: 70^M Proxy-Authorization: Digest username=ss,realm=asterisk,nonce=6d2db4b7,uri=sip:ddd...@xxx.xxx.xxx.xxx,algorithm=MD5,response= Contact: sss-sss- ^M Expires: 240^M User-Agent: Linksys/SPA942-6.1.3(a)^M Content-Length: 395^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: replaces^M Content-Type: application/sdp^M ^M v=0^M o=- 6363534 6363534 IN IP4 10.1.24.145^M s=-^M c=IN IP4 10.1.24.145^M t=0 0^M m=audio 16458 RTP/AVP 0 2 4 8 18 96 97 98 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:2 G726-32/8000^M a=rtpmap:4 G723/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:18 G729a/8000^M a=rtpmap:96 G726-40/8000^M a=rtpmap:97 G726-24/8000^M a=rtpmap:98 G726-16/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-15^M a=ptime:20^M a=sendrecv^M - --- Transmitting (NAT) to yyy.yyy.yyy.yyy:24050 --- SIP/2.0 100 Trying^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-3d87585d;received=yyy.yyy.yyy.yyy^M From: sss-sss- ;tag=bdfe4214c494d109o0^M To: ^M Call-ID: c4560330-de7ca...@10.1.24.145^m CSeq: 102 INVITE^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Supported: replaces^M Contact: ^M Content-Length: 0^M ^M --- Transmitting (NAT) to yyy.yyy.yyy.yyy:24050 --- SIP/2.0 183 Session Progress^M Via: SIP/2.0/UDP
[asterisk-users] Too many notify events causing Asterisk crash?
Hi, We've implemented a 'page-all' function for some of our customers, and we've noticed that on occasion the page-all will cause asterisk to crash (safe_asterisk then restarts it again). The particular customer has about 20 phones, and also has 5 Linksys 932 to monitor the state of these extensions. I'm not sure whether it is the page-all that causes the crash, or the subsequent NOTIFY storm to all of the 932s that are monitoring those extensions (since the page-all causes them all to go to In-Use and then Idle). Here's a log snippet of right before the crash (previous crashes look very similar). Also, this is a production box, so there's no way at this time I can recompile with debugging symbols. The Asterisk version is 1.4.18.1. Thanks. [Mar 14 12:56:06] VERBOSE[30141] logger.c: == Spawn extension (ext-paging, PAGExx6295, 8) exited non-zero on 'Local/pagexx6...@ext-paging-ef0f,2' [Mar 14 12:56:08] VERBOSE[30193] logger.c: == Connect attempt from '127.0.0.1' unable to authenticate [Mar 14 12:56:13] NOTICE[8327] chan_sip.c: Registration from 'sip:1...@warp2biz.com' failed for '71.119.123.229' - Wrong password [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6297[ext-local] new state Idle for Notify User xx6324 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6297[ext-local] new state Idle for Notify User xx6293 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6297[ext-local] new state Idle for Notify User xx6311 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6297[ext-local] new state Idle for Notify User xx6315 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6297[ext-local] new state Idle for Notify User xx6297 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6329[ext-local] new state Idle for Notify User xx6324 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6329[ext-local] new state Idle for Notify User xx6293 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6329[ext-local] new state Idle for Notify User xx6311 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6329[ext-local] new state Idle for Notify User xx6315 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6329[ext-local] new state Idle for Notify User xx6297 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6295[ext-local] new state Idle for Notify User xx6324 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6295[ext-local] new state Idle for Notify User xx6293 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6295[ext-local] new state Idle for Notify User xx6311 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6295[ext-local] new state Idle for Notify User xx6315 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6295[ext-local] new state Idle for Notify User xx6297 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6317[ext-local] new state Idle for Notify User xx6324 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6317[ext-local] new state Idle for Notify User xx6293 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6317[ext-local] new state Idle for Notify User xx6311 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6317[ext-local] new state Idle for Notify User xx6315 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6317[ext-local] new state Idle for Notify User xx6297 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6328[ext-local] new state Idle for Notify User xx6324 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6328[ext-local] new state Idle for Notify User xx6293 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6328[ext-local] new state Idle for Notify User xx6315 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6328[ext-local] new state Idle for Notify User xx6311 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6328[ext-local] new state Idle for Notify User xx6297 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6314[ext-local] new state Idle for Notify User xx6324 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6314[ext-local] new state Idle for Notify User xx6293 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6314[ext-local] new state Idle for Notify User xx6311 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6314[ext-local] new state Idle for Notify User xx6315 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed
[asterisk-users] Asterisk and sip router integration
Hi, Does anyone have some good examples of a Kamalio or OpenSips configuration that integrates with Asterisk? Essentially I want to use the SIP router as the UA, but still run all the calls through Asterisk (for dialplan, etc..) I've looked for examples on the project web sites, but I haven't found anything decent yet. Thanks. -- James ___ -- 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
[asterisk-users] Blind transfer from asterisk dialplan (and problems re-parking a call)
Hi,
Is there a way to do a blind transfer within an asterisk dialplan (like '##')?
The reason I need this (I think) rather than a regular Goto() is that
I'm trying to do one-touch parking.
I can park a call using one-touch parking and then pick it up again,
however if I try to re-park the call, it gets lost.
I think that is because asterisk thinks I'm still on the park extension.
As an example:
exten = _9X,1,Set(PARKINGEXTEN=${EXTEN})
exten = _9X,n,Set(RETURNEXT=${CUT(BLINDTRANSFER||1):4})
exten = _9X,n,GotoIf($[x${RETURNEXT} = x]?usechannel:find)
exten = _9X,n(usechannel),Set(RETURNEXT=${CUT(CHANNEL||1):4})
exten =
_9X,n(park),ParkAndAnnounce(pbx-transfer:PARKED|30|Local/parkannounce|ext-local,${RETURNEXT},1)
So here's what happens:
Someone calls in, the call is answered. A user speed dials using a
line key to extension 90, which parks the call on 90.
The BLF (linksys phones) lights up for that key, indicating the call is parked.
Going to another phone monitoring the 90 extension, you can unpark the
call by pressing the line key (which speeddials 90).
However, pressing that key again to re-park on 90 does not work, it
puts the phone on hold (because I think the phone thinks its on the
same extension).
Any ideas here?
Basically I have a customer that wants one-touch parking w/o having to
wait for an announcement.
Thanks.
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[asterisk-users] One way audio after IVR tree
Hi, I have a couple of users who are having a peculiar problem. On some outbound numbers where there is a deep IVR tree (3+ selections), and then a live person picks up, the live person will be unable to hear them on the phone, but they can hear the live person. I've done packet traces and it appears as though audio is being passed both ways, but the audio from the caller is severely muted before it gets to asterisk. Has anyone seen this before? It's almost like the phone thinks its still sending DTMF or something and mutes the audio. I've seen this happen on both linksys 942 and 962 phones. Thanks. -- James ___ -- 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
[asterisk-users] Minimum version for asterisk and iaxmodem
Hi, I'm trying to use iaxmodem against a very old version of asterisk (1.0.7 - its a debian sarge embedded system), yet when asterisk gets a call from iaxmodem, it says that the format for the call is unknown. Does anyone know if there is a minimum version of asterisk that is compatible with iaxmodem 1.1.0? Thanks. -- James ___ -- 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
Re: [asterisk-users] Minimum version for asterisk and iaxmodem
On Sat, Feb 7, 2009 at 1:19 PM, Lee Howard fax...@howardsilvan.com wrote: James Lamanna wrote: Hi, I'm trying to use iaxmodem against a very old version of asterisk (1.0.7 - its a debian sarge embedded system), yet when asterisk gets a call from iaxmodem, it says that the format for the call is unknown. Does anyone know if there is a minimum version of asterisk that is compatible with iaxmodem 1.1.0? I originally developed IAXmodem while using Asterisk 1.2.x. I have since migrated to Asterisk 1.4.x. I never attempted to use Asterisk 1.0.x with IAXmodem, and I have also never tried Asterisk 1.6.x (although I suspect others have without issue). Thanks Lee. I'm compiling 1.2.31 right now to see if I have more luck (1.4.x is giving me trouble w/ iaxmodem + hylafax on this ARM platform, but it works great on x86!). Thanks, Lee. -- James ___ -- 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
Re: [asterisk-users] Minimum version for asterisk and iaxmodem
So I got 1.2.31 compiled but I still can't do a loopback fax with
IAXModem, Hylafax and Asterisk.
The call connects, but the iaxmodem log fills up with Adjusting skew
xxx and the fax ultimately
fails with:
Feb 07 14:10:26.05: [ 3497]: MODEM No carrier
Feb 07 14:10:26.05: [ 3497]: Failure to receive silence
(synchronization failure). {E100}
Feb 07 14:10:26.05: [ 3497]: RECV FAX: Failure to receive silence
(synchronization failure). {E100}
Feb 07 14:10:26.05: [ 3497]: RECV FAX: end
Feb 07 14:10:26.05: [ 3497]: Failure to receive silence
(synchronization failure). {E100}
Feb 07 14:10:26.05: [ 3497]: SESSION END
-- Accepting AUTHENTICATED call from 127.0.0.1:
requested format = ulaw,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw),
priority = mine
-- Executing Dial(IAX2/iaxmodem0-11201,
IAX2/iaxmodem1/xx|60) in new stack
-- Called iaxmodem1/6266053472
-- Call accepted by 127.0.0.1 (format ulaw)
-- Format for call is ulaw
-- IAX2/iaxmodem1-1796 is ringing
-- IAX2/iaxmodem1-1796 answered IAX2/iaxmodem0-11201
-- Attempting native bridge of IAX2/iaxmodem0-11201 and IAX2/iaxmodem1-1796
On Sat, Feb 7, 2009 at 1:44 PM, James Lamanna jlama...@gmail.com wrote:
On Sat, Feb 7, 2009 at 1:19 PM, Lee Howard fax...@howardsilvan.com wrote:
James Lamanna wrote:
Hi,
I'm trying to use iaxmodem against a very old version of asterisk
(1.0.7 - its a debian sarge embedded system),
yet when asterisk gets a call from iaxmodem, it says that the format
for the call is unknown.
Does anyone know if there is a minimum version of asterisk that is
compatible with iaxmodem 1.1.0?
I originally developed IAXmodem while using Asterisk 1.2.x. I have since
migrated to Asterisk 1.4.x. I never attempted to use Asterisk 1.0.x with
IAXmodem, and I have also never tried Asterisk 1.6.x (although I suspect
others have without issue).
Thanks Lee.
I'm compiling 1.2.31 right now to see if I have more luck (1.4.x is
giving me trouble w/
iaxmodem + hylafax on this ARM platform, but it works great on x86!).
Thanks,
Lee.
-- James
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[asterisk-users] Running asterisk on ARM (TS-7800) 1.4.23.1
Hi, I'm trying to run Asterisk 1.4.23.1 on a small ARM linux board (TS-7800). Everything compiles fine, but on startup Asterisk always crashes while loading chan_sip. If chan_sip is removed, it starts up fine, but I really need SIP to work. Any ideas? Thanks. -- James ___ -- 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
