[asterisk-users] compile error: implicit declaration of function drv_dbg
hello,
I try to open the debug to compile dahdi with wcb4xxp,
=base.c
#ifdef DEBUG_LOWLEVEL_REGS
if (unlikely(DBG_REGS))
drv_dbg(b4->dev, "read 0x%02x from 0x%p\n", ret, b4->addr +
reg);
#endif
if (unlikely(pedanticpci)) {
udelay(3);
}
/usr/src/dahdi-linux/drivers/dahdi/wcb4xxp/base.c: In function __pci_in8:
/usr/src/dahdi-linux/drivers/dahdi/wcb4xxp/base.c:150: error: implicit
declaration of function drv_dbg
make[3]: *** [/usr/src/dahdi-linux/drivers/dahdi/wcb4xxp/base.o] Error 1
make[2]: *** [/usr/src/dahdi-linux/drivers/dahdi/wcb4xxp] Error 2
make[1]: *** [_module_/usr/src/dahdi-linux/drivers/dahdi] Error 2
make[1]: Leaving directory `/usr/src/kernels/2.6.18-8.el5-i686'
--
anybody knows how to drv_dbg in dahdi?
thanks!
lizhon
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Re: [asterisk-users] Multiple Agent Login
Sorry, I forgot to mention that I am using Asterisk 1.2.30 Hi friends, Do we have any way to prevent more than one Agent being logged in from the same extension? Also is there a way to limit an agent from logging in from more than one extension? I searched too much, but didn't find a solution. Please help. Thanks in advance. Shanavaz. I'm sure someone will tell me what is wrong with this answer, but in 1.4, agents login to Call Queues, not extensions. How you set up the queue determines whether the agent can login from more than one extension. If the agent's phone has two or three lines, he/she could be set up to have each line go to a different queue or both lines to the same queue. ___ -- 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] recording (mixmonitor) stopped of transfer/call parking after queue
Hi all, I enabled recording (mixmonitor) in queue and process started after queue member pick the call. But recording will stop after picking up by another extensions of call transfer/parking in the same call. Is it possible to continue to record the call for call parking/transfer, how? Rgds, ango ___ -- 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] Grandstream speakerphone?
I have been using a number of the Grandstream GXP-2000 (74 in production), GXP-2010 (1 in production), and BT-200 (15 in production) with great success. The only issue that we have had is killing power supplies, not sure if this is related to our power source or product. So far they have replaced all of them for us. I prefer the speaker phone on the GXP-2010 and the BT-200 phones. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio Sent: Wednesday, March 11, 2009 5:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Grandstream speakerphone? Idle curiosity: I like the look and feel of the Grandstreams, but it's been my experience that the speakerphones suck (esp. when compared to the pretty damn flawless Polycoms). I've used the BT-100/101 and GS-2000; have any of their newer models changed that? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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 ___ -- 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] Are .call files working with extensions.ael ?
On Wed, Mar 11, 2009 at 5:29 PM, Olivier wrote:
> Hello,
>
> With an extensions.ael enabled system, I keep getting whatever I change
> into my "astup.call" file :
>
> [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At least
> one of app or extension (or keyword message/pdu) must be specified, along
> with tech and dest in file /var/spool/asterisk/outgoing/astup.call
> [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:457 scan_service: Invalid file
> contents in /var/spool/asterisk/outgoing/astup.call, deleting
> [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:505 scan_thread: Failed to
> scan service '/var/spool/asterisk/outgoing/astup.call'
>
Olivier--
It's complaining that you don't have "Extension: " and "Priority:
. " lines in your call file, along with the context,
The Channel: lines calls one phone, the Context, Extension, and priority say
what to execute for the other channel,
and the two are bridged.
Whether the context, exten, and priority specified are in an AEL supplied
dialplan or an extensions.conf
dialplan, doesn't matter. You can even mix both together to form a dialplan.
Let's see, I have a call file laying around...
Channel: Sip/snom
Context: workext
Extension: 983075878001
Priority: 1
...
This will ring the phone specified in Channel, and when it answers, it will
run the extension you specify, and connect the two. (in this case it will
dial the "movie hot line" in Cody, WY, and the leading "98" says to use
a certain ISP to place the call.
murf
>
> With an extensions.conf enabled system, the same "astup.call" file would
> work.
>
> Has anyone tried ?
> Any hint ?
>
> Channel: sip/7...@mylocal
> CallerID: 692 <692>
> MaxRetries: 1
> WaitTime: 60
> RetryTime: 5
> Context: mylocal
> Extension: 00123457530
> #Priority: 1
>
> I suppose I should have written "mylocal" context in a different way as my
> extensions.ael includes :
>
> context mylocal {
> includes {
> subs;
> };
>
>700 => ...
> };
>
> Regards
>
> ___
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>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Steve Murphy
ParseTree Corp
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[asterisk-users] Are .call files working with extensions.ael ?
Hello,
With an extensions.ael enabled system, I keep getting whatever I change into
my "astup.call" file :
[Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At least
one of app or extension (or keyword message/pdu) must be specified, along
with tech and dest in file /var/spool/asterisk/outgoing/astup.call
[Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:457 scan_service: Invalid file
contents in /var/spool/asterisk/outgoing/astup.call, deleting
[Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:505 scan_thread: Failed to scan
service '/var/spool/asterisk/outgoing/astup.call'
With an extensions.conf enabled system, the same "astup.call" file would
work.
Has anyone tried ?
Any hint ?
Channel: sip/7...@mylocal
CallerID: 692 <692>
MaxRetries: 1
WaitTime: 60
RetryTime: 5
Context: mylocal
Extension: 00123457530
#Priority: 1
I suppose I should have written "mylocal" context in a different way as my
extensions.ael includes :
context mylocal {
includes {
subs;
};
700 => ...
};
Regards
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Re: [asterisk-users] Grandstream speakerphone?
I just got some GXP2000s and they seem to have decent speaker phones. I think I saw something about "improved speaker phone" in the sales lit. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio Sent: Wednesday, March 11, 2009 4:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Grandstream speakerphone? Idle curiosity: I like the look and feel of the Grandstreams, but it's been my experience that the speakerphones suck (esp. when compared to the pretty damn flawless Polycoms). I've used the BT-100/101 and GS-2000; have any of their newer models changed that? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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 ___ -- 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] Grandstream speakerphone?
On 12/03/2009 10:06 a.m., Ken D'Ambrosio wrote: > Idle curiosity: I like the look and feel of the Grandstreams, but it's > been my experience that the speakerphones suck (esp. when compared to the > pretty damn flawless Polycoms). I've used the BT-100/101 and GS-2000; > have any of their newer models changed that? We've been trialing some in our office, and the newer models sound nicer than the old. We had a problem when dialing a number quickly although haven't been able to repeat it. The Polycoms are still nicer though. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- 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] Grandstream speakerphone?
Idle curiosity: I like the look and feel of the Grandstreams, but it's been my experience that the speakerphones suck (esp. when compared to the pretty damn flawless Polycoms). I've used the BT-100/101 and GS-2000; have any of their newer models changed that? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] Digium B410P: misdn v1 or misdn v2 or dahdi + asterisk 1.6 ?
On Wed, 11 Mar 2009, Vieri wrote: > > Hi, > > Until now I've been using my Digium B410P cards with misdn 1.0.x. > > I would like to upgrade my systems and am now wondering which is the > "best" route to take: If it aint broke, don't fix it... Saying that, I can feel a need even now to look at 1.4 - I have many 1.2 systems out there using mISDN but there are 1 or 2 who seem to suffer the "stuck channel" thing - only with Snom phones though, never Grandstream... And I'm told 1.4 has cured all this, however ... Good luck with what you choose ... Gordon ___ -- 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] VLC
On Wed, 11 Mar 2009, Bex Vincent wrote: > When our users receive a voicemail we send it attached to an email. It > used to work fine, encoded in wav49 and read by Windows media player. > Recently the default player in the company has become VLC which is > unable to read wav49. I am trying to use OGG/VORBIS instead of wav49. I > can't get it working: Would "format = wav" do? While the files are 10x bigger, it's still only 1mb per minute. Unless you have to retain the files for a very long time and/or have a huge number of users, I can't see spending the CPU time to compress and decompress something that will probably only be listened to once and discarded. Personally, the increase in fidelity is good enough reason to me. Try "format = wav|wav49" and listen to the files with a decent set of speakers. I know the typical handset approaches two cans and a piece of wet string fidelity wise, but, since you say "attached to an email," your users will hear the difference. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Digium B410P: misdn v1 or misdn v2 or dahdi + asterisk 1.6 ?
On Wed, Mar 11, 2009 at 02:56:58PM -0500, Kevin P. Fleming wrote: > Vieri wrote: > > > - use the latest release of misdn v1 > > - upgrade to the latest "stable" kernel and use the built-in misdn v2 > > There is no support for mISDN v2 in Asterisk to my knowledge. It is available in a separate, out of tree module that is developed alongside mISDNv2 (chan_lcr). IIRC, though, hfcmulti is not yet supported by mISDNv2 . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] Digium B410P: misdn v1 or misdn v2 or dahdi + asterisk 1.6 ?
Vieri wrote: > - use the latest release of misdn v1 > - upgrade to the latest "stable" kernel and use the built-in misdn v2 There is no support for mISDN v2 in Asterisk to my knowledge. > - use misdn v2 as a seperate package (disable misdn in the kernel) See above. > - use dahdi's support for misdn with asterisk 1.6 (not sure if this is true) DAHDI does not have anything to do with mISDN. In Asterisk 1.6. the B410P card can be used with either chan_misdn and mISDN v1 or with the DAHDI wcb4xxp driver and chan_dahdi (using libpri for signaling) without mISDN being involved at all. The only major caveat with chan_dahdi/libpri is that NT-PtMP mode is not supported, but both TE modes are supported and NT-PTP mode is as well. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org ___ -- 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] Digium B410P: misdn v1 or misdn v2 or dahdi + asterisk 1.6 ?
Hi, Until now I've been using my Digium B410P cards with misdn 1.0.x. I would like to upgrade my systems and am now wondering which is the "best" route to take: - use the latest release of misdn v1 - upgrade to the latest "stable" kernel and use the built-in misdn v2 - use misdn v2 as a seperate package (disable misdn in the kernel) - use dahdi's support for misdn with asterisk 1.6 (not sure if this is true) At a first glance I think I may be better off with the latest misdn v1 but would like to know what other B410P users think about it. Thanks, Vieri ___ -- 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] update on Odd occurrence
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Wednesday, March 11, 2009 1:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] update on Odd occurrence On Wed, 11 Mar 2009, Danny Nicholas wrote: > I upgraded the E1000 driver on my machine from 7.3.20-k2-NAPI to > 8.0.9-NAPI. This unfortunately did nothing to resolve the problem. > The best workarounds I've come up with are: > > 1. use -l on scp and ftp > > 2. install wondershaper QOS and limit throughput to 32K. > > These are workarounds, but I'd really like a solution. I resolved my issue with an Intel mobo and an integrated e1000 by installing a Realtek card. Not my idea of a solution either. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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 Realtek? - Couldn't find a "real" network card? BTW, reducing the speed of card from 1000 to 100 seems to help if not solve the problem. ___ -- 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] update on Odd occurrence
On Wed, 11 Mar 2009, Danny Nicholas wrote: > I upgraded the E1000 driver on my machine from 7.3.20-k2-NAPI to > 8.0.9-NAPI. This unfortunately did nothing to resolve the problem. > The best workarounds I've come up with are: > > 1. use -l on scp and ftp > > 2. install wondershaper QOS and limit throughput to 32K. > > These are workarounds, but I'd really like a solution. I resolved my issue with an Intel mobo and an integrated e1000 by installing a Realtek card. Not my idea of a solution either. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] update on Odd occurrence
Hi gang, I upgraded the E1000 driver on my machine from 7.3.20-k2-NAPI to 8.0.9-NAPI. This unfortunately did nothing to resolve the problem. The best workarounds I've come up with are: 1. use -l on scp and ftp 2. install wondershaper QOS and limit throughput to 32K. These are workarounds, but I'd really like a solution. ___ -- 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] Multiple Agent Login
Hi, on file agents.conf use the option multiplelogin=no -- Humberto Figuera - Using Linux 2.6.26 Usuario GNU/Linux 369709 Caracas - Venezuela GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603 ___ -- 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] Problem with incoming and outgoing calls via TDM
On Wed, 11 Mar 2009, Rosa De Santis wrote: > > Hello all. > > Please, I'd like to know if somebody can help me with this problem. > I have successfully configured a PBX with Asterisk 1.4 and a Digium analog > card with 4 ports. > > This PBX has a lot of incoming and outgoing calls, and works perfect in > general, but there are some extrange cases where an incoming call is > bridget with an outgoing call, and the caller that is calling TO the PBX > can even hear the dtmf tones of the caller that is calling OUT the PBX, > and due the high traffic this is happening a lot. It seems that asterisk > is taking the zap channel to call out in the exact moment before it is > marked as busy with the incoming call. Please, is there any > configuration to avoid this? It's called "glare" and your options to "fix" it partly depend on what country you are in. I don't think it's totally fixable with analogue lines. Make sure you have the right country code specified in /etc/zaptel.conf and are using "Kewlstart". Eg. for the UK: fxsks=1 fxsks=2 fxsks=3 fxsks=4 loadzone=uk defaultzone=uk Your options are to replace the 'ks' with 'gs' or 'ls' - but others might be able to advise you what's best for your country/telco. But do make sure your incoming calls start at one end, and outgoing start at the other - so if the first line is connected to port 1, 2nd to port 2, etc. you want to dial-out starting at port 4 - that's the capital G option in dial - eg. dial(Zap/G1/...) the lower-case g will start outbound dialing at the lower port number. (And put the lines in the right group in /etc/asterisk/zapata.conf) Another solution might be to get more channels - are your users & callers complaining of busy tones? Gordon ___ -- 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] Problem with incoming and outgoing calls via TDM
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Wednesday, March 11, 2009 11:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with incoming and outgoing calls via TDM Rosa De Santis wrote: > Hello all. > > Please, I'd like to know if somebody can help me with this problem. > I have successfully configured a PBX with Asterisk 1.4 and a Digium analog card with 4 ports. > > This PBX has a lot of incoming and outgoing calls, and works perfect in general, but there are some extrange cases where an incoming call is bridget with an outgoing call, and the caller that is calling TO the PBX can even hear the dtmf tones of the caller that is calling OUT the PBX, and due the high traffic this is happening a lot. > It seems that asterisk is taking the zap channel to call out in the exact moment before it is marked as busy with the incoming call. > Please, is there any configuration to avoid this? > > Thanks a lot in advance. > Rosa. The situation you're referring to is called glare. You'll find discussion of it in the archives and on voip-info.org. You need to make sure you are seizing lines for outgoing calls in the reverse order that they are used for incoming calls. Check out the G dialing option for Zaptel/DAHDI channels (under Dialing a Group section): http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels If this doesn't work, your next best bet is to increase the number of lines you have. -Dave ___ -- 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 That's a good answer, Dave, but the problem could be as simple as the Dial command using Dial(Zap/1,X) instead of Dial(Zap/g1,X). Or the Zapata.conf (dadhi.conf) may have busydetect = no. ___ -- 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] Problem with incoming and outgoing calls via TDM
Rosa De Santis wrote: > Hello all. > > Please, I'd like to know if somebody can help me with this problem. > I have successfully configured a PBX with Asterisk 1.4 and a Digium analog > card with 4 ports. > > This PBX has a lot of incoming and outgoing calls, and works perfect in > general, but there are some extrange cases where an incoming call is bridget > with an outgoing call, and the caller that is calling TO the PBX can even > hear the dtmf tones of the caller that is calling OUT the PBX, and due the > high traffic this is happening a lot. > It seems that asterisk is taking the zap channel to call out in the exact > moment before it is marked as busy with the incoming call. > Please, is there any configuration to avoid this? > > Thanks a lot in advance. > Rosa. The situation you're referring to is called glare. You'll find discussion of it in the archives and on voip-info.org. You need to make sure you are seizing lines for outgoing calls in the reverse order that they are used for incoming calls. Check out the G dialing option for Zaptel/DAHDI channels (under Dialing a Group section): http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels If this doesn't work, your next best bet is to increase the number of lines you have. -Dave ___ -- 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] Problem with incoming and outgoing calls via TDM
Hello all. Please, I'd like to know if somebody can help me with this problem. I have successfully configured a PBX with Asterisk 1.4 and a Digium analog card with 4 ports. This PBX has a lot of incoming and outgoing calls, and works perfect in general, but there are some extrange cases where an incoming call is bridget with an outgoing call, and the caller that is calling TO the PBX can even hear the dtmf tones of the caller that is calling OUT the PBX, and due the high traffic this is happening a lot. It seems that asterisk is taking the zap channel to call out in the exact moment before it is marked as busy with the incoming call. Please, is there any configuration to avoid this? Thanks a lot in advance. Rosa. _ Get 5 GB of storage with Windows Live Hotmail. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5gb_112008___ -- 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] Portech MV3770 & Caller-ID
On Wed, Mar 11, 2009 at 04:16:43PM +0100, Christian Victor wrote: > 2009/3/11 Håkan Källberg > > Does anyone of you have Caller Presentation working in the other > > direction?? My mv370 is working well, execpt the Caller ID on outgoing > > GSM calls. This works with the SIM card/Provider I am using if I put > > the SIM card in a telephone, but not in mv370. I have tried options on the > > mobile setting page you talked about, with nor difference. I have > > tried to put the nummber of the SIM card as the Caller ID in the original > > Asterisk call too, just in case, but no. > > All I had to do is to enable the Caller ID ind the Mobile->Settings dialog > for each SIM (something like presentation/revocation afair). I did NOT set > the GSM number anywhere nor do I send it from Asterisk. That is what I'd expect too, but, no... Mobile->Settings->CLID Presentation-> Supression or Invocation it makes no difference. (and yes - I do reboot:-) When I move the SIM to a phone, it works well... I have the latest firmware: Firmware Version:Fri Sep 5 09:02:30 2008 And by the why a current 1.4 Asterisk, just updatdet to .23.2. Viele Grüße:Håkan pgpoJRRjXnEBy.pgp Description: PGP signature ___ -- 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] Portech MV3770 & Caller-ID
2009/3/11 Håkan Källberg > > Hello! > > Does anyone of you have Caller Presentation working in the other > direction?? My mv370 is working well, execpt the Caller ID on outgoing > GSM calls. This works with the SIM card/Provider I am using if I put > the SIM card in a telephone, but not in mv370. I have tried options on the > mobile setting page you talked about, with nor difference. I have > tried to put the nummber of the SIM card as the Caller ID in the original > Asterisk call too, just in case, but no. > All I had to do is to enable the Caller ID ind the Mobile->Settings dialog for each SIM (something like presentation/revocation afair). I did NOT set the GSM number anywhere nor do I send it from Asterisk. Don't forget to save and reboot the gateway. Chris ___ -- 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] Multiple Agent Login
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shanavaz E A Sent: Wednesday, March 11, 2009 9:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Multiple Agent Login Hi friends, Do we have any way to prevent more than one Agent being logged in from the same extension? Also is there a way to limit an agent from logging in from more than one extension? I searched too much, but didn't find a solution. Please help. Thanks in advance. Shanavaz. I'm sure someone will tell me what is wrong with this answer, but in 1.4, agents login to Call Queues, not extensions. How you set up the queue determines whether the agent can login from more than one extension. If the agent's phone has two or three lines, he/she could be set up to have each line go to a different queue or both lines to the same queue. ___ -- 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] Sending faxes with T.38 problem. Asterisk - 1.6.0.6
Santiago Gimeno wrote: > I finally solved the issue by changing the resolution and the width of > the TIFF file to one that is accepted by the fax standard. In my case > I changed to a resolution of 96x96 and a width of 1728. > > Now I am able to send faxes, but something weird is happening, the fax > received in the fax-machine has the black and white colours inverted. > Any ideas why this could be happening? There is a optional parameter in TIFF files which swaps black and white. It seldom affects people, so it was only fixed very recently. spandsp-0.0.6pre6 automatically allows for this swapping. Alternatively, whatever you use to generate the TIFF can probably be set to generate files with black and white the normal way around. 96x96 is not a valid FAX resolution. Did you mean 204dpi x 98dpi, or 204dpi x 196dpi? Those are the valid standard and fine resolutions for FAX. Steve ___ -- 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] Multiple Agent Login
Hi friends, Do we have any way to prevent more than one Agent being logged in from the same extension? Also is there a way to limit an agent from logging in from more than one extension? I searched too much, but didn't find a solution. Please help. Thanks in advance. Shanavaz. ___ -- 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] VLC
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bex Vincent Sent: Wednesday, March 11, 2009 3:57 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VLC Hi All, When our users receive a voicemail we send it attached to an email. It used to work fine, encoded in wav49 and read by Windows media player. Recently the default player in the company has become VLC which is unable to read wav49. I am trying to use OGG/VORBIS instead of wav49. I can't get it working: In voicemail.conf: format = ogg The result is as follow: [Mar 11 09:42:17] WARNING[24867]: format_ogg_vorbis.c:527 ogg_vorbis_seek: Seeking is not supported on OGG/Vorbis streams! -- x=0, open writing: /var/spool/asterisk/voicemail/default/0213149689/tmp/V2adNF format: ogg, 0x81ec648 -- User hung up [Mar 11 09:42:25] WARNING[24867]: format_ogg_vorbis.c:533 ogg_vorbis_tell: Telling is not supported on OGG/Vorbis streams! [Mar 11 09:42:25] WARNING[24867]: format_ogg_vorbis.c:514 ogg_vorbis_trunc: Truncation is not supported on OGG/Vorbis streams! -- Recording was 0 seconds long but needs to be at least 2 - abandoning Nothing gets recorded in the file. Has anybody done this before? Either get ogg work with voicemail or get VLC to read wav49. Cheers Vincent Why don't you just install SOX and convert the file that way? You could just do a command like this System(/usr/bin/sox in.ogg out.mp3) ___ -- 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] Portech MV3770 & Caller-ID
On Tue, Mar 10, 2009 at 02:11:58PM +0100, Christian Victor wrote: > 2009/3/10 Sasa > > > Hi, I have modified in Mobile/Setting the parameter "SIP From" from > > "tel/user" to "tel/tel" and now I view the correct incoming number. > > Thanks. > > > > Glad I could help. It took me nearly a month to figure that out. ;-) Hello! Does anyone of you have Caller Presentation working in the other direction?? My mv370 is working well, execpt the Caller ID on outgoing GSM calls. This works with the SIM card/Provider I am using if I put the SIM card in a telephone, but not in mv370. I have tried options on the mobile setting page you talked about, with nor difference. I have tried to put the nummber of the SIM card as the Caller ID in the original Asterisk call too, just in case, but no. All I want, or rather all that would be possible is to show the number of the SIM card. Then I know that I get a transferred call from my Asterisk. It would be wonderull to get the number of the original caller too, but this will not be allowed by the provider, and this is really a Good Thing too. Håkan pgpUX8PMN2ftJ.pgp Description: PGP signature ___ -- 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] Sending faxes with T.38 problem. Asterisk - 1.6.0.6
I finally solved the issue by changing the resolution and the width of the
TIFF file to one that is accepted by the fax standard. In my case I changed
to a resolution of 96x96 and a width of 1728.
Now I am able to send faxes, but something weird is happening, the fax
received in the fax-machine has the black and white colours inverted. Any
ideas why this could be happening?
Best regards,
Santi
On Tue, Mar 10, 2009 at 6:53 PM, Santiago Gimeno
wrote:
> Thanks for the tip. Sadly, it didn't work. I keep getting the same error:
>
> [Mar 10 18:49:48] WARNING[18855]: app_fax.c:176 phase_e_handler: Error
> transmitting fax. result=11: Far end cannot receive at the resolution of the
> image.
>
> regards,
>
> Santi
>
>
> On Tue, Mar 10, 2009 at 6:36 PM, Matthew Fredrickson
> wrote:
>
>> Santiago Gimeno wrote:
>> > Hello,
>> >
>> > Thanks everybody for the answers.
>> >
>> > >Could be. Would you post the Cisco config relevant to this?
>> >
>> > dial-peer voice 5 voip
>> > description ** **
>> > preference 1
>> > destination-pattern 1…
>> > voice-class codec 1
>> > session protocol sipv2
>> > session target ipv4:1.1.1.1
>> > session transport udp
>> > dtmf-relay rtp-nte
>> > fax-relay ecm disable
>>
>> I think, that at least if you're using T.38, you may want to try
>> enabling ECM. ECM can cause significant problems in a high-packet loss,
>> non-T.38 environment, but I would think that in a T.38 environment, if
>> you can keep ECM enabled, that would be a good thing.
>>
>> Matthew Fredrickson
>> Digium, Inc.
>>
>> > fax nsf 00
>> > fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through
>> > g711alaw
>> > no vad
>> >
>> >
>> > >And upon further examination... don't put T38CALL in as a variable. It
>> > will cause the initial INVITE to only
>> > >have T38. Leave it out and things should hopefully reinvite.
>> >
>> > I have removed the T38CALL variable and it looks better but it still
>> > doesn't work.
>> > Now asterisk sends an initial INVITE with audio media in the SDP. The
>> > CISCO accepts this call after contacting the fax-machine. Then the CISCO
>> > sends a re-INVITE with the T.38 SDP. Asterisk accepts this re-INVITE.
>> > But finally the fax transmission fails and the asterisk verbose trace
>> is:
>> >
>> > *CLI> -- Attempting call on SIP/080913216...@outbound-calls for
>> > 22...@fax-out:1 (Retry 1)
>> > == Using SIP RTP CoS mark 5
>> > == Using UDPTL CoS mark 5
>> >> Channel SIP/outbound-calls-0822aae8 was answered.
>> > == Starting SIP/outbound-calls-0822aae8 at fax-out,2,1 failed so
>> > falling back to exten 's'
>> > -- Executing [...@fax-out:1] Set("SIP/outbound-calls-0822aae8",
>> > "FAXFILE=/root/santi/fax/prueba.tif") in new stack
>> > -- Executing [...@fax-out:2]
>> > SIPDtmfMode("SIP/outbound-calls-0822aae8", "inband") in new stack
>> > -- Executing [...@fax-out:3] SendFAX("SIP/outbound-calls-0822aae8",
>> > "/root/santi/fax/prueba.tif") in new stack
>> > [Mar 10 17:15:28] WARNING[17125]: app_fax.c:176 phase_e_handler: Error
>> > transmitting fax. result=11: Far end cannot receive at the resolution of
>> > the image.
>> > [Mar 10 17:15:28] WARNING[17125]: app_fax.c:621 transmit: Transmission
>> error
>> > == Spawn extension (fax-out, s, 3) exited non-zero on
>> > 'SIP/outbound-calls-0822aae8'
>> >
>> > Any ideas?
>> >
>> > Thanks. Best regards,
>> >
>> > Santi
>> >
>> >
>> >
>> > On Tue, Mar 10, 2009 at 4:26 PM, Joshua Colp > > > wrote:
>> > >
>> > > - "Santiago Gimeno" > > > wrote:
>> > >
>> > > >
>> > > > **The call-file I'm using is:
>> > > >
>> > > > Channel: SIP/08099...@outbound-
>> > > > calls
>> > > > MaxRetries: 3
>> > > > WaitTime: 30
>> > > > Set: LOCALSTATIONID=2
>> > > > Set: LOCALHEADERINFO=T38 fax
>> > > > Set: T38CALL=1
>> > > > Set: T38TXDETECT=yes
>> > > > CallerID: 2
>> > > > Context: fax-out
>> > > > Extension: 2
>> > > > priority:1
>> > > >
>> > >
>> > > And upon further examination... don't put T38CALL in as a variable.
>> > It will cause the initial INVITE to only
>> > > have T38. Leave it out and things should hopefully reinvite.
>> > >
>> > > --
>> > > Joshua Colp
>> > > Digium, Inc. | Software 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--
>> > >
>> > > 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 --
>> >
>> > asterisk-u
Re: [asterisk-users] how to configure for incoming message-summary SUBSCRIBE
My understanding of current SIP MWI handling is: - no matter if an endpoint subscribed to receive message summaries, Asterisk will a summary to it if sip.conf mailbox entry is filled. - I couldn't find any SIP hardphone setting (i used a Thomson ST2030), that would make the hardphone send a SUBSCRIBE without rejecting corresponding NOTIFYs from Asterisk (I confess I didn't spend much time in my trials). bottom line: - I set the phone not to send any SUBSCRIBE as it accepted default NOTIFYs from Asterisk and rejected any custom NOTIFYs I could build. Hope this helps ... ___ -- 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] how to configure for incoming message-summary SUBSCRIBE
Hi! AFAIS the incoming SUBSCRIBE is handled in the same context as INVITE - but how should I handle the SUBSCRIBE in the context? thanks klaus SUBSCRIBE sip:u+431234...@foobar.at:5160 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.82:39982;branch=z9hG4bK-d8754z-3116e1207913aa4e-1---d8754z-;rport Max-Forwards: 70 Contact: To: "schlopy" From: "schlopy";tag=376b6b2e Call-ID: ZDY1MmExZDdlNGE0MGI0NzgxZGQxMjA5YWNmMTZiYzA. CSeq: 2 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1100l stamp 47546 Authorization: Digest username="u+431234567",realm="foobar.at",nonce="fdsfa709",uri="sip:u+431234...@foobar.at:5160",response="foobar",algorithm=MD5 Event: message-summary Content-Length: 0 <-> --- (14 headers 0 lines) --- Creating new subscription Sending to 81.189.99.50 : 39982 (NAT) Found peer 'u+431234567' Looking for u+431234567 in fromSipPots (domain foobar.at) <--- Transmitting (NAT) to 11.111.11.11:39982 ---> SIP/2.0 404 Not Found ___ -- 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] SIP keep-alive with CRLF?
Hi! Ist it possible with Asterisk to send SIP keep-alives with CRLF instead of OPTIONS (qualify)? The OPTIONS are very noisy :-) thanks klaus ___ -- 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] How to do Load-Balancing for Asterisk with OpenSIPS
2009/3/10 Ali Jawad > Great Job Bogdan > > > On Tue, Mar 10, 2009 at 12:52 PM, Bogdan-Andrei Iancu < > bog...@voice-system.ro> wrote: > >> Hi, >> >> When trying to cluster Asterisk boxes to gain scalability and more >> performance, there is now a new simple and efficient solution for doing >> it. >> >> OpenSIPS/OpenSER 1.5 can now implement traffic routing based on load. >> Shortly, when OpenSIPS routes calls to a set of destinations, it is able >> to keep the load status (as number of ongoing calls) of each destination >> and to choose to route to the less loaded destination (at that moment). >> OpenSIPS is aware of the capacity of each destination - it is >> preconfigured with the maximum load accepted by the destinations. >> >> This is an idea Load-Balancer to front your Asterisk cluster. A nice >> tutorial about how to do LB for your Asterisk is available: >> http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing >> >> Regards, >> Bogdan >> >> The best server ever, Great Job! ___ -- 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] VLC
Hi All, When our users receive a voicemail we send it attached to an email. It used to work fine, encoded in wav49 and read by Windows media player. Recently the default player in the company has become VLC which is unable to read wav49. I am trying to use OGG/VORBIS instead of wav49. I can't get it working: In voicemail.conf: format = ogg The result is as follow: [Mar 11 09:42:17] WARNING[24867]: format_ogg_vorbis.c:527 ogg_vorbis_seek: Seeking is not supported on OGG/Vorbis streams! -- x=0, open writing: /var/spool/asterisk/voicemail/default/0213149689/tmp/V2adNF format: ogg, 0x81ec648 -- User hung up [Mar 11 09:42:25] WARNING[24867]: format_ogg_vorbis.c:533 ogg_vorbis_tell: Telling is not supported on OGG/Vorbis streams! [Mar 11 09:42:25] WARNING[24867]: format_ogg_vorbis.c:514 ogg_vorbis_trunc: Truncation is not supported on OGG/Vorbis streams! -- Recording was 0 seconds long but needs to be at least 2 - abandoning Nothing gets recorded in the file. Has anybody done this before? Either get ogg work with voicemail or get VLC to read wav49. Cheers vincent ___ -- 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] How to read installed spandsp version ?
Hello, On my Lenny system, I've got libspandsp.a, libspandsp.la files and so on present in /usr/lib. How could I write a shell script that would read among those files and tell "installed spandsp is version 0.0.4pre12 or version 0.0.6pre3" ? This is something autoconf tools must be able to do but I know how to write any line of code with these tools. While at it, how is Asterisk menuselect checking that app_fax application requisites (ie spandsp is present in appropriate version) are satisfied ? With a bristuff install, I keep getting though I thought installed spandsp correctly (with apt-get install libspandsp1 libspandsp-dev). *** The existing menuselect.makeopts file did not specify that 'app_fax' should not be included. However, either some dependencies for this module were not found or a conflict exists. Either run 'make menuselect' or remove the existing menuselect.makeopts file to resolve this issue. *** Regards ___ -- 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
