Re: [asterisk-users] How long will Asterisk 1.4.x supported/maintained
On Mon, Nov 17, 2008 at 07:46:10PM +0100, Philipp Kempgen wrote: Tilghman Lesher schrieb: On Thursday 13 November 2008 08:16:42 Klaus Darilion wrote: Is there somewhere a statement from Digium how long they will support Asterisk 1.4? There is no statement, because we haven't even discussed when the EOL for 1.4 will be reached. Certainly that means it won't happen for at least the next 60 days, but beyond that, I really don't know. For the average non-techie user who does not want to compile themselves that may sound funny (if not scary). When Debian Lenny (featuring Asterisk 1.4) is finally going to be released that version might not even be supported any more. Debian Lenny was frozen at July, and thus had 1.4.21.2 . Does that indicate Debian (don't really know about other distros) is too slow? Debian freezes Asterisk for 1.5-2 years. Does it mean the development goes too fast? When you install a PBX, do you keep it up-to-date with latest version of Asterisk? OR do you freeze it at some point? Is it a problem with VoIP in general? Does it mean there is no point for a distro to provide VoIP packages because if you want roughly the version everybody else is using you will have to compile it anyway? We're already working on 1.6 packages (they're basically working, but I have to figure out a saner way with the configuration files). One potential way is to use backports. We try to make sure that the Asterisk packages are at always buildable on the Stable platform (through the backport script). This is far from providing QA, but at least it reduces the barrier of participation for others. I don't have good answers here. It's also not clear to me how things will work out with the 1.6.x branches. Those seem to be modeled after the kernel, but that model of the Linux kernel works well because most people use distor kernel (which means that the distros do most of the QA), and those distributions actively participate in the development process and push fixes upstream. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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] test OpenVox B400P and junghans card for dahdi BRI wcb4xxp
On Tue, Nov 18, 2008 at 02:23:56PM +0800, lizhong zhu wrote: hello, all of users: after dig the code, i found that dahdi wcb4xxp is only for digium B410P. it has VPM hardware based echo cancellation, which Junghans and openvox bri cards do not have. anyone can tell me how to disable the ec_write methond to support other HFC BRI cards? regards! zhu My basic work in progress is here: http://bugs.digium.com/view.php?id=13897 Please submit your patches. Please also use latest svn (or 2.1.0-rc4) as it seems to include a number of other fixes (in the D-channel handling) BTW: keeping to one thread can help others follow this. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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] HPEC performance
On Mon, Nov 17, 2008 at 07:11:33PM -0700, Joseph L. Casale wrote: Does this make a significant improvement? The box in question I was going to try this with has a 4 port TDM card w/ plenty of horsepower, but I do intend to later migrate to a Soekris unit running Astlinux and therefore might not have the power to run it after. If the difference is significant, I may move to an ITX board so I could use a bigger CPU, but only if the hassle is worthwhile. The first thing you should try is reduce the number of taps. But slightly off-topic: If you want to use OSLEC instead, I lately managed to squeese better performance from it using the MMX optimizations: http://bugs.digium.com/view.php?id=13500 http://sourceforge.net/mailarchive/forum.php?thread_name=20081113215314.GM31838%40xorcom.comforum_name=freetel-oslec -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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
[asterisk-users] Asterisk Realtime and device contexts
Hello all, I need some help/infomation/correction regarding storing the SIP peers in a database table via ARA. I've alreday created the SQLite3 database table and the related ODBC plumbing and have Asterisk (1.4) validating the SIP devices from a table and updating it correctly with dynamic infomation, for example the device's IP address. Problem is: I have a scenario whereby I change the default context of the devices from time to time. But when the devices place a call, Asterisk does not place them in the new context that I've set in the database table. This is also the case when I force the devices to SIP re-register. sip show peer DEVICE shows that it is still in the old context. sip prune realtime DEVICE does not help Does anybody have any idea why this is happening? Does Asterisk ignore subsequent updates to the context field? Caching? Can caching be truned off/patched to be off by default ? Thank you. Sawan ___ -- 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] Caching Asterisk SIP useragent info?
Apologies that I did not make myself clear the first time, I meant the process of Asterisk saving useragent data for its users. Is it configurable via asterisk or is it just the re-register settings on the SNOM phone? Thanks again Paul. Veselin K On Tue, Nov 18, 2008 at 10:21:14AM +1100, Paul Hales wrote: The process for upgrading would greatly depend on how Asterisk was installed in the first place. If Asterisk was installed from source, then a fresh download of source followed by the usual configure/make/etc commands would do the trick. PaulH Veselin K wrote: Hello Paul, thanks for the reply. Could you please tell me what is the process called so I can research it further. Thank you. Veselin K On Mon, Nov 17, 2008 at 10:47:47AM +1100, Paul Hales wrote: This process has been greatly improved in the latest versions of Asterisk - might be time to upgrade. PaulH [EMAIL PROTECTED] wrote: Hello, I'm running an Asterisk 1.4.14 on a linux machine. Serving SIP Snom users. I've noticed that each time Asterisk is restarted, for the first 5-10 minutes, the SIP users can dial but cannot be dialed until each phone re-registers itself against the server. So only after the Saved useragent...for peer 111 line appears on the Asterisk console, then the 111 user can be reached. What exactly is this process? Is it that the phones send their extension/password details to the server at specific intervals or does the server send a broadcast message, looking for phones? Is there any way to cache/save this SIP useragent information so in case the server is restarted, the user need not wait for their phone to re-register? Also I believe that it is sufficient for the user to just pickup their handset in order to force their phone to re-register quicker. However I'd like to avoid asking the users to do that. Thank you much. Veselin K ___ -- 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 ___ -- 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 ___ -- 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] HPEC performance
On Mon, 17 Nov 2008, Joseph L. Casale wrote: Does this make a significant improvement? The box in question I was going to try this with has a 4 port TDM card w/ plenty of horsepower, but I do intend to later migrate to a Soekris unit running Astlinux and therefore might not have the power to run it after. If the difference is significant, I may move to an ITX board so I could use a bigger CPU, but only if the hassle is worthwhile. HPEC did make an improvement in some of my cases, but I found the Digium licensing procedure to be a hassle, so switched to OSLEC. What you might want to do it try OSLEC, and if that performs OK for you, then you might consider buying the HPEC licenses - I imagine their code/cpu usage might be the same, but then if you try OSLEC and it works for you, you might not bother with HPEC... 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
[asterisk-users] busy-level / busy-limit Asterisk 1.4.22
Hi to all the busy-level / busy-limit setting in sip.conf is available for Asterisk 1.4.22 ? This is a piece of my sip.conf: [202] type=friend secret=202 host=dynamic; This device registers with us username=202; Username to use when calling this device before registration limitonpeers = yes call-limit = 2 busy-level = 1 The directive busy-level is ignored I've also tried busy-limit but without any result... Thanks -- /*/ nik600 http://www.kumbe.it ___ -- 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] busy-level / busy-limit Asterisk 1.4.22
On 18 Nov 2008, at 10:30, nik600 wrote: the busy-level / busy-limit setting in sip.conf is available for Asterisk 1.4.22 ? http://www.voip-info.org/wiki/view/Asterisk+sip+busy-level ___ -- 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] Use the NEW ulaw/alaw codecs (slower, but cleaner)
Steve Underwood [EMAIL PROTECTED] writes: In spandsp I do the G.711 conversions algorithmically. Most modern processors have a where is the top 1 instruction, and that reduces the calculations to something very fast. Very nice! I'd like to see the code, but I'm too lazy to go look through all of spandsp... Can you tell me which file I should look in? /Benny ___ -- 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] busy-level / busy-limit Asterisk 1.4.22
Thanks, is it possibile to retrieve a patch from Asterisk trunk? how? On Tue, Nov 18, 2008 at 11:54 AM, Steve Howes [EMAIL PROTECTED] wrote: On 18 Nov 2008, at 10:30, nik600 wrote: the busy-level / busy-limit setting in sip.conf is available for Asterisk 1.4.22 ? http://www.voip-info.org/wiki/view/Asterisk+sip+busy-level ___ -- 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 -- /*/ nik600 http://www.kumbe.it ___ -- 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] FOP with Asterisk 1.6. No call Information.
Hi All, For some reason the Asterisk Flash Operator Panel is not working since moving to Asterisk 1.6 from 1.4. I did a complete install onto new hardware. FOP will show an extension and trunk offline when it is offline. It will also show a call in progress to MeetMe but it will not show any details about calls between extensions or calls over a trunk. Please Help. ___ -- 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] Use the NEW ulaw/alaw codecs (slower, but cleaner)
In article [EMAIL PROTECTED], Benny Amorsen [EMAIL PROTECTED] wrote: Steve Underwood [EMAIL PROTECTED] writes: In spandsp I do the G.711 conversions algorithmically. Most modern processors have a where is the top 1 instruction, and that reduces the calculations to something very fast. Very nice! I'd like to see the code, but I'm too lazy to go look through all of spandsp... Can you tell me which file I should look in? g711.c - but you've probably discovered that by now... -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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
Re: [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)
Benny Amorsen wrote: Steve Underwood [EMAIL PROTECTED] writes: In spandsp I do the G.711 conversions algorithmically. Most modern processors have a where is the top 1 instruction, and that reduces the calculations to something very fast. Very nice! I'd like to see the code, but I'm too lazy to go look through all of spandsp... Can you tell me which file I should look in? If you want to find the G.711 code, g711.c might be a good place to look. 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] Asterisk with or without OpenSER
Hello, I am running a small installation of asterisk and looking for future expansion of it to handle thousands of users. From what I read I see that usually large installation place OpenSER (or similar solution) in front of Asterisk in order to provide high call rate because OpenSER does only signalling while Asterisk does all. My question is: If Asterisk also does only signalling (i.e. the voice traffic goes directly between the phones and not via asterisk) is it still that slow? I preffer to have one software package rather than dealing with two. Thanks! __Yehavi: ___ -- 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] Asterisk not reading fast DTMFs, was: PBX - PRI - * - Telco not working
On Mon, Nov 17, 2008 at 10:20 AM, Tony Mountifield [EMAIL PROTECTED] wrote: If I do this from an NEC digital extension I get 14149692, but if I do it from an NEC POTS extension I get 1942124000 That looks like when you pick up the analogue phone and dial 9, it immediately opens the outgoing line and sends the 141 acces code, but is doing so at the same time you carry on dialling 692. So the digits clash with each other. Notice you have 1414 interleaved with 922000. It appears like the digits generated by the NEC (1414) are overriding the digits coming in from the phone, and either obliterating the latter, or splitting them up (in the case of the 2, which gets chopped in half by a short burst of 1). OK, I removed the 1414 prefix from the NEC system. And now I have found a basic problem. If I connect a POTS phone to the analogue extensions and dial fast (like an autodial) asterisk doesn't read the digits properly. If I connect manually and dial slowly, asterisk reads all the digits correctly and can handle the call. Is there any way that i can get asterisk to read the faster DTMF digits? Mikel -- http://lindsaar.net/ Rails, RSpec and Life blog ___ -- 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] Picked up calls die in exactly 20 seconds
On Mon, Nov 17, 2008 at 6:04 PM, Juan Carlos Castro y Castro [EMAIL PROTECTED] wrote: Weird thing happening when a call is picked up. Whether by *8 feature, or by directed pickup via dialplan, either with Pickup() or with Pickup2(), the same thing happens: the call is picked up successfully, and after exactly 20 seconds talking, the call is terminated. The originating end gets a hangup, while the side that did the pickup goes mute. Anyone experienced anything similar? Throw an answer() in after pickup() and see if it still does the same. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Asterisk not reading fast DTMFs, was: PBX - PRI - * - Telco not working
If I connect a POTS phone to the analogue extensions and dial fast (like an autodial) asterisk doesn't read the digits properly. If I connect manually and dial slowly, asterisk reads all the digits correctly and can handle the call. Is there any way that i can get asterisk to read the faster DTMF digits For example. On the POTS phone I dial: 95523025 And the following comes up in the caller log: == CDR updated on DAHDI/21-1 -- Executing [EMAIL PROTECTED]:1] Dial(DAHDI/21-1, DAHDI/g2/29350525,,Tr) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g2/29350525 -- DAHDI/38-1 is proceeding passing it to DAHDI/21-1 -- Channel 0/7, span 2 got hangup request, cause 1 -- Hungup 'DAHDI/38-1' So it gets all the right digits... just interleaved. 2 9 3 5 0 5 25 955 2 3 025 Any ideas? As I said before, if i manually dial the digits with 1 second lags between each button press, it calls out fine. Mikel -- http://lindsaar.net/ Rails, RSpec and Life blog ___ -- 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 command from 1.4 to 1.6
Say in 1.4 there was a meetme command that would show active meetme conferences. What is that same command in 1.6? I looked at core show help - didnt see it. I looked at UPGRADE.txt and didnt see it. Thanks, Jerry ___ -- 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] Asterisk with or without OpenSER
Look into FreeSwitch. http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ On Tue, Nov 18, 2008 at 7:29 AM, Yehavi Bourvine [EMAIL PROTECTED] wrote: Hello, I am running a small installation of asterisk and looking for future expansion of it to handle thousands of users. From what I read I see that usually large installation place OpenSER (or similar solution) in front of Asterisk in order to provide high call rate because OpenSER does only signalling while Asterisk does all. My question is: If Asterisk also does only signalling (i.e. the voice traffic goes directly between the phones and not via asterisk) is it still that slow? I preffer to have one software package rather than dealing with two. Thanks! __Yehavi: ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] FOP with Asterisk 1.6. No call Information.
David Klaverstyn wrote: Hi All, For some reason the Asterisk Flash Operator Panel is not working since moving to Asterisk 1.6 from 1.4. I did a complete install onto new hardware. FOP will show an extension and trunk offline when it is offline. It will also show a call in progress to MeetMe but it will not show any details about calls between extensions or calls over a trunk. This is a known issue. I would suggest you subscribe to the FOP mailing list, Nicolas is offering a beta version for 1.6 and is asking for feedback on it. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Asterisk 1.4.21.2 and gtalk2voip
Hi, Ii try to connect an Asterisk server running 1.4.21.2 version with gtalk2voip services. Everything is fine till the call for DTMF test: there is no audio and Asterisk shows [Nov 18 14:51:47] WARNING[20502]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 1 (Critical Response) [Nov 18 14:51:47] WARNING[20502]: chan_sip.c:1972 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. == Spawn extension (ServiceNumbers, 104, 7) exited non-zero on 'SIP/TEST-LEG-08306f78' Does anyone have successfull connected to this service with recent Asterisk version? Regards -- Daniel ___ -- 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] Crash when rebooting or unload xorcom modules
Hi, my system crashes when trying to reboot or when I unload the xpp kernel module. For the reboot its quite annoying because I can no longer reboot the server remotely as it crashes and wont come back. When I try to unload the module I have of course stopped asterisk and unloaded depending modules before. I tried with the bristuff of junghanns: http://www.junghanns.net/downloads/bristuff-0.4.0-RC3b.tar.gz http://www.junghanns.net/downloads/bristuff-0.4.0-RC3c.tar.gz http://www.junghanns.net/downloads/bristuff-0.4.0-RC3d.tar.gz I am running ubuntu gutsy with kernel 2.6.22-14-server. Has someone else those problems? Here is some more information from syslog. Nov 18 15:20:01 MIXpbx kernel: [ 600.866873] WARNING: at /build/buildd/linux-source-2.6.22-2.6.22/lib/kref.c:33 kref_get() Nov 18 15:20:01 MIXpbx kernel: [ 600.866887] [kref_get+61/64] kref_get +0x3d/0x40 Nov 18 15:20:01 MIXpbx kernel: [ 600.866901] [kobject_get+15/32] kobject_get+0xf/0x20 Nov 18 15:20:01 MIXpbx kernel: [ 600.866907] [get_device+14/32] get_device+0xe/0x20 Nov 18 15:20:01 MIXpbx kernel: [ 600.866914] [driver_detach+45/208] driver_detach+0x2d/0xd0 Nov 18 15:20:01 MIXpbx kernel: [ 600.866922] [bus_remove_driver +103/144] bus_remove_driver+0x67/0x90 Nov 18 15:20:01 MIXpbx kernel: [ 600.866927] [f8a849da] unregister_xpp_bus+0xa/0x20 [xpp] Nov 18 15:20:01 MIXpbx kernel: [ 600.866938] [f8a8d4da] xbus_core_shutdown+0x2a/0x40 [xpp] Nov 18 15:20:01 MIXpbx kernel: [ 600.866949] [f8a8d4fa] xpp_zap_cleanup+0xa/0xf [xpp] Nov 18 15:20:01 MIXpbx kernel: [ 600.866958] [sys_delete_module +298/400] sys_delete_module+0x12a/0x190 Nov 18 15:20:01 MIXpbx kernel: [ 600.866964] [remove_vma+57/80] remove_vma+0x39/0x50 Nov 18 15:20:01 MIXpbx kernel: [ 600.866973] [sysenter_past_esp +107/161] sysenter_past_esp+0x6b/0xa1 Nov 18 15:20:01 MIXpbx kernel: [ 600.866979] [svc_disconnect+80/304] svc_disconnect+0x50/0x130 Nov 18 15:20:01 MIXpbx kernel: [ 600.866985] === I have in the same system a junghanns bri card. Could that be a problem? Best regards, Loic. ___ -- 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 Barge specific extensions
Hi amit not you if you can create a group of extensions to spy, but for
example if your extensions are of 3 digits your you can create something
like that...
exten = _*5XXX,1,ChanSpy(SIP/${EXTEN:2},bq)
it configures the hint inside the dialplan to be able to see the state of
the extensions
best regards
2008/11/17 amit salunkhe [EMAIL PROTECTED]
Hi All
Can anybody help me for dial plan to barge or Spy(ExtenSpy)
specificor selective extemsions among 20 extension in my office.
lets say my office extension range is 301-320 i want to barge only 3
extension say 320, 302,314.
is this possible to barge specific extension? . Plz help me for this.I
am using Asterisk 1.4.9 SIP channels.
Regards
Amit
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Re: [asterisk-users] Asterisk not reading fast DTMFs, was: PBX - PRI - * - Telco not working
In article [EMAIL PROTECTED], Mikel Lindsaar [EMAIL PROTECTED] wrote: If I connect a POTS phone to the analogue extensions and dial fast (like an autodial) asterisk doesn't read the digits properly. If I connect manually and dial slowly, asterisk reads all the digits correctly and can handle the call. Is there any way that i can get asterisk to read the faster DTMF digits For example. On the POTS phone I dial: 95523025 And the following comes up in the caller log: == CDR updated on DAHDI/21-1 -- Executing [EMAIL PROTECTED]:1] Dial(DAHDI/21-1, DAHDI/g2/29350525,,Tr) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g2/29350525 -- DAHDI/38-1 is proceeding passing it to DAHDI/21-1 -- Channel 0/7, span 2 got hangup request, cause 1 -- Hungup 'DAHDI/38-1' So it gets all the right digits... just interleaved. 2 9 3 5 0 5 25 955 2 3 025 Any ideas? As I said before, if i manually dial the digits with 1 second lags between each button press, it calls out fine. Well that IS weird! It looks to me like the NEC is collecting up some digits itself (e.g. that it receives before it gets Answer status from Asterisk), and then sending on the collected digits once it has connected, but these are then overlapping with the rest of the digits that are being passed through from the phone in-band. I think the source of your problems now is the behaviour of the NEC unit. So you need to understand exactly what it does with DTMF and how it wants to interact with the Asterisk unit behind it. I don't think Asterisk is the problem any more... Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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] Configuring Sangoma BRI with zaptel?
Hello, there has been a post to this list somewhere arount april which said that it is possible to use a Sangoma BRI A500 card with zaptel and asterisk bristuff. That is, without sangoma_brid and sangoma_mgd daemons and without woomera channels. Could anybody give me a short hint how to configure this? I tried wanpipe-driver + zaptel + asterisk-bristuffed, but I couldn't get zaptel to recognize the sangoma channels. modprobe wanpipe did load zaptel module and others but no spans appeared in /proc/zaptel or /etc/zaptel.conf. I tried various config options of the wanpipe setup tool, but to no avail. genzaptelconf -d displays correct cardinfo but doesn't seem to get the channels. Config: debian etch with kernel 2.6.18-5-amd64 on x86_64 sangoma a503de (PCIe 6x BRI w/ Echo Cancel) asterisk 1.4.13-BRIstuffed-0.4.0-test4 (from pkg-voip.buildserver.net) zaptel 1.4.7 (from pkg-voip) wanpipe 3.3.14 (newest beta) Same config (without wanpipe of course) works well with a digium TE220 (PCIe 2x PRI). Any hints would be greatly appreciated as I'm banging my head about this for some days now ;-) Claus -- CHECON EDV-Consulting und Redaktion Claus Herwig * Barer Straße 70 * 80799 München +49 89 27826981 * Fax 27826982 * [EMAIL PROTECTED] ___ -- 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] Asterisk not reading fast DTMFs, was: PBX - PRI - * - Telco not working
On Wed, Nov 19, 2008 at 2:13 AM, Tony Mountifield [EMAIL PROTECTED]wrote: In article [EMAIL PROTECTED], Mikel Lindsaar [EMAIL PROTECTED] wrote: For example. On the POTS phone I dial: 95523025 And the following comes up in the caller log: == CDR updated on DAHDI/21-1 -- Executing [EMAIL PROTECTED]:1] Dial(DAHDI/21-1, DAHDI/g2/29350525,,Tr) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g2/29350525 -- DAHDI/38-1 is proceeding passing it to DAHDI/21-1 -- Channel 0/7, span 2 got hangup request, cause 1 -- Hungup 'DAHDI/38-1' So it gets all the right digits... just interleaved. As I said before, if i manually dial the digits with 1 second lags between each button press, it calls out fine. Well that IS weird! It looks to me like the NEC is collecting up some digits itself (e.g. that it receives before it gets Answer status from Asterisk), and then sending on the collected digits once it has connected, but these are then overlapping with the rest of the digits that are being passed through from the phone in-band. I think the source of your problems now is the behaviour of the NEC unit. So you need to understand exactly what it does with DTMF and how it wants to interact with the Asterisk unit behind it. I don't think Asterisk is the problem any more... Which I would agree with 100% if it were not for the fact that this same NEC system was working without ANY modification on and E1 the day before. The setup was: NEC == E1 == Telco To which I changed it to: NEC == CAT5 == TE210P:1 = * = TE210P:2 == E1 == Telco ie... just inserted the Asterisk box in between. I plug the NEC back straight to the Telco and all works well again. Unless Asterisk is expecting inband DTMF and the NEC was doing out of band with the Telco :/ That would make sense.. but how to force it to out of band? Mikel -- http://lindsaar.net/ Rails, RSpec and Life blog ___ -- 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] Incoming Transfer
I have incoming analog and SIP DIDs that all ring multiple sip extensions with a Dial command as the first exten. I am curious to know if it's possible for the incoming caller to transfer out of the Dial command while in progress and dial a single extension? Thanks! jlc ___ -- 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] FOP with Asterisk 1.6. No call Information.
FOP is not compatible with Asterisk 1.6, you should look into the FOP list as the author is looking for people to try a new version to make it work. On Tue, 2008-11-18 at 21:14 +1000, David Klaverstyn wrote: Hi All, For some reason the Asterisk Flash Operator Panel is not working since moving to Asterisk 1.6 from 1.4. I did a complete install onto new hardware. FOP will show an extension and trunk offline when it is offline. It will also show a call in progress to MeetMe but it will not show any details about calls between extensions or calls over a trunk. Please Help. ___ -- 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 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- 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] meetme command from 1.4 to 1.6
On Tue, Nov 18, 2008 at 08:08:51AM -0500, Jerry Geis wrote: Say in 1.4 there was a meetme command that would show active meetme conferences. What is that same command in 1.6? I looked at core show help - didnt see it. I looked at UPGRADE.txt and didnt see it. The 1.4 command was simply meetme. If you try running meetme on 1.6, you will see the help command core show help meetme. Listed there is the ability to list all conferences using meetme list. Jeff ___ -- 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] sound quality between two back-to-back asterisk
Hi, I have two asterisks that are connected to each other via a back-to-back E1 link using a pair of sangoma cards. With the following scenario: SIP-PHONE - Asterisk - E1 - Asterisk - SIP-PHONE, the sound quality degrades significantly. I can't understand why as the amound of packet lost should be very minimum. Does anyone know why? Does it have anything to do with what codec to use? Thanks, Mark ___ -- 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] diax debian package
I can not seem to find DIAX Debian package, I know it exists; but I can not find it via Google. I know Diax is for Windows only but someone created package for debian and I can not seem to find it. -- #Joseph GPG KeyID: ED0E1FB7 ___ -- 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] Asterisk 1.4.21.2 and gtalk2voip
Administrator TOOTAI a écrit : Hi, Ii try to connect an Asterisk server running 1.4.21.2 version with gtalk2voip services. Everything is fine till the call for DTMF test: there is no audio and Asterisk shows [Nov 18 14:51:47] WARNING[20502]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 1 (Critical Response) [Nov 18 14:51:47] WARNING[20502]: chan_sip.c:1972 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. == Spawn extension (ServiceNumbers, 104, 7) exited non-zero on 'SIP/TEST-LEG-08306f78' Reply to myself: had to put nat=yes as default in sip.conf and now it's OK -- Daniel ___ -- 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] diax debian package
On Tue, Nov 18, 2008 at 10:33:44AM -0700, Joseph wrote: I can not seem to find DIAX Debian package, I know it exists; but I can not find it via Google. I know Diax is for Windows only but someone created package for debian and I can not seem to find it. DIAX? http://www.laser.com/dante/diax/diax.html This is a full featured and very small IAX based software phone for Microsoft Windows platforms (only). Kiax maybe? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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
[asterisk-users] Do Digium Digital Cards Handle Remote Loopback Command?
If the telco sends a remote loopback command on a T1/PRI circuit, will the Digium card act appropriately or do we need to put a CSU on the line to handle such tests? -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ -- 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] Do Digium Digital Cards Handle Remote Loopback Command?
George Pajari wrote: If the telco sends a remote loopback command on a T1/PRI circuit, will the Digium card act appropriately or do we need to put a CSU on the line to handle such tests? As far as I know it should loop up. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] setting up callback
Greetings Asterisk users!
I'm trying to setup Asterisk system to act as a callback system together
with callcentric (http://callcentric.com) but it appears that I hit common
DTMF issue and I want to workaround this problem. Basically my current
setup is the following:
1) I have dedicated Asterisk server that it is linked to my callcentric
account
2) I have US phone number (DID) from callcentric attached to my account
3) I want to make calls from my cell phone to (real US) callcentric number
and receive a callback to my cell phone number. After receiving callback,
I enter 4-digit password to auth myself and then I get a line via DISA
feature of Asterisk.
I guess my setup is very common, and all is great (e.g. I'm able to
receive a callback, enter password and then get callcentric line), except
that callcentric does not appear to be getting DTMF tones from my cell
phone correctly and I am unable to make a call.
I have searched all day long today and all I was able to find is that some
people have callback DTMF working with callcentric fine and others not. I
tweaked my sip.conf with all possible combinations of dtmfmode setting,
but still no luck.
Maybe I want something strange, but it appears that in my case Asterisk is
able to read DTMF tones correctly while making callback and asking me to
enter password to authenticate myself (I am able to pass authentication
process with no problems), so what I want to do is to use that instead of
using DISA feature of Asterisk. In other words I want something like this:
- I call callcentric from my cell
- Asterisk calls me back using callcentric line
- I enter 4-digit password to authenticate first
- if authentication went through, I type a phone number I wish to call
- Asterisk initiates a SIP call to provided phone number through
callcentric, and all this has to work so that I can speak and hear remote
party on my cell phone.
I hope the above scheme is clear enough to understand.
The problem is that I cannot understand how to implement the above -
should this be done with WaitExten() feature? If so, can someone share
examples of their setup? I would appreciate any pointers to implement the
above.
My current GSM provider in Russia is Megafon, and I believe this has
something to do with them that DTMF tones don't get passed correctly.
Here's my current parts of config files responsible for callback:
sip.conf:
register = 1777286:[EMAIL PROTECTED]/1862772
...
[callcentric]
type=peer
context=from-callcentric
host=callcentric.com
username=1777286
secret=XXX
fromuser=1777286
fromdomain=callcentric.com
disallow=all
allow=alaw
dtmfmode=inband
canreinvite=no
;rfc2833compensate=yes
insecure=very
extensions.conf:
NOTE: 1862772 is a real phone # I have in my callcentric account
[from-callcentric]
exten = 1862772,1,NoOp(callcentric callback to ${CALLERID(num))
exten = 1862772,2,Wait(1)
exten = 1862772,3,system(cp /var/spool/asterisk/skelett.call
/var/spool/asterisk/skelett.tmp.call)
exten = 1862772,4,system(echo 'Channel:
SIP/+${CALLERID(num)[EMAIL PROTECTED]'
/var/spool/asterisk/skelett.tmp.call)
exten = 1862772,5,system(mv /var/spool/asterisk/skelett.tmp.call
/var/spool/asterisk/outgoing)
exten = 1862772,6,HangUp
[callback-dialtone-auth]
exten = s,1,answer()
exten = s,n,authenticate(5678)
exten = s,n,DISA(no-password,home)
/var/spool/asterisk/skelett.call:
Context: callback-dialtone-auth
Extension: s
MaxRetries: 2
RetryTime: 1
Thank you,
Mikhail.
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Re: [asterisk-users] Asterisk not reading fast DTMFs, was: PBX - PRI - * - Telco not working
I plug the NEC back straight to the Telco and all works well again. I just got on the phone to Digium and we've raised a ticket with some pri intense debugging going on. I'll update the list on findings. Mikel -- http://lindsaar.net/ Rails, RSpec and Life blog ___ -- 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] Asterisk with or without OpenSER
On Nov 18, 2008, at 4:29 AM, Yehavi Bourvine wrote: Hello, I am running a small installation of asterisk and looking for future expansion of it to handle thousands of users. From what I read I see that usually large installation place OpenSER (or similar solution) in front of Asterisk in order to provide high call rate because OpenSER does only signalling while Asterisk does all. My question is: If Asterisk also does only signalling (i.e. the voice traffic goes directly between the phones and not via asterisk) is it still that slow? I preffer to have one software package rather than dealing with two. Thanks! __Yehavi: Asterisk is capable of handling both thousands of users and large call volumes easily. I have done both, without SER or other front-end software. The specifics of your installation of course may push you towards other solutions that may involve a pure SIP proxy, but without more data I would suggest that Asterisk is usually sufficient with no more thought for design than other solutions would require. I would suggest creating a test environment that mimics your expected higher load situations and testing for yourself, and believe little of what others claim, including myself. Create an array of Asterisk machines that simulate user populations, and point them at your testing system. This will allow you to determine, given your specific environment and requirements, if various solutions can be used successfully. My bet is that you'll find that Asterisk as your single package will be sufficient. Hints: Will SIP redirection work for you? How about re-INVITEs? Does Linux HA give any solutions? L3 load sharing? JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- 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] sound quality between two back-to-back asterisk
On Tue, Nov 18, 2008 at 11:00 PM, mark morreny [EMAIL PROTECTED]wrote: Hi, I have two asterisks that are connected to each other via a back-to-back E1 link using a pair of sangoma cards. With the following scenario: SIP-PHONE - Asterisk - E1 - Asterisk - SIP-PHONE, the sound quality degrades significantly. I can't understand why as the amound of packet lost should be very minimum. Does anyone know why? Does it have anything to do with what codec to use? Thanks, Mark what is the sip phone that you are using ? is it a IP phone instrument or a softphone ? Try running with ulaw or alaw (g711) . Cause we found that certain softphones with gsm or other codecs like speex can produce really bad audio. Thanks Regards, Godson Gera http://godson.in ___ -- 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] Asterisk with or without OpenSER
Thanks! 2008/11/18 John Todd [EMAIL PROTECTED] Hints: Will SIP redirection work for you? How about re-INVITEs? Does Linux HA give any solutions? L3 load sharing? 1. What do you mean by SP redirection in this context? 2. I am using re-invites, and made sure that the audio does not pass via the asterisk server. 3. L3 load sharing: I did some work on this and this is the method I came with: - I use MySQL with replication between two servers. - Some of the phones register to one server and the others to the other (i.e., this way I load balance). - Since the MySQL database is replicated, each server knows where the destination is registered. If it is registered on the other sevrer then the caller is redirected there. 4. HA? didn't give too much thought to it yet, but it is on my list. BTW, how do you emulate large number of clients without buying some test equipment like IXIA? Thanks! __Yehavi: ___ -- 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] Use the NEW ulaw/alaw codecs (slower, but cleaner)
Interestingly the Linux kernel has this for find-last-set:
static inline unsigned long __fls(unsigned long word)
{
asm(bsr %1,%0
:=r (word)
:rm (word));
return word;
}
spandsp has this (Everything non-x86 has been removed):
static __inline__ int top_bit(unsigned int bits)
{
int res;
__asm__ ( xorl %[res],%[res];\n
decl %[res];\n
bsrl %[bits],%[res]\n
: [res] =r (res)
: [bits] rm (bits));
return res;
}
I haven't measured which one is best.
/Benny
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Re: [asterisk-users] Incoming Transfer
On Tue, Nov 18, 2008 at 9:25 PM, Joseph L. Casale [EMAIL PROTECTED] wrote: I have incoming analog and SIP DIDs that all ring multiple sip extensions with a Dial command as the first exten. I am curious to know if it's possible for the incoming caller to transfer out of the Dial command while in progress and dial a single extension? Yes it is possible , use 'd' option in Dial command to do that. Then caller can press a single digit and it will go to that extension in the current dialplan context or goes to single digits extension in context specified by EXITCONTEXT channel variable. exten = s,1,Dial(SIP/123,,d) Thanks Regards, Godson Gera. http://godson.in ___ -- 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] Realtime MOH
Hi, I'm having troubles using music on hold with realtime (ast 1.6.0.1). Everything seems ok, but no clases are show. If I try to make a cal nothing is played and says theres no moh class. Musiconhold.conf [general] cachertclasses=yes ; use 1 instance of moh class for all users who are using it, ; decrease consumable cpu cycles and memory ; disabled by default [default] mode=files directory=/var/lib/asterisk/moh extconfig.conf musiconhold = mysql,database if I do a Realtime load from CLI to see if is accesing correctly to the family it seems ok, is returning my parameters on my table. Any idea?? 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
Re: [asterisk-users] Do Digium Digital Cards Handle Remote Loopback Command?
Alex Balashov wrote: George Pajari wrote: If the telco sends a remote loopback command on a T1/PRI circuit, will the Digium card act appropriately or do we need to put a CSU on the line to handle such tests? As far as I know it should loop up. It will, although the card does not handle it alone, the driver must be loaded and operational to respond to the remote loop up/down requests. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] Fwd: Polycom phone time behind one hour.
Have you verified that the NTP server has the correct time? Also, if you're grabbing the time from a source set to GMT you'd need to set the gmtOffset field. Doug Smith wrote: Tried to submit this email this morning and didn't see it in the list. I apologize if it is a dupe. I've inherited a customized Asterisk installation. After the past time change all clocks in my office are behind by one hour. After some digging it appears we have: A /tftproot/sip.conf that is being pushed out to our phones. I found the following line that seems to be what controls timezone information and DST. I put in carriage returns to make it easier to read as it is all one line. Can anyone see anything obvious (I have missed after reviewing many times) with this config that would cause my phones to be behind an hour? I tried changing overrideDHCP=0 to a 1 with no luck. |SNTP tcpIpApp.sntp.resyncPeriod=3600 tcpIpApp.sntp.address=207.207.*.* (Address replaced with asterisk to protect our server IP) tcpIpApp.sntp.address.overrideDHCP=0 tcpIpApp.sntp.gmtOffset= tcpIpApp.sntp.gmtOffset.overrideDHCP=0 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=9 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=4 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0/ | Any help at resolving this would be greatly appreciated. Many of our office workers are annoyed that their times are behind an hour now. Thanks, Doug Smith Alchemy Systems ___ -- 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] Use the NEW ulaw/alaw codecs (slower, but cleaner)
Benny Amorsen wrote:
Interestingly the Linux kernel has this for find-last-set:
static inline unsigned long __fls(unsigned long word)
{
asm(bsr %1,%0
:=r (word)
:rm (word));
return word;
}
spandsp has this (Everything non-x86 has been removed):
static __inline__ int top_bit(unsigned int bits)
{
int res;
__asm__ ( xorl %[res],%[res];\n
decl %[res];\n
bsrl %[bits],%[res]\n
: [res] =r (res)
: [bits] rm (bits));
return res;
}
I haven't measured which one is best.
Measurement is the wrong issue to look at. The Digium one is just wrong.
The bsr or bsrl does not set the result register if the source is zero.
It indicated the zero condition through a flag. Its faster to avoid
testing that flag, and just preload the result register with a well
known value.
Steve
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Re: [asterisk-users] Caching Asterisk SIP useragent info?
if you upgrade Asterisk to one of the latest versions, this issue will be fixed - the SIP entries are NOT cleared on a reload. PaulH Veselin K wrote: Apologies that I did not make myself clear the first time, I meant the process of Asterisk saving useragent data for its users. Is it configurable via asterisk or is it just the re-register settings on the SNOM phone? Thanks again Paul. Veselin K On Tue, Nov 18, 2008 at 10:21:14AM +1100, Paul Hales wrote: The process for upgrading would greatly depend on how Asterisk was installed in the first place. If Asterisk was installed from source, then a fresh download of source followed by the usual configure/make/etc commands would do the trick. PaulH Veselin K wrote: Hello Paul, thanks for the reply. Could you please tell me what is the process called so I can research it further. Thank you. Veselin K On Mon, Nov 17, 2008 at 10:47:47AM +1100, Paul Hales wrote: This process has been greatly improved in the latest versions of Asterisk - might be time to upgrade. PaulH [EMAIL PROTECTED] wrote: Hello, I'm running an Asterisk 1.4.14 on a linux machine. Serving SIP Snom users. I've noticed that each time Asterisk is restarted, for the first 5-10 minutes, the SIP users can dial but cannot be dialed until each phone re-registers itself against the server. So only after the Saved useragent...for peer 111 line appears on the Asterisk console, then the 111 user can be reached. What exactly is this process? Is it that the phones send their extension/password details to the server at specific intervals or does the server send a broadcast message, looking for phones? Is there any way to cache/save this SIP useragent information so in case the server is restarted, the user need not wait for their phone to re-register? Also I believe that it is sufficient for the user to just pickup their handset in order to force their phone to re-register quicker. However I'd like to avoid asking the users to do that. Thank you much. Veselin K ___ -- 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 ___ -- 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 ___ -- 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
[asterisk-users] Fwd: Polycom phone time behind one hour.
Tried to submit this email this morning and didn't see it in the list. I apologize if it is a dupe. I've inherited a customized Asterisk installation. After the past time change all clocks in my office are behind by one hour. After some digging it appears we have: A /tftproot/sip.conf that is being pushed out to our phones. I found the following line that seems to be what controls timezone information and DST. I put in carriage returns to make it easier to read as it is all one line. Can anyone see anything obvious (I have missed after reviewing many times) with this config that would cause my phones to be behind an hour? I tried changing overrideDHCP=0 to a 1 with no luck. SNTP tcpIpApp.sntp.resyncPeriod=3600 tcpIpApp.sntp.address=207.207.*.* (Address replaced with asterisk to protect our server IP) tcpIpApp.sntp.address.overrideDHCP=0 tcpIpApp.sntp.gmtOffset= tcpIpApp.sntp.gmtOffset.overrideDHCP=0 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=9 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=4 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0/ Any help at resolving this would be greatly appreciated. Many of our office workers are annoyed that their times are behind an hour now. Thanks, Doug Smith Alchemy Systems ___ -- 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 Barge specific extensions
Use AGI,
in PHP, something like that :)
$validExtensions = array(320, 302, 314);
$agi-exec(ZapBarge, $validExtensions[1]);
Thats just an example with phpAGI, you can modify it as you wish.
I am using it.
So, limit user input, they can only Barge valid extensions in array.
Also, you can use g (group) option with ChanSpy
Set a variable to channel which you are going to spy, like:
Set(SPYGROUP=115577)
And then use it
exten = _123,1,NoOp(Going to spy my workmate)
exten = _123,n,ChanSpy(SIP/,g(115577))
then you can only Spy channels with SPYGROUP variable which has value
'115577'...
Take a look ZapBarge and ChanSpy with
'core show application ZapBarge' and 'core show application ChanSpy', then
you can get an idea.
On Tue, Nov 18, 2008 at 4:39 PM, troxlinux [EMAIL PROTECTED] wrote:
Hi amit not you if you can create a group of extensions to spy, but for
example if your extensions are of 3 digits your you can create something
like that...
exten = _*5XXX,1,ChanSpy(SIP/${EXTEN:2},bq)
it configures the hint inside the dialplan to be able to see the state of
the extensions
best regards
2008/11/17 amit salunkhe [EMAIL PROTECTED]
Hi All
Can anybody help me for dial plan to barge or Spy(ExtenSpy)
specificor selective extemsions among 20 extension in my office.
lets say my office extension range is 301-320 i want to barge only 3
extension say 320, 302,314.
is this possible to barge specific extension? . Plz help me for
this.I am using Asterisk 1.4.9 SIP channels.
Regards
Amit
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Re: [asterisk-users] Asterisk not reading fast DTMFs, was: PBX - PRI - * - Telco not working
I have a weird thought... Is the PBX possibly passing the digits both inband and via PRI signaling so Asterisk is getting two digit streams at the same time and totally freaking out? Mikel Lindsaar wrote: I plug the NEC back straight to the Telco and all works well again. I just got on the phone to Digium and we've raised a ticket with some pri intense debugging going on. I'll update the list on findings. Mikel -- http://lindsaar.net/ Rails, RSpec and Life blog ___ -- 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] Asterisk 1.6 call files Disposition=NO ANSWER
Hi Guys, Since moving to Asterisk 1.6, whenever I am using call files the call is always logged with a disposition of NO ANSWER even though the call is connected and answered. The duration displays the correct time. Can anyone explain as to why when using call files the disposition is not correct? ___ -- 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] changing the size of voice packets
John Todd wrote: There was discussion recently (on -dev? on -users? on IRC?) about how there are some shortcomings on RTP packetization/transcoding. It appears, though I have not confirmed this, that trying to move a 20ms G.711 stream from a client, though Asterisk, to a remote gateway using 40ms G.711 will NOT work correctly. The 20ms packet size is passed through without aggregating to 40ms, or vice versa - no change in packetization (though I don't know which side takes precedence.) Going the opposite directon for dis-aggregation (which is what you want to do) I assume would fail in similar ways. I don't recall if changing the codec made any difference on the packetization between two bridged channels. In the past (trunk pre-1.4 and 1.4) both handled aggregation properly, with one important caveat: 1. The media actually flows through Asterisk (no RTP re-invites) If the media is re-invited, it is up to the clients/peers To honor the packetization the remote end requested. If the media is not reinvited and is 100% compatible, codec and packetization, it will go through the packet-to-packet bridge. At one point the P2P bridge did not know about packetization differences and would just relay the RTP packets. I believe that was fixed a long time ago. For what it's worth, 10ms is the maximum rate for most codecs. This creates twice as many packets as 20ms, three times as many as 30ms, etc. - hopefully your network hardware has sufficient power or your call volumes are reasonably low so as not to produce an overwhelming number of Packets Per Second (PPS). Decreasing sampling interval also gets you closer to reaching your NIC's threshhold of PPS, which often is not huge. I seem to recall asking the person who reported that to open a bug in Mantis, but I can't find it, though I didn't look exhaustively. If you can verify this and/or it's relevant to you, please open a ticket so that it at least will be reviewed. I'd open it myself, but I'm a bit resource constrained at the moment in an airport lobby. ___ -- 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] help with dahdi
I am installing dahdi on a machine lspci 00:00.0 Host bridge: Advanced Micro Devices [AMD] RS780 Host Bridge 00:01.0 PCI bridge: Hewlett-Packard Company Unknown device 9602 00:04.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 0) 00:05.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 1) 00:06.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 2) 00:07.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 3) 00:11.0 SATA controller: ATI Technologies Inc SB700/SB800 SATA Controller [AHCI mode] 00:12.0 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI0 Controller 00:12.1 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI1 Controller 00:12.2 USB Controller: ATI Technologies Inc SB700/SB800 USB EHCI Controller 00:13.0 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI0 Controller 00:13.1 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI1 Controller 00:13.2 USB Controller: ATI Technologies Inc SB700/SB800 USB EHCI Controller 00:14.0 SMBus: ATI Technologies Inc SBx00 SMBus Controller (rev 3a) 00:14.1 IDE interface: ATI Technologies Inc SB700/SB800 IDE Controller 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia 00:14.3 ISA bridge: ATI Technologies Inc SB700/SB800 LPC host controller 00:14.4 PCI bridge: ATI Technologies Inc SBx00 PCI to PCI Bridge 00:18.0 Host bridge: Advanced Micro Devices [AMD] Family 11h HyperTransport Configuration (rev 40) 00:18.1 Host bridge: Advanced Micro Devices [AMD] Family 11h Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] Family 11h DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] Family 11h Miscellaneous Control 00:18.4 Host bridge: Advanced Micro Devices [AMD] Family 11h Link Control 01:05.0 VGA compatible controller: ATI Technologies Inc RS780M/RS780MN [Radeon HD 3200 Graphics] 01:05.1 Audio device: ATI Technologies Inc RS780 Azalia controller 08:00.0 System peripheral: JMicron Technologies, Inc. Unknown device 2382 08:00.2 SD Host controller: JMicron Technologies, Inc. Unknown device 2381 08:00.3 System peripheral: JMicron Technologies, Inc. Unknown device 2383 08:00.4 System peripheral: JMicron Technologies, Inc. Unknown device 2384 09:00.0 Ethernet controller: Atheros Communications Inc. AR242x 802.11abg Wireless PCI Express Adapter (rev 01) 0a:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8101E PCI Express Fast Ethernet controller (rev 02) dahdi complete 2.0.0 compiles fine. I am running centos 5.2 x86_64. the service starts fine. lsmod | grep dahdi dahdi_dummy38984 0 dahdi 231888 1 dahdi_dummy crc_ccitt 35265 1 dahdi dahdi_dummy loads as shown. When compiling asterisk 1.4.22 it compiles fine. when running I get the message: ] ERROR[10981]: asterisk.c:3036 main: Asterisk has detected a problem with your DAHDI configuration and will shutdown for your protection. You have options: 1. You only have to compile DAHDI support into Asterisk if you need it. One option is to recompile without DAHDI support. 2. You only have to load DAHDI drivers if you want to take advantage of DAHDI services. One option is to unload DAHDI modules if you don't need them. 3. If you need DAHDI services, you must correctly configure DAHDI. dahdi_speed gives: Count: 1782120 dahdi_test never somes back What dont I have correct? Thanks, Jerry ___ -- 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] Fwd: Polycom phone time behind one hour.
Insert your offset into this line: tcpIpApp.sntp.gmtOffset= eg - EST (GMT -5) = -18000 Bryan M. Johns Shelton | Johns 678.248.2637 Office 678.810.0730 Direct 678.303.3424 Fax Support: [EMAIL PROTECTED] http://www.sheltonjohns.com On Nov 18, 2008, at 5:46 PM, Doug Smith wrote: Tried to submit this email this morning and didn't see it in the list. I apologize if it is a dupe. I've inherited a customized Asterisk installation. After the past time change all clocks in my office are behind by one hour. After some digging it appears we have: A /tftproot/sip.conf that is being pushed out to our phones. I found the following line that seems to be what controls timezone information and DST. I put in carriage returns to make it easier to read as it is all one line. Can anyone see anything obvious (I have missed after reviewing many times) with this config that would cause my phones to be behind an hour? I tried changing overrideDHCP=0 to a 1 with no luck. SNTP tcpIpApp.sntp.resyncPeriod=3600 tcpIpApp.sntp.address=207.207.*.* (Address replaced with asterisk to protect our server IP) tcpIpApp.sntp.address.overrideDHCP=0 tcpIpApp.sntp.gmtOffset= tcpIpApp.sntp.gmtOffset.overrideDHCP=0 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=9 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=4 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0/ Any help at resolving this would be greatly appreciated. Many of our office workers are annoyed that their times are behind an hour now. Thanks, Doug Smith Alchemy Systems ___ -- 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] help with dahdi
lsmod | grep dahdi dahdi_dummy38984 0 dahdi 231888 1 dahdi_dummy crc_ccitt 35265 1 dahdi How did you compile and install this? Did you simply make, make install, make config and chkconfig dahdi on? I assume you edited your /etc/dahdi/modules as your lsmod only shows the dummy? What does dmesg and messages have to say about dahdi? jlc ___ -- 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] help with dahdi
/lsmod | grep dahdi //dahdi_dummy38984 0 //dahdi 231888 1 dahdi_dummy //crc_ccitt 35265 1 dahdi / How did you compile and install this? Did you simply make, make install, make config and chkconfig dahdi on? I assume you edited your /etc/dahdi/modules as your lsmod only shows the dummy? What does dmesg and messages have to say about dahdi? I compiled dahdi 2.0 complete with: make all; make install; linux/build_tools/genudevrules; make config dahdi_dummy is what I am looking for. I dont have any cards installed. Jerry ___ -- 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] help with dahdi
dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.0.0 dahdi_dummy: RTC rate is 1024 dahdi: Registered tone zone 0 (United States / North America) sorry I forgot dmesg. Jerry ___ -- 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] question about connecting with Mobile Base Station
Hi, Is it possible to connect Asterisk with a mobile base station to handle call switching? What kind of protocol will I need to use to convert to sip? Any pointer or info will be greatly appreciated. Best Regards, Mark ___ -- 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] question about connecting with Mobile Base Station
On Tue, Nov 18, 2008 at 22:30, mark morreny [EMAIL PROTECTED] wrote: Hi, Is it possible to connect Asterisk with a mobile base station to handle call switching? What kind of protocol will I need to use to convert to sip? Any pointer or info will be greatly appreciated. There are various devices. PCI GSM card, GSM to Ethernet, or the most basic is GSM to analog, then you connect it to asterisk with e.g. X100 card or SPA3000. Either the PCI or Ethernet devices should work very well -- since the call from the GSM network continues to be digital. An analog adapter will have a slower call setup time, can not support SMS or data and might have echo issues and by definition of a digital-to-analog and subsequent analog-to-digital conversion the quality of the call will be worse (but probably not noticeable). Here is one example: http://www.junghanns.net/en/GSM-PCI_produkt.html ___ -- 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] help with dahdi
I compiled dahdi 2.0 complete with: make all; make install; linux/build_tools/genudevrules; make config As per the readme, I did #make, make install, make config and then double checked chkconfig and although I think /etc/dahdi/modules is for controlling what loads. I suspect as I also have many CentOS 5.2x64 boxes that your issues lies with your genudevrules execution. My dmesg shows the same... Try as I did (and as the readme suggests), my guess is it will be fine. jlc ___ -- 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] two sip listening ports for single asterisk
Rizwan Hisham wrote: Is it possible to make single asterisk server listen on two different ports? Rizwan, There is no way to make a single instance of Asterisk listen on multiple ports. However, you can use an iptables REDIRECT to achieve the same functionality. To redirect a single port with iptables: iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5062 -j REDIRECT --to-ports 5060 This example redirects UPD port 5062 to port 5060, which effectively allows Asterisk to listen on both of them. Remember to save the rule so that it survives a reboot. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- 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] Aeterisk NOW 1.5beta1 - CDR problem....
hello all, Is there any problem with Aeterisk NOW 1.5beta1 with the cdr logging.My *Code:* *CLI cdr status CDR logging: enabled CDR mode: simple CDR output unanswered calls: no CDR registered backend: cdr_manager CDR registered backend: cdr-custom CDR registered backend: mysql *Code:* *CLI cdr mysql status Connected to [EMAIL PROTECTED], port 3306 using table cdr for 30 minutes, 3 seconds. Wrote 0 records since last restart. shows the CDR is enabled in the CSV and in the MYSQL.But nothing is recording.I checked in the /etc/asterisk/ folder and found that there is no cdr.conf and cdr_custom.conf files.I manually added and tried and the result was same. Also there is no file called Master.csv in the asteriskcdr log.Did any body know what may be the reason?. Thanks, Bipin ___ -- 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] question about connecting with Mobile Base Station
Hi Andrew, Thank you for your info. I am actually looking for connecting mobile base station with asterisk via E1. Any idea on where I should start looking? Thanks, Mark On Wed, Nov 19, 2008 at 1:03 PM, Andrew Joakimsen [EMAIL PROTECTED]wrote: On Tue, Nov 18, 2008 at 22:30, mark morreny [EMAIL PROTECTED] wrote: Hi, Is it possible to connect Asterisk with a mobile base station to handle call switching? What kind of protocol will I need to use to convert to sip? Any pointer or info will be greatly appreciated. There are various devices. PCI GSM card, GSM to Ethernet, or the most basic is GSM to analog, then you connect it to asterisk with e.g. X100 card or SPA3000. Either the PCI or Ethernet devices should work very well -- since the call from the GSM network continues to be digital. An analog adapter will have a slower call setup time, can not support SMS or data and might have echo issues and by definition of a digital-to-analog and subsequent analog-to-digital conversion the quality of the call will be worse (but probably not noticeable). Here is one example: http://www.junghanns.net/en/GSM-PCI_produkt.html ___ -- 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
