Re: [asterisk-users] DTMF-troubles with Snom
Jonas What is the dtmf setting on all peers involved in the call? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Jonas Kellens jonas.kell...@telenet.be Sent: Wednesday, January 05, 2011 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] DTMF-troubles with Snom Hello list, I'm having DTMF-troubles with a Snom phone. I want to know if it's the Snom or Asterisk that makes the trouble. I'm playing a prompt, then make a choice for 2 : [Jan 5 17:06:38] VERBOSE[29172] file.c: [Jan 5 17:06:38] -- SIP/test1-0701 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' (language 'nl') [Jan 5 17:06:39] VERBOSE[29172] pbx.c: [Jan 5 17:06:39] -- Executing [...@sub-routing:52] WaitExten(SIP/test1-0701, 15) in new stack [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF begin '2' received on SIP/test1-0701 [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF begin ignored '2' on SIP/test1-0701 [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on SIP/test1-0701, duration 160 ms [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on SIP/test1-0701 What follows is a prompt again, and it automatically chooses option 2 : [Jan 5 17:06:41] VERBOSE[29172] file.c: [Jan 5 17:06:41] -- SIP/test1-0701 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl') [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on SIP/test1-0701, duration 160 ms [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on SIP/test1-0701 Even without pressing 2 on the Snom phone, option 2 is chosen in the menu. The above is different when I do the same with a Grandstream device : [Jan 5 17:14:15] VERBOSE[29384] file.c: [Jan 5 17:14:15] -- SIP/test6-0714 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' ( language 'nl') [Jan 5 17:14:17] VERBOSE[29384] pbx.c: [Jan 5 17:14:17] -- Executing [...@sub-routing:52] WaitExten(SIP/test6-0714, 15) in new stack [Jan 5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan 5 17:14:18] doing dnsmgr_lookup for 'ssw4.brussels.weepee.org' [Jan 5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan 5 17:14:18] doing dnsmgr_lookup for 'ssw4.brussels.weepee.org' [Jan 5 17:14:21] DTMF[29384] channel.c: DTMF begin '2' received on SIP/test6-0714 [Jan 5 17:14:21] DTMF[29384] channel.c: DTMF begin ignored '2' on SIP/test6-0714 [Jan 5 17:14:21] DTMF[29384] channel.c: DTMF end '2' received on SIP/test6-0714, duration 100 ms [Jan 5 17:14:21] DTMF[29384] channel.c: DTMF end passthrough '2' on SIP/test6-0714 [Jan 5 17:14:38] VERBOSE[29384] file.c: [Jan 5 17:14:38] -- SIP/test6-0714 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl') [Jan 5 17:14:39] VERBOSE[29384] pbx.c: [Jan 5 17:14:39] -- Executing [...@sub-routing:52] WaitExten(SIP/test6-0714, 15) in new stack [Jan 5 17:14:44] DTMF[29384] channel.c: DTMF begin '2' received on SIP/test6-0714 [Jan 5 17:14:44] DTMF[29384] channel.c: DTMF begin ignored '2' on SIP/test6-0714 [Jan 5 17:14:44] DTMF[29384] channel.c: DTMF end '2' received on SIP/test6-0714, duration 100 ms [Jan 5 17:14:44] DTMF[29384] channel.c: DTMF end passthrough '2' on SIP/test6-0714 Here I explicitly chose option 2 by pressing on button 2. What is going on with the Snom ? There is a difference in duration (160ms vs 100ms). Is that the problem ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI-Macro w/Agruments
OK, I need to dial a macro from AGI and needs to pass an argument.
Ok, I found an bug report, but it was stated un fixable? really after 5
years?
https://issues.asterisk.org/view.php?id=2470
I found this email in the archive, but no solution other then the dodgy work
around?
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg85048.html
I have tried this, but it doesn't work.
$AGI-set_variable('DAILNO', $BranchPhone);
$AGI-exec(Macro,agidial);
And my macro:
[macro-agidial]
exten = s,1,AGI(getcid.pl,${CALLERID(NUM)},1)
exten = s,2,NoOp(DIALNO=${DIALNO})
exten = s,3,Dial(SIP/${dial...@sipprovider,60)
exten = s,4,GotoIf($[${DIALSTATUS} = CONGESTION]?10)
exten = s,6,Hangup()
exten = s,10,Dial(IAX2/SERVER2/${DIALNO})
exten = s,12,Hangup()
but when the macro is called, Dialno = nothing.
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[asterisk-users] Definations of READ/WRITE parameters of manager.conf contexts?
Hi Everyone, I want to know each and every parameter's detail that can be included in the read= write= in manager.conf Where can I find this? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk replying to wrong port for NOTIFY messages
Hi Jeff, On Thu, Jan 6, 2011 at 11:28 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 5 Jan 2011, James Lamanna wrote: See the following SIP trace. Where in the world does Asterisk get port 1025 to respond to? This is asterisk 1.6.x. Hi James, I'm sure it would be the NAT translated port on the public side of the customer's firewall... Unfortunately its not. All clients are on symmetric NAT. Here's an ngrep trace, you can see the NAT port in the VIA is the same as the source port: U xxx.xxx.xxx.44:8155 - xx.xxx.xxx.7:5060 NOTIFY sip:pbx1.warp2biz.com SIP/2.0..Via: SIP/2.0/UDP 192.168.1.127:8155;branch=z9hG4bK-4b50c77d..From: zz sip:zzz...@pbx1.example.com;tag=5281a88170274fa2o0..To: sip:pbx1.example.com..Call-ID: c914b8d-532f2...@192.168.1.127..cseq: 14492 NOTIFY..Max-Forwards: 70..Con tact: zz sip:zzz...@192.168.1.127:8155..Event: keep-alive..User-Agent: Cisco/SPA509G-7.4.6-0002fdff9097..Content-Length: 0 # U xx.xxx.xxx.7:5060 - xx.xxx.xxx.44:1025 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.127:8155;branch=z9hG4bK-4b50c77d;received=xx.xxx.xxx.44..From: zz sip:zzz...@pbx1.example.com;tag=5281a88170274fa2o0..To: sip:pbx1.example.com;tag=as62dac391..Call-ID: c914b8d-532f2...@192.168.1.127..cseq: 14492 NOTIFY..User- Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Content-Length: 0 -- James j Thanks. -- James --- SIP read from zzz.zzz.zzz.44:9363 --- NOTIFY sip:pbx1.mydomain.com SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3^M From: xxx-xxx- sip:xxx...@pbx1.mydomain.com;tag=467525dd6fac949do0^M To: sip:pbx1.mydomain.com^M Call-ID: 707176dd-38f47...@192.168.1.140^m CSeq: 118907 NOTIFY^M Max-Forwards: 70^M Contact: xxx-xxx- sip:xx...@192.168.1.140:9363^M Event: keep-alive^M User-Agent: Cisco/SPA509G-7.4.6-0002fdff90a4^M Content-Length: 0^M ^M - [Jan 5 13:46:36] VERBOSE[3919] logger.c: --- (11 headers 0 lines) --- [Jan 5 13:46:36] VERBOSE[3919] logger.c: --- Transmitting (no NAT) to zzz.zzz.zzz.44:1025 --- SIP/2.0 200 OK^M Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3;received=zzz.zzz.zzz.44^M From: xxx-xxx- sip:xx...@pbx1.mydomain.com;tag=467525dd6fac949do0^M To: sip:pbx1.mydomain.com;tag=as0493c604^M Call-ID: 707176dd-38f47...@192.168.1.140^m CSeq: 118907 NOTIFY^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO^M Supported: replaces^M Content-Length: 0^M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?
On 01/06/2011 11:34 AM, mgra...@mstvp.com wrote: We should also be very clear that the Siren codecs are supported on the Polycom SoundStation conference phones and the VVX-1500 Business Media Phones. These codecs are not supported in the SoundPoint desk phones. The SoundPoint series support the more basic G.722 codec in the IP335/450/550/560/650/670 models. The SoundPoint IP6000 and IP7000 conference phones (and maybe the IP5000, I haven't checked) also support G.722.1 and G.722.1C. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Benefit of PRI vs SIP trunk calls
PRICAUSE will give you lots of info on why a call was hungup on. Not sure if SIP will give you the same. On Thu, Jan 6, 2011 at 9:06 AM, Jim Dickenson dicken...@cfmc.com wrote: Does Asterisk, currently using version 1.4, get any more information about the result of an outbound call made over a PRI line compared to a call via a SIP trunk? As an example, in a PRI call there is this message that shows up on the console: [2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network. for a call to a fax machine. Does asterisk set anything that a dialplan can access that can know the call was to a fax machine? If a call is placed to a number that is disconnected so a special information tone is played can either a PRI call or a SIP call know this without analyzing the audio stream? Are there reasons to prefer the use of PRI over SIP or SIP over PRI? I would like people's opinions as to if one form is better than the other in any meaningful way. Thanks for you feed-back. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Benefit of PRI vs SIP trunk calls
On Jan 6, 2011, at 8:08 PM, Joel Maslak wrote: On Thu, Jan 6, 2011 at 7:06 AM, Jim Dickenson dicken...@cfmc.com wrote: Are there reasons to prefer the use of PRI over SIP or SIP over PRI? [snip] I run the PBX for my organization which has about 160 extensions. I wouldn't even think of doing anything but PRI for the main lines because (A) for our size organization where we are located, we're talking a couple hundred dollars a month difference between PRI and SIP in cost so it's nearly break-even in cost which means cost difference isn't a huge motivator, (B) it supports FAX, modems, and TTYs - perfectly, (C) Quality is 100% consistent. In addition, the reliability is good enough that I'm willing to use it for 911. [snip] I have to agree with most of what Joel said in his message. For me, the main problem with many sip implementations is that your phone service will be only as reliable as your internet service. If you have a dedicated internet line that is highly reliable, that's not a big deal, but DSL, Cable, and the like aren't reliable enough for our needs. Having said that, one downside of a PRI is that you are paying for all of those channels, even when you aren't using them. Companies like Paetec and most other large telcos are offering SIP trunks over an MPLS circuit, running on a T1 loop. This covers the reliability problem, as you are running over the same type of circuit as your PRI, and it allows you to take advantage of unused channels as data bandwidth. This is especially helpful for folks who have a data T1 and a PRI, as they can get higher bandwidth for data when there isn't much voice traffic. Because they use G.729, you can also fit more calls on the same circuit. That choice of codec eliminates the ability to send/receive faxes, though, and it's likely expensive when compared to other SIP solutions, but it does appear to be pretty slick. Another benefit of SIP is that it doesn't require a Digium, Sangoma, or similar interface card in the server, simplifying migrations and reducing cost in many scenarios. Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 0.7.5 released
The AstLinux Team is happy to announce the release of AstLinux 0.7.5 with options for both Asterisk 1.8.1.1 and Asterisk 1.4.36. More information about the release is available on our website: http://www.astlinux.org/content/astlinux-075-release Direct links to the installation files are available here: http://www.astlinux.org/release/075-asterisk-1811 http://www.astlinux.org/release/075-asterisk-1436 All current users are encouraged to upgrade to one of those releases. A firmware upgrade can be performed from the web interface or from the command line. Command line upgrade: (for Asterisk 1.4) upgrade-run-image check http://mirror.astlinux.org/firmware (should report astlinux-0.7.5) then upgrade-run-image upgrade http://mirror.astlinux.org/firmware or (for Asterisk 1.8) upgrade-run-image check http://mirror.astlinux.org/ast18-firmware (should report astlinux-0.7.5) upgrade-run-image check http://mirror.astlinux.org/ast18-firmware If you are upgrading from an Asterisk 1.4 base to Asterisk 1.8, you will need to manually update any Asterisk related configuration files. Please ask any questions about this release on the AstLinux-user's mailing list. -- The AstLinux Team -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call queues on load-balanced asterisks
Hello Pan, You can user DB for this just make real time configuration of Queue and make all asterisk server connected to Same DB if more load then use replication for different server on DB, also So that Quque name should be same for all server and asterisk can call same agent. you didnot mentioned that which purpose youwere use queue other wise i can give answer in better way. regards Dhaval On Fri, Jan 7, 2011 at 5:08 PM, Pan B. Christensen p...@ibidium.no wrote: Hello, I have been asked to implement the following design: Load-balanced Kamailio servers handling registrations and routing. Load-balanced asterisk feature servers handling voicemail and other things Kamailio cannot do. Plus several load-balanced gateways, but they are not relevant to my question. All this is working fine. I've now been asked to start implementing calling queues, and my question is this: How can I implement the same queue on multiple Asterisk servers? Let's say that 10 people call the same queue. These calls would then currently be distributed 5 to Asterisk A and 5 to Asterisk B. How can I make Asterisk A respect the 5 people queued on the other server and vice versa? Will the customer need to change their design to make the feature servers master-slave with failover instead of load-balanced? Mvh Pan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream GXE2504A codec disable option
Dear All Among all the readers anybody have ever work on Granstream device GXE2504A which act as ippbx and having GUI to configure and maintain. We are facing one problem with this device, thsi device reply or adding codec like ilbc,G.721 which is not supported by our Asterisk server or our SBC. We want to disable this codecs, but form available GUI we not able to see any option to disble it. If anybody having any experince with this device plz share the same to disable the scuh codecs. Regards Amit -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF-troubles with Snom
Hello, I have tried several settings. Normally I set it to rfc 2833 on most phone types (Grandstream/YeaLink/Cisco SPA). Works always. With Snom you have the option : SIP info : on/off/always Neither of these settings make any difference... What setting do you have in your Snom phones ?? The problem only occurs when calling with a Snom phone... If you have 1 IVR-menu, then there's no problem. But when you add an extra layer (a second IVR-menu), then your first input is used for the second IVR also. Strange ! Kind regards, Jonas. On 01/07/2011 04:36 PM, Bryant Zimmerman wrote: Jonas What is the dtmf setting on all peers involved in the call? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 *From*: Jonas Kellens jonas.kell...@telenet.be *Sent*: Wednesday, January 05, 2011 4:55 PM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: [asterisk-users] DTMF-troubles with Snom Hello list, I'm having DTMF-troubles with a Snom phone. I want to know if it's the Snom or Asterisk that makes the trouble. I'm playing a prompt, then make a choice for 2 : [Jan 5 17:06:38] VERBOSE[29172] file.c: [Jan 5 17:06:38] -- SIP/test1-0701 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' (language 'nl') [Jan 5 17:06:39] VERBOSE[29172] pbx.c: [Jan 5 17:06:39] -- Executing [...@sub-routing:52] WaitExten(SIP/test1-0701, 15) in new stack *[Jan 5 17:06:41] DTMF[29172] channel.c: DTMF begin '2' received on SIP/test1-0701 [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF begin ignored '2' on SIP/test1-0701 [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on SIP/test1-0701, duration 160 ms [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on SIP/test1-0701* What follows is a prompt again, and it automatically chooses option 2 : [Jan 5 17:06:41] VERBOSE[29172] file.c: [Jan 5 17:06:41] -- SIP/test1-0701 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl') *[Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on SIP/test1-0701, duration 160 ms [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on SIP/test1-0701* Even without pressing 2 on the Snom phone, option 2 is chosen in the menu. The above is different when I do the same with a Grandstream device : [Jan 5 17:14:15] VERBOSE[29384] file.c: [Jan 5 17:14:15] -- SIP/test6-0714 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' ( language 'nl') [Jan 5 17:14:17] VERBOSE[29384] pbx.c: [Jan 5 17:14:17] -- Executing [...@sub-routing:52] WaitExten(SIP/test6-0714, 15) in new stack [Jan 5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan 5 17:14:18] doing dnsmgr_lookup for 'ssw4.brussels.weepee.org' [Jan 5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan 5 17:14:18] doing dnsmgr_lookup for 'ssw4.brussels.weepee.org' *[Jan 5 17:14:21] DTMF[29384] channel.c: DTMF begin '2' received on SIP/test6-0714 [Jan 5 17:14:21] DTMF[29384] channel.c: DTMF begin ignored '2' on SIP/test6-0714 [Jan 5 17:14:21] DTMF[29384] channel.c: DTMF end '2' received on SIP/test6-0714, duration 100 ms [Jan 5 17:14:21] DTMF[29384] channel.c: DTMF end passthrough '2' on SIP/test6-0714* [Jan 5 17:14:38] VERBOSE[29384] file.c: [Jan 5 17:14:38] -- SIP/test6-0714 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl') [Jan 5 17:14:39] VERBOSE[29384] pbx.c: [Jan 5 17:14:39] -- Executing [...@sub-routing:52] WaitExten(SIP/test6-0714, 15) in new stack *[Jan 5 17:14:44] DTMF[29384] channel.c: DTMF begin '2' received on SIP/test6-0714 [Jan 5 17:14:44] DTMF[29384] channel.c: DTMF begin ignored '2' on SIP/test6-0714 [Jan 5 17:14:44] DTMF[29384] channel.c: DTMF end '2' received on SIP/test6-0714, duration 100 ms [Jan 5 17:14:44] DTMF[29384] channel.c: DTMF end passthrough '2' on SIP/test6-0714* Here I explicitly chose option 2 by pressing on button 2. What is going on with the Snom ? There is a difference in duration (160ms vs 100ms). Is that the problem ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update
