Re: [asterisk-users] Duplicate incoming channel into two outgoing channels
On 27.03.2014 10:39, jg wrote: Wouldn't it make more sense to handle this by just monitoring the calls and doing everything else with normal data processing? Basically yes, but the whole idea is a workaround to fix issues in legacy systems. klaus -- _ -- 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] IAXModem or T38Modem?
j == joakimsen joakim...@gmail.com writes: j I wouldn't mind if someone posted on the list a known working provider j with the proper configuration to use T.38. In my case I don't consider j it an issue with the provider because they sent the proper T.38 j Invite, but Asterisk IMO does not know how to handle it. Are you using a single credential-tuple with the provider? If the provider supports T.38 and if you can separate out fax lines, there is no need to stick asterisk between them and t38modem. Just have t38modem access the provider directly. Hylafax will handle the rest. (Look for things like sub-account, peer and/or trunk configs.) -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- 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] Video calls using Cisco phones are 176x144(QCIF) and 15FPS both ways
We are experiencing an issue with our Cisco 9971 and 8945 phones where H264 video calls are connecting at 176x144 resolution instead of 640x480. Soft clients can connect at higher resolutions and the 9971 can even receive video at a higher resolution (although it still sends 176x144). I contacted one of the developers and he suggested the passthrough of SDP attributes is not working correctly. Has anyone else experienced this problem? We're running Asterisk 11.8.1. Below are the video parts of the sip debug for one of the phones during a video call. Should I be seeing the a=imageattr in the SIP OK message? --- SIP read from UDP:10.168.154.71:5060 --- INVITE sip:7872@10.162.26.15;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.168.154.71:5060;branch=z9hG4bK1182b2d3 From: Shawn Hughes sip:7871@10.162.26.15 ;tag=20bbc0df35ef052672e68696-0b174da0 To: sip:7872@10.162.26.15 Call-ID: 20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71 Max-Forwards: 70 Date: Fri, 28 Mar 2014 13:51:41 GMT CSeq: 102 INVITE User-Agent: Cisco-CP8945/9.4.1 Contact: sip:7871@10.168.154.71:5060;transport=udp;video Authorization: Digest username=7871,realm=asterisk,uri= sip:7872@10.162.26.15 ;user=phone,response=f51a7522b01c90b81509d2274e9b69bb,nonce=5b43e5a6,algorithm=MD5 Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.2,X-cisco-xsi-8.5.1 Allow-Events: kpml,dialog Content-Length: 685 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 27778 0 IN IP4 10.168.154.71 s=SIP Call t=0 0 m=audio 10032 RTP/AVP 0 8 18 102 9 116 101 c=IN IP4 10.168.154.71 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:102 L16/16000 a=rtpmap:9 G722/8000 a=rtpmap:116 iLBC/8000 a=fmtp:116 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv m=video 10034 RTP/AVP 97 c=IN IP4 10.168.154.71 b=TIAS:200 a=rtpmap:97 H264/9 a=fmtp:97 profile-level-id=428014;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=36000;max-fs=1200 a=imageattr:97 send [x=640,y=480] [x=640,y=360] [x=352,y=288] [x=176,y=144] recv [x=640,y=480] a=sendrecv - --- (19 headers 24 lines) --- Sending to 10.168.154.71:5060 (no NAT) Using INVITE request as basis request - 20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71 Found peer '7871' for '7871' from 10.168.154.71:5060 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 102 Found RTP audio format 9 Found RTP audio format 116 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format L16 for ID 102 Found audio description format G722 for ID 9 Found audio description format iLBC for ID 116 Found audio description format telephone-event for ID 101 Found RTP video format 97 Found video description format H264 for ID 97 Capabilities: us - (gsm|ulaw|alaw|g722|h264), peer - audio=(ulaw|alaw|g729|ilbc|g722|slin16)/video=(h264)/text=(nothing), combined - (ulaw|alaw|g722|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.168.154.71:10032 Peer video RTP is at port 10.168.154.71:10034 Looking for 7872 in from-internal (domain 10.162.26.15) list_route: hop: sip:7871@10.168.154.71:5060;transport=udp SIP/2.0 200 OK Via: SIP/2.0/UDP 10.168.154.71:5060 ;branch=z9hG4bK1182b2d3;received=10.168.154.71 From: Shawn Hughes sip:7871@10.162.26.15 ;tag=20bbc0df35ef052672e68696-0b174da0 To: sip:7872@10.162.26.15;tag=as1c2f9ae5 Call-ID: 20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71 CSeq: 102 INVITE Server: FPBX-2.11.0(11.8.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:7872@10.162.26.15:5060 Content-Type: application/sdp Content-Length: 467 v=0 o=root 283568327 283568327 IN IP4 10.162.26.15 s=Asterisk PBX 11.8.1 c=IN IP4 10.162.26.15 b=CT:3600 t=0 0 m=audio 13434 RTP/AVP 0 8 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 15496 RTP/AVP 97 a=rtpmap:97 H264/9 a=fmtp:97 profile-level-id=428014;max-mbps=36000;max-fs=1200;packetization-mode=0;level-asymmetry-allowed=1 a=sendrecv -- _ -- 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
Re: [asterisk-users] Video calls using Cisco phones are 176x144(QCIF) and 15FPS both ways
Matt Rabbitt wrote: We are experiencing an issue with our Cisco 9971 and 8945 phones where H264 video calls are connecting at 176x144 resolution instead of 640x480. Soft clients can connect at higher resolutions and the 9971 can even receive video at a higher resolution (although it still sends 176x144). I contacted one of the developers and he suggested the passthrough of SDP attributes is not working correctly. Has anyone else experienced this problem? We're running Asterisk 11.8.1. Below are the video parts of the sip debug for one of the phones during a video call. Should I be seeing the a=imageattr in the SIP OK message? It looks as though the passthrough for fmtp is indeed working but as the imageattr attribute is currently unsupported/not used/not passed through it is probably causing your resolution problem. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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] Video calls using Cisco phones are 176x144(QCIF) and 15FPS both ways
What would need to be changed in the source code to accommodate this? Can the imageattr attribute be hard coded into h264_format_attr_sdp_generate() in res_format_attr_h264.c? On Mon, Mar 31, 2014 at 9:07 AM, Joshua Colp jc...@digium.com wrote: Matt Rabbitt wrote: We are experiencing an issue with our Cisco 9971 and 8945 phones where H264 video calls are connecting at 176x144 resolution instead of 640x480. Soft clients can connect at higher resolutions and the 9971 can even receive video at a higher resolution (although it still sends 176x144). I contacted one of the developers and he suggested the passthrough of SDP attributes is not working correctly. Has anyone else experienced this problem? We're running Asterisk 11.8.1. Below are the video parts of the sip debug for one of the phones during a video call. Should I be seeing the a=imageattr in the SIP OK message? It looks as though the passthrough for fmtp is indeed working but as the imageattr attribute is currently unsupported/not used/not passed through it is probably causing your resolution problem. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video calls using Cisco phones are 176x144(QCIF) and 15FPS both ways
Matt Rabbitt wrote: What would need to be changed in the source code to accommodate this? Can the imageattr attribute be hard coded into h264_format_attr_sdp_generate() in res_format_attr_h264.c? A lot. Yes, you could hard code it. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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
[asterisk-users] Function REGEX
Hi I need help to use the function REGEX. My question is if is possible test a expression as [X123 == 5123] ( If an extension corresponding to a previously defined regular expression). I saw various examples about this function, but nothing as the my needs. I do not understanding exactly how to works this function. Thank's Att, *Rafael dos Santos Saraiva* http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 -- _ -- 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] Function REGEX
Here is an example from one of my production dialplans
same =
n,ExecIf(${REGEX(^1205|^1256|^1850|^1718|^1212|^1917|^1347|^1646|^1929
${CALLERID(num)})}]?Hangup)
Assuming you meant 0-9 and not the literal X (which means nothing special in
regular expressions):
same = n,ExecIf(${REGEX(^[0-9]5123$ ${EXTEN})}]?Hangup)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael dos Santos
Saraiva
Sent: Monday, March 31, 2014 11:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Function REGEX
Hi
I need help to use the function REGEX. My question is if is possible test a
expression as [X123 == 5123] ( If an extension corresponding to a previously
defined regular expression). I saw various examples about this function, but
nothing as the my needs. I do not understanding exactly how to works this
function.
Thank's
Att,
Rafael dos Santos Saraiva
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Function REGEX
All working fine.
Thank you for your help.
Att,
*Rafael dos Santos Saraiva*
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230
2014-03-31 12:29 GMT-03:00 Eric Wieling ewiel...@nyigc.com:
Here is an example from one of my production dialplans
same =
n,ExecIf(${REGEX(^1205|^1256|^1850|^1718|^1212|^1917|^1347|^1646|^1929
${CALLERID(num)})}]?Hangup)
Assuming you meant 0-9 and not the literal X (which means nothing special
in regular expressions):
same = n,ExecIf(${REGEX(^[0-9]5123$ ${EXTEN})}]?Hangup)
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael dos Santos
Saraiva
Sent: Monday, March 31, 2014 11:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Function REGEX
Hi
I need help to use the function REGEX. My question is if is possible test
a expression as [X123 == 5123] ( If an extension corresponding to a
previously defined regular expression). I saw various examples about this
function, but nothing as the my needs. I do not understanding exactly how
to works this function.
Thank's
Att,
Rafael dos Santos Saraiva
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
_
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Re: [asterisk-users] CLI command to see if SRTP is active?
On Fri, Mar 28, 2014 at 7:39 PM, Patrick Laimbock patr...@laimbock.com wrote: Hi, I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI command to see if SRTP is active on a channel/call. I went through sip show ... and core show channel... and did not see any mentioning of SRTP while there is an SRTP call active. I don't have any encrypted calls up in front of me at this second to provide an example, however if you are just wanting to verify SRTP is active for a call you will see SRTP related messages on the CLI if you turn up DEBUG message verbosity and have it going to the console, or else possibly with output from rtp set debug on. As for the show channels type commands, it may say something about encryption rather than SRTP directly. I'll take a look later if I get a chance. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://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] CLI command to see if SRTP is active?
On Mon, Mar 31, 2014 at 1:26 PM, Rusty Newton rnew...@digium.com wrote: On Fri, Mar 28, 2014 at 7:39 PM, Patrick Laimbock patr...@laimbock.com wrote: Hi, I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI command to see if SRTP is active on a channel/call. I went through sip show ... and core show channel... and did not see any mentioning of SRTP while there is an SRTP call active. I don't have any encrypted calls up in front of me at this second to provide an example, however if you are just wanting to verify SRTP is active for a call you will see SRTP related messages on the CLI if you turn up DEBUG message verbosity and have it going to the console, or else possibly with output from rtp set debug on. As for the show channels type commands, it may say something about encryption rather than SRTP directly. I'll take a look later if I get a chance. I see your issue on the tracker now, posting the link here for the sake of those who read the archives https://issues.asterisk.org/jira/browse/ASTERISK-23564 -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://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
[asterisk-users] DAHDI-Linux v2.9.1.1 Now Available
The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.9.1.1 dahdi-linux-complete-2.9.1.1+2.9.1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete Fix for bug preventing DAHDI from building in a chroot environment Issues closed in this release: DAHLIN-337 Shortlog of dahdi-linux changes since v2.9.1: Tzafrir Cohen (1): firmware: Honor DESTDIR when installing firmware. The diffstat from the dahdi-linux v2.9.1 release: drivers/dahdi/firmware/Makefile | 31 --- 1 file changed, 16 insertions(+), 15 deletions(-) For a full list of changes in these releases, please see the shortlog at: http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.9.1.1 Issues found in this release can be reported in the DAHDI-Linux [1] and DAHDI-Tools [2] projects at https://issues.asterisk.org/jira [1] https://issues.asterisk.org/jira/browse/DAHLIN [2] https://issues.asterisk.org/jira/browse/DAHTOOL Thank you for your continued support of Asterisk! -- _ -- 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] Need more meetme users -- hitting some limit
On Fri, Mar 21, 2014 at 11:26:22AM -0700, Steve Edwards wrote: On Fri, 21 Mar 2014, Steve Totaro wrote: I found below here: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe If you have too many conferences, one CPU may not be able to mix all the audio and you will have audio problems even if there are 7+ other CPUs that are essentially idle while waiting for one CPU to mix everything. You should be able to handle 512 conference participants on a modern server system without problem. The current trunk of DAHDI linux limits the number of open pseudo channels to 512 for this reason. [1] With 312 calls distributed across 100 meetmes, 'top' shows 1 core at 32%, 1 core at 6% and the rest basically idle. So it looks like meetme() is still a single CPU application, but I have plenty of CPU headroom. Coincidentally, 512 is my target. Any clues on how to get 200 more? Steve, If you're looking to reduce the CPU overhead of processing meetme conferences, this email from awhile ago may be of some help: http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/51750/focus=51777 Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to build DAHDI-Linux in mock chroot
On Sunday, March 30, 2014 02:24:35 PM Anthony Messina wrote: On Sunday, March 30, 2014 07:07:47 PM Tzafrir Cohen wrote: On Fri, Mar 28, 2014 at 07:57:54PM -0500, Anthony Messina wrote: On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote: Unfortunately, after http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc 1c 1fb1 2cc0661f3810ef47ad33206b2e398 I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I believe this is related to the Makefile calling install_firmware with only 2 args, where install_firmware is a shell script with DESTDIR set to $3, which is empty. In this case, the DESTDIR evaluates to /usr/lib/hotplug/firmware, rather than buildroot_destdir/usr/lib/hotplug/firmware. make -C drivers/dahdi/firmware hotplug-install DESTDIR=/builddir/build/BUILDROOT/dahdi-linux-2.9.1-1.fc20.x86_64 HOTPLUG_FIRMWARE=yes make[1]: Entering directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/usr/lib/hotplug/firmware mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/lib/firmware Installing dahdi-fw-oct6114-032.bin to hotplug firmware directories install: cannot create regular file '/usr/lib/hotplug/firmware': No such file or directory make[1]: *** [hotplug-install] Error 1 make[1]: Leaving directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' make: *** [install-firmware] Error 2 https://issues.asterisk.org/jira/browse/DAHLIN-337 Thanks for your report. I hope to get it fixed soon. I should note that this specific target does not belong in a proper chroot build, as it downloads from outside. How can I get those firmware files properly included? This is the spec file I use: https://messinet.com/rpms/browser/dahdi-linux/dahdi-linux.spec DAHDI-Linux-2.9.1.1 fixes this issue. Thank you. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Need more meetme users -- hitting some limit
On Mon, 31 Mar 2014, Shaun Ruffell wrote: If you're looking to reduce the CPU overhead of processing meetme conferences, this email from awhile ago may be of some help: http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/51750/focus=51777 Thanks for the clue. I can hit my target of 512 on an Intel E3-1240v3 with 'pre-packaged' Asterisk so I'm good for now. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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
