Re: [asterisk-users] openvz
On Fri, Sep 3, 2010 at 6:11 AM, mattias m...@mjw.se wrote: Can i run asterisk on a openvz vps or do i need a kernel? I dont use dadi -- Works just fine for our voicemail server (~450 users). CP. -- _ -- 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] Update the LCD with the callee's name after dialing
On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com wrote: Thank you Andrew, I will check it out. We are currently running 1.4. -Matt On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham lath...@gmail.com wrote: Remote Party ID in trunk, it works There are hacks for other versions. We use the patch in https://issues.asterisk.org/view.php?id=6643. Works great. CP -- _ -- 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] Is this failed Asterisk setup typical?
On Thu, Jun 3, 2010 at 6:16 AM, Gilles codecompl...@free.fr wrote: Hello I just read this article and would like some feedback from experienced Asterisk users: === Failed open source VoIP deployment leads to hosted VoIP strategy By Jessica Scarpati snip http://searchunifiedcommunications.techtarget.com/news/article/0,289142,sid186_gci1508323_mem1,00.html (free registration required) === So it looks like this company had the following issues: * No in-house technical expertise to set up and maintain Asterisk * Not enough bandwidth * DID module apparently not reliable Based on your experience, are those problems typical? Thank you. Not in our experience as a 500-phone, 20-site install for a municipal government. We are just migrating from our first generation install to replacement hardware (to new blades from servers that are now 5 years old) and are still committed to Asterisk for it. Done right, Asterisk saved us over three quarters of a million dollars over a big-C install. CP -- _ -- 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] RPID on called party
https://issues.asterisk.org/view.php?id=6643 CP On Thu, Apr 1, 2010 at 7:36 AM, Ondrej Valousek webs...@s3group.cz wrote: Hello, Did anyone manage to force asterisk to put Remote-party-ID attribute on the SIP outgoing call? I.e. When A calls B, I want that A gets a name of B displayed on his phone. Note that name of A gets displayed on the B's phone fine, but this is not what I want. This works with Cisco Call manager fine - the RPID is sent as a part of the response to the SIP INVITE this way: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.60.20:5060;branch=z9hG4bK42892c32;rport From: Ondrej Valousek sip:7...@192.168.60.20 sip:7...@192.168.60.20 ;tag=as4786d518 To: sip:1...@192.168.62.12 sip:1...@192.168.62.12 ;tag=f75ff5d8-1023-4240-bc4b-d7eeb6d0d77d-42063104 Date: Tue, 30 Mar 2010 13:53:15 GMT Call-ID: 465a9c200587260d164f451409489...@192.168.60.20 CSeq: 102 INVITE Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY Allow-Events: presence *Remote-Party-ID: Paul Ryan sip:1...@192.168.62.12 sip:1...@192.168.62.12 ;party=called;screen=yes;privacy=off* Contact: sip:1...@192.168.62.12:5060 sip:1...@192.168.62.12:5060 Content-Length: 0 But I can not make it working with Asterisk. Does anyone have any glue how to achieve this WITHOUT patching asterisk? I am happy to upgrade to the latest/greatest version, I just do not want to patch. Many thanks, Ondrej -- _ -- 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] Polycom phone DND state
On Fri, Jan 22, 2010 at 7:50 AM, Mike l...@virtutel.ca wrote: I know having Asterisk aware of Polycom Do No Disturb state wasn't working before (1.4), but is this working in any recent version? Is there any custom way of doing this? Our Asterisk servers (1.2 and 1.4) get SIP response 603 Decline when our Polycoms are on DND. CP -- _ -- 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] Echo issue
On Tue, Dec 8, 2009 at 7:37 AM, Noah Miller noahisaacmil...@gmail.com wrote: Hi - I am having echo issues on our Asterisk box using a PRI circuit. I was using the software echo cancellation and that helped a bit but didn't solve it completely. So I went and bought a Digium echo cancellation module for the TE121 card. That made it even worst, getting more echo on external calls and between internal extension to extension. The echo doesn't happen all the time, but enough to get complaints from our users. Completely fed up with the issue, I removed the module from the card. Can someone guide me on how to fix/tune/address the echo issues. You can likely eliminate most echo on a PRI by setting txgain and rxgain. Are you using dahdi or zaptel? If Dahdi, what do your system.conf and chan_dahdi.conf look like? If zaptel, what do your zaptel.conf and zapata.conf look like? When you say you have echo on calls that are internal extension to internal extension, are the endpoints using dahdi/zaptel or some voip technology (sip, iax, mgcp, skinny, etc)? If voip, any echo is acoustically generated by the endpoints themselves. On voip calls I've often had this happen when the endpoints are using headsets, or have gain levels set very high. - Noah We found ourselves in a similar situation during our rollout and solved it with a quad-span Ditech echo-cancellation appliance (http://www.ditechnetworks.com/products/quad-2_echo-canceller.html). It's a couple of grand, but after months of playing with software EC, the hardware modules and every zaptel setting we could find, this appliance removed echo like flipping a switch. The metrics we later obtained from it clearly showed that we simply had tail on a long loop to an old CO switch that exceeded the maximum 128ms that either software EC or the hardware module could handle. The side benefits are that we get all sorts of metrics from the appliance, and we also get adaptive gain, which solved another problem we had with trying to find gain settings that suited both softly-spoken and strident users. The support from Ditech was excellent and we haven't looked back. CP ___ -- 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] Receptionist GUI?
On Mon, Oct 5, 2009 at 11:34 AM, Ken D'Ambrosio k...@jots.org wrote: Hey, all. Just wondering if there's a GUI out there -- preferably OSS, but I'll take what-have-you -- that a) can run on an Ubuntu/Debian box, and b) allows a receptionist to see what calls are in-process, and forward calls from their phone to somewhere else. Thanks! -Ken I can add a recommendation for iSymphony - cheaper than dirt, easy to configure, and the users like it. CP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is DNS SRV enough for failover?
This is so wrong it's not funny. The caching of DNS SRV records acts in its favor when it comes to failover - the UAs already have the information they need in their resolver cache to perform the failover without having to make another DNS query. The TTL you need to worry about is that of the SIP registration - UAs will typically renew their SIP registration at the half-life of the TTL. A short SIP registration TTL will permit devices which are not actively placing calls to failover more quickly than they otherwise would. Of course, a balance must be struck between short SIP registration TTLs, and the amount of SIP registration traffic this generates. YMMV. CP On Tue, 2008-09-30 at 22:07 -0400, Alex Balashov wrote: Nhadie wrote: hi, i'm using DNS SRV for failover, i tried to test shutting the server down, sip client should still register on the other server but it did not. i'm using x-lite which i don't know if it's doing a srv query. does this mean SRV is not enough for failover? if a client has dns caching would this cause a problem? SRV records are DNS. DNS is cached. Ergo, SRV records are cached. Ergo, if they are cached excessively - either because the TTL is long, or in defiance of the TTL - it can cause a problem. No, DNS is not a good way to do real-time failover for anything. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom phones and DNS SRV
Just in case anyone is having DNS SRV timeouts with their Polycom phones, the following Polycom KB article should help: http://knowledgebase.polycom.com/kb/search.do?cmd=displayKCdocType=kcexternalId=12856sliceId=SAL_PUBLIC_1_2dialogID=7620671stateId=1 We have set tcpIpApp.port.rtp.mediaPortRangeStart to 65000. Based on our experience and the fact that the phone's DNS resolver starts over from port 1026 on a reboot and increments from there, this should give us about a year before the ports overlap again, in the unlikely event that the phones won't get rebooted in the meantime. YMMV. CP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Resilience using DNS or phone feature ?
Oliver, We use DNS SRV records combined with short TTLs to provide failover. Thankfully, we have only used it when moving phones from one server to another in preparation for upgrades, but it worked like a champ then. CP On Wed, 2008-09-10 at 15:02 +0200, Olivier wrote: Hi, I'm planning to deploy SIP hardphones in a serverless location. Phones would be connected to 2 different Asterisk servers, one backing up the other. I would like to offer resilience and I'm wondering about the best way to do it. Phones themselves can register to a backup SIP proxy if first proxy fails but, AFAIK, can't fall back to main server from backup server when main server recovers. I'm wondering if should use DNS, phone multi-registration feature, or a combination of DNS and phone multi-registration feature, to implement resilience. Your opinion ? Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail- Recorded Mesage Low Volume
Well, that was sorta my point. CP Steve Edwards wrote: A quick grep through the Asterisk (1.2.28) sources shows res_monitor using soxmix if the channel variable MONITOR_EXEC is not defined -- but nothing in app_voicemail. Am I missing something? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail- Recorded Mesage Low Volume
Hi Daniel, I'm intrigued by this and wanted to try it out - but I'm wondering how you get Asterisk to call sox at all during Voicemail()? Our server doesn't even have sox installed, so I'm not sure how to go about tricking Asterisk into running a different one. CP Daniel Hazelbaker wrote: On Apr 1, 2008, at 5:22 PM, [EMAIL PROTECTED] wrote: Can the volume of the recorded voice mail message be changed? If so, what I am doing wrong? Any input would be greatly appreciated. Thanks. I had a similar problem in our setup where we e-mail the recorded messages to e-mail retrieval. But this also helps standard phone retrieval too. What I did was edit the /usr/sbin/safe_asterisk script and add: PATH=/usr/local/bin:$PATH At the top of the script. This would let me override the default sox implementation that Asterisk uses. Then I loaded in a script (called sox) that would compress and normalize the recorded audio (It compresses to deal with the spikes of the noise of the handset being hung up, etc.). It works pretty well for us and makes the volume pretty good so we don't have to crank up the volume on our computers or phones to listen to voicemail messages. And we can't adjust the rxgain as it is already a good volume for normal calls. Daniel --CUT-- #!/bin/sh # # $1 = -v # $2 = number # $3 = inFile # $4 = outFile # REALSOX=/usr/bin/sox if [ $1 != -v ]; then $REALSOX $* exit $? fi INFILE=$3 OUTFILE=$4 # # Perform the gain adjustment. # $REALSOX $INFILE $OUTFILE compand 0.1,0.3 -60,-60,-30,-15,-20,-12,-4,-8,-2,-7 0 0 0.2 --CUT-- ___ -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Asterisk to Nortel Succession 4.0 sip...
What type of Nortel? How are you connected to the Nortel? CP Eugen Soare wrote: Well I am entering into a realm that I don't know. 3 sites with Asterisk 1 site with Nortel Asterisk/Sip calls working fine between the 3 sites. Asterisk to Nortel set calls working fine. (call comes from asterisk to nortel and rings telephone, people answer and talk happens, hangup call clears) Nortel to Asterisk. Set on Nortel gets a busy signal. Any suggestions on what to look for? Much appreciated! Eugen ___ -- 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] Polycom IP600 + PC share same switch port with VLAN
Have you set the VLAN tag on the phone? CP Lee, John (Sydney) wrote: Hi all, I have been googling and testing without any luck and would appreciate any guidance from anyone. A port has already been configured on the CISCO switch with the following: interface FastEthernet2/0/1 description VOIP VLAN 100 switchport access vlan 100 switchport mode access duplex full speed 100 I plugged the phone into the port and everything worked as far as VOIP is concerned. Then I plug a PC into the PC port of the Polycom phone with the hope that I only need one port to support 2 devices. (I wanted the VOIP phone to use VLAN 100 and PC just the native VLAN) PROBLEM: However, I found that I could not get the PC (using DHCP) to get an IP address at all. It seems to be that the traffic from the PC is also tagged as VLAN 100 as well. I was told by others that there is a setting on the Polycom phone which allows the traffic of the PC, under this type of settings, to go native. Can anyone please help? 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 ___ -- 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] Polycom IP4000 - Device does not match ACL
Try 'ip4000_1' instead of '207' for your address. CP Kevin DeGraaf wrote: I am trying to configure a Polycom IP4000 for use with Asterisk 1.4.5 on a flat local network. I followed the provisioning guides that I found on the Web, and I have the phone downloading bootrom.ld, sip.ld, and a bunch of configuration files. This all works properly. However, I receive the following error: NOTICE[27345]: chan_sip.c:14725 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for 'x.x.x.229' - Device does not match ACL I can place calls from the IP4000, but I cannot receive them: WARNING[27480]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Here are the relevant (IMHO) config sections. == sip.conf == [ip4000_1] [EMAIL PROTECTED] type=friend secret=password qualify=yes nat=no host=dynamic canreinvite=no == Polycom per-phone config on TFTP server == reg.1.displayName=207 reg.1.address=207 reg.1.label=207 reg.1.type=private reg.1.lcs= reg.1.thirdPartyName= reg.1.auth.userId=ip4000_1 reg.1.auth.password=password == Polycom company-wide config on TFTP server == server voIpProt.server.1.address=x.x.x.55/ SIP outboundProxy voIpProt.SIP.outboundProxy.address=x.x.x.55/ /SIP I've tried using x.x.x.55 as both the proxy value only, the server value only, and (in the given example) both. I also added the following to sip.conf, to no avail: deny=0.0.0.0/0.0.0.0 permit=x.x.x.0/255.255.255.0 Any ideas about what I've missed would be appreciated. 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] hi
Sounds like good security practice to me. YMMV. CP sandeep.s wrote: Hi, my sip phone is unreachable for external network(global ip) Thanks, sandeep.s ___ --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] Open Asterisk Exchange Project
Hi Michelle, We added to the bounty for this feature some time ago[1], and had a developer all lined up. He was unwilling to proceed, because Digium said that our work would never get accepted because they were already working on it. The IMAP support in 1.4 must have been it - doesn't work for us. I'm sure we'd still be interested in providing the bounty for MS Exchange VM integration if there's enough interest to get it going again. CP [1] http://www.voip-info.org/wiki/view/Asterisk+Bounty+VoiceMail-n-Email+Synchronization Michelle Dupuis wrote: Well, we can already integrate to major platforms via SMTP. The real value is in deep integration to the most popular email platform in business: Exchange. I would love to see smart Exchange integration, where deleting the VM attached email will delete the corresponding message from asterisk. My clients would eat that up. MD ___ --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 as non-root/best practices
Well, there you go then - either add /usr/sbin to your path, or provide a full path thusly: /usr/sbin/asterisk -r CP Robert McNaught wrote: not in path [EMAIL PROTECTED] echo $PATH /usr/kerberos/bin:/usr/lib/courier-imap/bin:/usr/local/bin:/bin:/usr/bin:/usr/X11R6/bin:/home/admin/bin Is /sbin in your path? CP Robert McNaught wrote: my problem is that a non-privileged user, eg admin, cannot log in and connect to the console by issuing the following [EMAIL PROTECTED] asterisk -r bash: asterisk: command not found [EMAIL PROTECTED] whereis asterisk asterisk: /usr/sbin/asterisk /usr/lib/asterisk /usr/include/asterisk /usr/include/asterisk.h /usr/share/man/man8/asterisk.8 what is the best way to solve this problem? i have tried adding admin ALL=(ALL) ALL- I will prune back once I verify I can get this working into visudo, but even that returns asterisk:command not found Does anyone out there know the best way around this - I tried adding in a symbolic link in /usr/bin/asterisk to point to the /home/asterisk/asterisk-bin/sbin/asterisk file, which worked, but is a hack around the problem and don't believe this is the way It seems that non-privileged users cannot run commands in sbin, but can in bin directories Robert ___ --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 as non-root/best practices
Is /sbin in your path? CP Robert McNaught wrote: my problem is that a non-privileged user, eg admin, cannot log in and connect to the console by issuing the following [EMAIL PROTECTED] asterisk -r bash: asterisk: command not found [EMAIL PROTECTED] whereis asterisk asterisk: /usr/sbin/asterisk /usr/lib/asterisk /usr/include/asterisk /usr/include/asterisk.h /usr/share/man/man8/asterisk.8 what is the best way to solve this problem? i have tried adding admin ALL=(ALL) ALL- I will prune back once I verify I can get this working into visudo, but even that returns asterisk:command not found Does anyone out there know the best way around this - I tried adding in a symbolic link in /usr/bin/asterisk to point to the /home/asterisk/asterisk-bin/sbin/asterisk file, which worked, but is a hack around the problem and don't believe this is the way It seems that non-privileged users cannot run commands in sbin, but can in bin directories Robert ___ --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] Voice mail Uniden UIP-200 phones
Try dtmfmode=inband CP Lyle Giese wrote: I had a working system using * 1.0 and then 1.2 and now Asterisk 1.4.13 with addons 1.4.4, zaptel 1.4.6, libpri 1.4.2. I have a mix of Grandstream (GXP2000), Uniden uip-200, Linksys Wireless G, and analog phones via Adtran chan bank. When I went to * 1.4.13, the Uniden phones stopped being able to login to voicemail. All phones are on same lan with Asterisk. I get 'Login incorrect' from Allison. I go to any other phone and I can log in just fine. Just not from our two Uniden phones. I have no problem placing calls. In the messages log, I see: app_voicemail.c: Unable to read password or app_voicemail.c:Couldn't read username Again, going to a different phone other than one of my two Uniden phones and no problem accessing and retreiving voicemail. In sip.conf against the UIP-200's I have: nat=never dtmfmode=rfc2833 Otherwise, I stayed with the standard Uniden provided config files served up via tftp and only made the minimum required changes to config files in Asterisk. I am running firmware 4.77(also tried downgrading firmware on phones to 4.63). Any suggestions? Thanks, Lyle Giese ___ --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] Caller ID Question
Disable URI dialing on your phones. CP Rob Schall wrote: I have an asterisk 1.4 setup with a PRI installed and working. We are using a Polycom 501 to test the setup.. Inbound calls work great as do phone to phone calls. However in all cases, the caller id is a bit odd. It shows: 99 sip:[EMAIL PROTECTED] what cause's this? How do I get just 99? 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 ___ --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
