[asterisk-users] AsteriskNow CDR Reports problem
Hello all, I installed yesterday the most recent AsteriskNow distro and used the built-in upgrade options to bring it up to date. My problem is that CDRs are not stored. I went through dozens of forums and sites I found on Gogle relating to that issue (very common), but I couldn't find a solution for Asterisk 1.8 (in previous versions, reinstalling the addons solved the issue). Even after compiling Asterisk 1.8.18.0, it still doesn't work. I noticed that on the FreePBX forum, people refuse to help, blaming this issue on Digium AsteriskNOW distro. I personally think they are right that it's not their responsibility, but the bottom line is that a simple fix should be available somewhere and I couldn't find it, so I assume I'm not the only person struggling with this issue. If anyone can help, kindly advise. Thanks, Michael -- _ -- 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] Google Voice Routing
Chris Datfung wrote: Hi, Hola, I'm trying to get Incoming Google Voice calls to ring on my Iaxy. I'm using Asterisk 11.0.1. Based on the the following configurations can someone help me figure out why incoming Google voice calls are not ringing on the Iaxy? Did chan_motif successfully load? If it didn't it would not attach itself to your Google account, so incoming session creation attempts would be ignored. Are there additional parts to your configuration files? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Wierd RTP issue
Richard Kenner wrote: I have a peculiar RTP issue. I'm experimenting with Jitsi as a softphone on one of my desktop Windows machines. That machine can either be connected to Asterisk via an VPN connection (with a static IP address) or not (via NAT). When it's connected via NAT, all is OK. When it's connected with VPN, the following occurs: The voice path inbound to Jitsi works fine when Jitsi originates the call, no matter what it's calling. The voice path inbound to Jitsi works fine when it's called from another SIP device. The voice path inbound to Jitsi is silent when it's called from something on the other side of a PRI via DAHDI. What's the configuration like for Jitsi in sip.conf? What version of Asterisk? What does the SIP signaling look like? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Meetme on short network
Jerry Geis wrote: I am running asterisk 1.4.43 on a really small network for testing, all on same switch. I launch a meetme between my server and 5 asterisk clients that are all on 10 foot network cables all connected to the same switch. The meetme is fine everything is in sync Then I reboot one of the clients. When it reboots I automatcially bring it back into the conference. however now its not really in sync. By not in sync do you mean that there is a delay between when the speaker speaks and when the client hears it? I'm trying to understand why that might be??? I thought it would. The conference is a listen only conference. Its not off or out of sync by much - but it is noticable. There's always going to be some amount of delay. It takes time to encode the audio, send it, mix it (in this case), receive it, decode it, and have it pass through a jitterbuffer (which by definition of being a buffer introduces delay). How much of a delay are you hearing? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Incorrect DTMF detection in Asterisk 1.8
Amit Salunkhe wrote: Hi All, Hola, I'm using 1.8 Asterisk and i havet set DTMF mode=rfc2833 in SIP global default settings. but when user sending DTMf event with SIP info method my asterisk accepting that DTMF. If default or global setting is rfc2833 then how come asterisk accepting SIP info dtmf event? what to check please guide The dtmfmode option normally controls just the sending of DTMF, but in the case of RFC2833 this overlaps some since it has to be negotiated within the SDP. For the other supported DTMF methods there is no negotiation that occurs and they are accepted at any time. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Google Voice Routing
On Mon, Nov 26, 2012 at 2:19 PM, Joshua Colp jc...@digium.com wrote: I'm trying to get Incoming Google Voice calls to ring on my Iaxy. I'm using Asterisk 11.0.1. Based on the the following configurations can someone help me figure out why incoming Google voice calls are not ringing on the Iaxy? Did chan_motif successfully load? If it didn't it would not attach itself to your Google account, so incoming session creation attempts would be ignored. Hi Joshua, How can I verify that chan_motif successfully loaded? I didn't see any errors during the build process. Are there additional parts to your configuration files? I ran make examples after I installed asterisk, so the rest of the configuration files are what ever defaults are normally created. Thanks, Chris -- _ -- 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] Google Voice Routing
Chris Datfung wrote: On Mon, Nov 26, 2012 at 2:19 PM, Joshua Colp jc...@digium.com mailto:jc...@digium.com wrote: I'm trying to get Incoming Google Voice calls to ring on my Iaxy. I'm using Asterisk 11.0.1. Based on the the following configurations can someone help me figure out why incoming Google voice calls are not ringing on the Iaxy? Did chan_motif successfully load? If it didn't it would not attach itself to your Google account, so incoming session creation attempts would be ignored. Hi Joshua, How can I verify that chan_motif successfully loaded? I didn't see any errors during the build process. You can manually load it using module load chan_motif.so and it will say if it has been loaded or the error if it could not be loaded. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Google Voice Routing
On Mon, Nov 26, 2012 at 3:53 PM, Joshua Colp jc...@digium.com wrote: Chris Datfung wrote: On Mon, Nov 26, 2012 at 2:19 PM, Joshua Colp jc...@digium.com mailto:jc...@digium.com wrote: Hi Joshua, How can I verify that chan_motif successfully loaded? I didn't see any errors during the build process. You can manually load it using module load chan_motif.so and it will say if it has been loaded or the error if it could not be loaded. Hi Joshua, I can confirm that chan_motif succesfully loaded: asterisk*CLI module load chan_motif.so Unable to load module chan_motif.so Command 'module load chan_motif.so' failed. [Nov 26 09:04:33] WARNING[28686]: loader.c:868 load_resource: Module 'chan_motif.so' already exists. I restarted Asterisk but Google Voice calls are still not forwarded to my iaxy. Any other ideas how to debug this? Thanks, Chris -- _ -- 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] No matching peer for 'callerID' from '85.xx.xx.2:5060'
Hi list, I face the following problem on incoming calls from my provider which uses Asterisk 1.6.1.25, our asterisk being 1.8.17.0. Incoming calls are not sended to the context set in provider sip.conf definition, but are going to the default context setted in [general]. Provider uses few IP's for incoming calls which are not the one used for register. Here are the logs: [2012-11-26 14:18:45] VERBOSE[1785] chan_sip.c: --- (15 headers 22 lines) --- [2012-11-26 14:18:45] VERBOSE[1785] chan_sip.c: Sending to 85.xx.xx.2:5060 (no NAT) [2012-11-26 14:18:45] VERBOSE[1785] chan_sip.c: Using INVITE request as basis request - 07403bb3412fc5206dec905b4eb26...@85.xx.xx.2 [2012-11-26 14:18:45] VERBOSE[1785] chan_sip.c: No matching peer for '0033x' from '85.xx.xx.2:5060' ... [2012-11-26 14:38:48] VERBOSE[1785] chan_sip.c: Peer audio RTP is at port 85.x.xx.2:16566 [2012-11-26 14:38:48] VERBOSE[1785] chan_sip.c: Looking for 027xxin default-guest (domain 217.yy.yy.yy) [2012-11-26 14:38:48] VERBOSE[1785] chan_sip.c: list_route: hop: sip:0033xx...@85.xx.xx.2 [2012-11-26 14:38:48] VERBOSE[1785] chan_sip.c: RDNIS for this call is 027xx (reason ) Our asterisk is registered with the provider, registerer IP from the provider being 85.xx.xx.3: Sip.conf [general] context=default-guest;where incoming calls ended ... register = 01234567:mysec...@sip.provider.net/01234567 [01234567] type=peer defaultuser=01234567 secret=mysecret host=sip.provider.net deny=0.0.0.0/0.0.0.0 permit=85.xx.xx.0/255.255.255.0 directmedia=no qualify=yes dtmfmode=rfc2833 disallow=all allow=ulaw,alaw context=from-Provider insecure=port,invite fromdomain = sip.provider.net fromuser=01234567 sendrpid = yes nat=yes What is wrong? Thanks for any hint -- Daniel -- _ -- 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] Queues and Distinctive Ring with Alert-Info
On 26/11/2012 10:14 AM, Klaverstyn, David C wrote:
Hi All,
I’m new to Queues and I have created one as follows which seems to work ok.
[david-test]
strategy = rrmemory
timeout = 10
retry = 0
maxlen = 0
announce-frequency = 0
announce-holdtime = no
member = SIP/121
member = SIP/122
member = SIP/123
I’m wondering how do you change the SipAddHeader/Alert-Info when a call
comes from a queue so users know it is a queue that is calling?
Is something like the following supposed to work?
exten = 0453451564,1,SipAddHeader(Alert-Info: n=Classic-4;w=3;c=4)
exten = 0453451564,2,Queue(david-test)
Seems to work with Asterisk 1.8.18.0.
I'm using extensions.ael and have tested the following;
400 = {
SIPAddHeader(Alert-Info: n=Classic-4;w=3;c=4);
Queue(400,inrt,,,30);
Hangup();
};
Larry.
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Re: [asterisk-users] Google Voice Routing
Chris Datfung wrote: Hi Joshua, I can confirm that chan_motif succesfully loaded: asterisk*CLI module load chan_motif.so Unable to load module chan_motif.so Command 'module load chan_motif.so' failed. [Nov 26 09:04:33] WARNING[28686]: loader.c:868 load_resource: Module 'chan_motif.so' already exists. I restarted Asterisk but Google Voice calls are still not forwarded to my iaxy. Any other ideas how to debug this? Nothing else immediately springs to mind I'm afraid. Everything looks as though it should be working and I've checked the code to make sure the session initiation is proper. I'll see if I can reproduce this over the next few days in my spare time. To others using chan_motif - are you experiencing the same issue? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] No matching peer for 'callerID' from '85.xx.xx.2:5060'
Administrator TOOTAI wrote: Hi list, Hola, I face the following problem on incoming calls from my provider which uses Asterisk 1.6.1.25, our asterisk being 1.8.17.0. Incoming calls are not sended to the context set in provider sip.conf definition, but are going to the default context setted in [general]. Provider uses few IP's for incoming calls which are not the one used for register. You will need to create separate SIP peers that match on each IP address and direct them accordingly to the correct context. A secondary option is to enable anonymous guest support, but I would not recommend that as it can pose a security risk. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Errors Compiling Libpri-1.4.13
Hi Dear List members , this is coming rather late but I took your advice and went ahead to install Dahdi before installing libpri-1.4.13 and the error messages are now different.(see attachment) Kindly help . I have tried this several times and I get stuck on Libpri installation. Your input is highly sought and appreciated. thanks Adolphus Enaboifo On Mon, Nov 19, 2012 at 8:47 PM, Adolphus Enaboifo adolphus.enabo...@osenkorp.com wrote: Good Day dear members, We are trying to test asterisk in our office to extend the reach of our present proprietary pabx system if successful. I am using an oracle virualbox 4.2.4 as the virtual server platform with ubuntu 12.04.1 server as the operating system. I get errors while trying to compile Libpri 1.4.13. (check attachment} Can you guys please help me prescribe a fix. thanks Adolphus Enaboifo gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT copy_string.o -MF .copy_string.o.d -MP -c -o copy_string.o copy_string.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pri.o -MF .pri.o.d -MP -c -o pri.o pri.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT q921.o -MF .q921.o.d -MP -c -o q921.o q921.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT prisched.o -MF .prisched.o.d -MP -c -o prisched.o prisched.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT q931.o -MF .q931.o.d -MP -c -o q931.o q931.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pri_aoc.o -MF .pri_aoc.o.d -MP -c -o pri_aoc.o pri_aoc.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pri_cc.o -MF .pri_cc.o.d -MP -c -o pri_cc.o pri_cc.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pri_facility.o -MF .pri_facility.o.d -MP -c -o pri_facility.o pri_facility.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT asn1_primitive.o -MF .asn1_primitive.o.d -MP -c -o asn1_primitive.o asn1_primitive.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose.o -MF .rose.o.d -MP -c -o rose.o rose.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_address.o -MF .rose_address.o.d -MP -c -o rose_address.o rose_address.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_etsi_aoc.o -MF .rose_etsi_aoc.o.d -MP -c -o rose_etsi_aoc.o rose_etsi_aoc.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_etsi_cc.o -MF .rose_etsi_cc.o.d -MP -c -o rose_etsi_cc.o rose_etsi_cc.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_etsi_diversion.o -MF .rose_etsi_diversion.o.d -MP -c -o rose_etsi_diversion.o rose_etsi_diversion.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_etsi_ect.o -MF .rose_etsi_ect.o.d -MP -c -o rose_etsi_ect.o rose_etsi_ect.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_etsi_mwi.o -MF .rose_etsi_mwi.o.d -MP -c -o rose_etsi_mwi.o rose_etsi_mwi.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_other.o -MF .rose_other.o.d -MP -c -o rose_other.o rose_other.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_q931.o -MF .rose_q931.o.d -MP -c -o rose_q931.o rose_q931.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_qsig_aoc.o -MF .rose_qsig_aoc.o.d -MP -c -o rose_qsig_aoc.o rose_qsig_aoc.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_qsig_cc.o -MF .rose_qsig_cc.o.d -MP -c -o rose_qsig_cc.o rose_qsig_cc.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_qsig_ct.o -MF .rose_qsig_ct.o.d -MP -c -o rose_qsig_ct.o rose_qsig_ct.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_qsig_diversion.o -MF .rose_qsig_diversion.o.d -MP -c -o rose_qsig_diversion.o rose_qsig_diversion.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_qsig_mwi.o -MF .rose_qsig_mwi.o.d -MP -c -o rose_qsig_mwi.o rose_qsig_mwi.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_qsig_name.o -MF .rose_qsig_name.o.d -MP -c -o rose_qsig_name.o rose_qsig_name.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT version.o -MF .version.o.d -MP -c -o version.o version.c ar rcs libpri.a copy_string.o pri.o q921.o prisched.o q931.o pri_aoc.o pri_cc.o pri_facility.o asn1_primitive.o rose.o rose_address.o rose_etsi_aoc.o rose_etsi_cc.o rose_etsi_diversion.o rose_etsi_ect.o rose_etsi_mwi.o rose_other.o rose_q931.o rose_qsig_aoc.o
Re: [asterisk-users] No matching peer for 'callerID' from '85.xx.xx.2:5060'
Hi, If were on this subject I'll throw in my question Does named acl lists in asterisk 11 help for this or only for registrations? Thanks, -Original Message- From: Joshua Colp jc...@digium.com Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 26 Nov 2012 10:28:05 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No matching peer for 'callerID' from '85.xx.xx.2:5060' Administrator TOOTAI wrote: Hi list, Hola, I face the following problem on incoming calls from my provider which uses Asterisk 1.6.1.25, our asterisk being 1.8.17.0. Incoming calls are not sended to the context set in provider sip.conf definition, but are going to the default context setted in [general]. Provider uses few IP's for incoming calls which are not the one used for register. You will need to create separate SIP peers that match on each IP address and direct them accordingly to the correct context. A secondary option is to enable anonymous guest support, but I would not recommend that as it can pose a security risk. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] No matching peer for 'callerID' from '85.xx.xx.2:5060'
isr...@gmail.com wrote: Hi, If were on this subject I'll throw in my question Does named acl lists in asterisk 11 help for this or only for registrations? ACLs don't control SIP peer matching, so no. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] No matching peer for 'callerID' from '85.xx.xx.2:5060'
Thought so but hoped other wise Thanks --Original Message-- From: Joshua Colp To: ? ?? To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No matching peer for 'callerID' from '85.xx.xx.2:5060' Sent: Nov 26, 2012 4:40 PM isr...@gmail.com wrote: Hi, If were on this subject I'll throw in my question Does named acl lists in asterisk 11 help for this or only for registrations? ACLs don't control SIP peer matching, so no. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Wierd RTP issue
What's the configuration like for Jitsi in sip.conf? Just fullname and md5secret plus a phones section that reads: [phones](!) type=friend host=dynamic context=SIP_Phones cc_agent_policy=generic cc_monitor_policy=generic disallow=all allow=gsm allow=ulaw allow=g729 allow=h264 What version of Asterisk? 10.7.1 What does the SIP signaling look like? I don't follow. It's just the standard INVITE/Ring/OK. -- _ -- 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] Wierd RTP issue
Richard Kenner wrote: What's the configuration like for Jitsi in sip.conf? Just fullname and md5secret plus a phones section that reads: [phones](!) type=friend host=dynamic context=SIP_Phones cc_agent_policy=generic cc_monitor_policy=generic disallow=all allow=gsm allow=ulaw allow=g729 allow=h264 What NAT settings are globally in use? Do you have directmedia turned off or on? This really does indeed feel like a weird NAT issue that is probably configuration related (probably both in Jitsi and Asterisk). -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Wierd RTP issue
What NAT settings are globally in use? nat=yes Do you have directmedia turned off or on? I've tried both ways, but I normally have it off. This really does indeed feel like a weird NAT issue that is probably configuration related (probably both in Jitsi and Asterisk). Except that: (1) It *works* when there's NAT and *doesn't* work when everything has a static IP. (2) I see the RTP packets arriving: if it were NAT, I'd expect *not* to see them. (3) It depends on the direction of the call and on whether it's SIP-SIP or DAHDI-SIP (and directmedia is off). -- _ -- 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] Wierd RTP issue
Richard Kenner wrote: What NAT settings are globally in use? nat=yes Do you have directmedia turned off or on? I've tried both ways, but I normally have it off. This really does indeed feel like a weird NAT issue that is probably configuration related (probably both in Jitsi and Asterisk). Except that: (1) It *works* when there's NAT and *doesn't* work when everything has a static IP. (2) I see the RTP packets arriving: if it were NAT, I'd expect *not* to see them. (3) It depends on the direction of the call and on whether it's SIP-SIP or DAHDI-SIP (and directmedia is off). Yeah this is so weird that packet captures are really needed. A working call and a non-working call, along with what IP ranges are what. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Wierd RTP issue
Yeah this is so weird that packet captures are really needed. A working call and a non-working call, along with what IP ranges are what. There are *tremendous* numbers of RTP packets, of course. Are those captures really going to be useful? That's the problem. If they *are* going to be useful, then how many packets should I save? I did look at the sip debug output, as I said, and those look the same. I ran into this on a machine that I won't be at for another two weeks, but I can see if I can reproduce it on similar machine. -- _ -- 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] Wierd RTP issue
Richard Kenner wrote: Yeah this is so weird that packet captures are really needed. A working call and a non-working call, along with what IP ranges are what. There are *tremendous* numbers of RTP packets, of course. Are those captures really going to be useful? That's the problem. If they *are* going to be useful, then how many packets should I save? I did look at the sip debug output, as I said, and those look the same. Not that many RTP packets are required. It's just important to see the SIP signaling and where traffic is coming/going from with the network topology in mind. That way a clearer picture of where it's saying media should go to, where it's sending media from, etc can be gleamed. Once that is figured out then the problem can be isolated. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Wierd RTP issue
Not that many RTP packets are required. It's just important to see the SIP signaling and where traffic is coming/going from with the network topology in mind. That way a clearer picture of where it's saying media should go to, where it's sending media from, etc can be gleamed. Once that is figured out then the problem can be isolated. OK, I'll try to reproduce on this machine and send that off. However, I did look at the SIP signaling and src/dst IP addresses and they're all as expected between the two calls: I really fear that the difference is in the contents of the RTP stream. -- _ -- 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] No matching peer for 'callerID' from '85.xx.xx.2:5060'
Le 26/11/2012 15:28, Joshua Colp a écrit : Administrator TOOTAI wrote: Hi list, Hola, I face the following problem on incoming calls from my provider which uses Asterisk 1.6.1.25, our asterisk being 1.8.17.0. Incoming calls are not sended to the context set in provider sip.conf definition, but are going to the default context setted in [general]. Provider uses few IP's for incoming calls which are not the one used for register. You will need to create separate SIP peers that match on each IP address and direct them accordingly to the correct context. A secondary option is to enable anonymous guest support, but I would not recommend that as it can pose a security risk. Second option was the one I used till you gave me the solution ;-) Thanks for your support -- Daniel -- _ -- 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] Wierd RTP issue
Not that many RTP packets are required. It's just important to see the SIP signaling and where traffic is coming/going from with the network topology in mind. That way a clearer picture of where it's saying media should go to, where it's sending media from, etc can be gleamed. Once that is figured out then the problem can be isolated. OK, I reproduced it on this machine. It's a total of only 1293 packets, taken on this end. First call didn't work: I heard nothing coming inbound. Second call worked, well enough that there was feedback (both phones and the desktop were in the same room). You can find the file at: http://www.gnat.com/~kenner/wierdAsteriskJitsi.pcap -- _ -- 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] Wierd RTP issue
Richard Kenner wrote: Not that many RTP packets are required. It's just important to see the SIP signaling and where traffic is coming/going from with the network topology in mind. That way a clearer picture of where it's saying media should go to, where it's sending media from, etc can be gleamed. Once that is figured out then the problem can be isolated. OK, I reproduced it on this machine. It's a total of only 1293 packets, taken on this end. First call didn't work: I heard nothing coming inbound. Second call worked, well enough that there was feedback (both phones and the desktop were in the same room). Few suggestions: 1. Remove allow=gsm from your sip.conf and reload 2. Disable ZRTP in Jitsi by going into Options - Accounts - Selecting account - Edit - Security - Uncheck Enable support to encrypt calls. See if that improves the situation. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Wierd RTP issue
1. Remove allow=gsm from your sip.conf and reload That did it! Thanks! But why should that have been an issue? 2. Disable ZRTP in Jitsi by going into Options - Accounts - Selecting account - Edit - Security - Uncheck Enable support to encrypt calls. That was one of the first things I tried a few days ago. No change. -- _ -- 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] Wierd RTP issue
Richard Kenner wrote: 1. Remove allow=gsm from your sip.conf and reload That did it! Thanks! But why should that have been an issue? The way you had things configured Asterisk was prioritizing GSM over ULAW, so until Jitsi started responding it sent GSM. This apparently upset Jitsi a little bit and caused the problem you heard (or didn't hear, hehe). If you still want to allow GSM you can try moving the allow=gsm to below allow=ulaw. This should change the priority. Glad it seems to be working for you now, though! Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Errors Compiling Libpri-1.4.13
this is coming rather late but I took your advice and went ahead to install Dahdi before installing libpri-1.4.13 and the error messages are now different.(see attachment) This is compile error is reported by newer gcc compiler versions. It is already fixed in libpri SVN. https://issues.asterisk.org/jira/browse/PRI-145 Richard -- _ -- 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] Wierd RTP issue
The way you had things configured Asterisk was prioritizing GSM over ULAW, so until Jitsi started responding it sent GSM. I thought I might have seen something like that in the packets, but it didn't look like it showed up in the SDP negotiations, so seemed peculiar to me. Unclear why this only happens with a static IP and not NAT, but oh well. -- _ -- 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] Meetme on short network
By not in sync do you mean that there is a delay between when the speaker speaks and when the client hears it? There's always going to be some amount of delay. It takes time to encode the audio, send it, mix it (in this case), receive it, decode it, and have it pass through a jitterbuffer (which by definition of being a buffer introduces delay). How much of a delay are you hearing? Josh, I am using a source file so its not speak live voice. when I say not in sync I don't care about a delay per say - its that 4 out of 5 of the clients are saying the same thing at the same time and the one I rebooted is just slightly off not identical to 1-4. So I have 5 clients where the hardware is identical, on the same switch, same length network cable, etc... I reboot one unit so even though it goes away and then rejoins the MeetMe - I would think that the server is still sending out audio at the same time as the other 1-4 units and would take the exact same amount of time to decode and all that and should be in sync with clients 1-4. again - dont care about delay - was just expecting the 1-5 units to all be in sync. Thanks Jerry -- _ -- 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] Asterisk and fwbuilder
Hi List, Until recently, I've been running an Asterisk server behind an MS ISA 2004 firewall. In general, this has worked fine - I've been able to connect to my SIP provider to make/receive calls (sipgate.co.uk in the UK and callcentric.com in the US), and DHADI runs the one traditional analogue line I have here. Then, the ISA server went pop, for the umpteenth time. Rather than replace it with yet another flaky second hand Dell server, I've put a spangly new 64-bit HP server in, which needs a 64-bit OS, hence Linux Mint. And, because I'm not entirely sure how well (if at all) ISA Server would work in a VMPlayer, I decided to use Linux's approach to firewalls, aka IPtables, using the GUI program fwbuilder. I finally got most of my network going through iptables/fwbuilder, but I cannot for the life of me make Asterisk talk SIP to the outside world. All attempts to register with sipgate fail. Callcentric appears to register OK, but attempting to make a call and it throws a critical packet not received error aborts. I have (I think) port forwarded 5060 UDP, 5060 TCP and 1-2 UDP to the Asterisk box, as well as IAX2. IAX2 works just fine, I have an external phone connected in using it. Has anyone used fwbuilder to create the rules required to let an Asterisk server make receive calls via SIP? Thanks in advance, Ade. -- _ -- 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] Wierd RTP issue
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 11/24/12 4:07 PM, Richard Kenner wrote: I have a peculiar RTP issue. I'm experimenting with Jitsi as a softphone on one of my desktop Windows machines. That machine can either be connected to Asterisk via an VPN connection (with a static IP address) or not (via NAT). When it's connected via NAT, all is OK. When it's connected with VPN, the following occurs: The voice path inbound to Jitsi works fine when Jitsi originates the call, no matter what it's calling. The voice path inbound to Jitsi works fine when it's called from another SIP device. The voice path inbound to Jitsi is silent when it's called from something on the other side of a PRI via DAHDI. I've run Wireshark on my desktop and see the RTP packets coming at the same rate and protocol (g711) in all the cases and sip set debug peer xyz doesn't shed any light on the situation (the SDP data looks similar in the working and non-worknig cases). Does anybody have any ideas what to look at next? The most common problem is that the VPN network is not declared as a localnet on Asterisk so it assumes that it has to do NAT and so replaces the external IP for communication. Make sure that your VPN segment is in sip.conf. - -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -BEGIN PGP SIGNATURE- Version: GnuPG/MacGPG2 v2.0.18 (Darwin) Comment: GPGTools - http://gpgtools.org Comment: Using GnuPG with undefined - http://www.enigmail.net/ iEYEARECAAYFAlCzsncACgkQqmNh+MyHzx7VOACdFnmfl2q1ruLAyJC3KxB2hWjL C/sAn2pBt5ltCJKCgLzEMUhSQxw8YQVL =spv9 -END PGP SIGNATURE- -- _ -- 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] Google Voice Routing
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 26/11/2012 04:26, Joshua Colp a écrit : To others using chan_motif - are you experiencing the same issue? I didn't use chan_motif since testing a few weeks ago, so I may I have broke my configuration, but Google Voice seems to be broken now. Call is received, but Asterisk does nothing: --- XMPP received from 'google-cathy' --- iq type=set to=cathy.fou...@gmail.com/asterisk-x217D1B44 id=078099D69B89C046 from=jeandenis.gir...@gmail.com/gmail.3027C461jin:jingle action=session-initiate sid=c1654741541 initiator=jeandenis.gir...@gmail.com/gmail.3027C461 xmlns:jin=urn:xmpp:jingle:1jin:content name=audio creator=initiatorrtp:description media=audio ssrc=731587560 xmlns:rtp=urn:xmpp:jingle:apps:rtp:1rtp:payload-type id=103 name=ISAC clockrate=16000/rtp:payload-type id=104 name=ISAC clockrate=32000/rtp:payload-type id=107 name=speex clockrate=16000rtp:parameter name=bitrate value=22000//rtp:payload-typertp:payload-type id=9 name=G722 clockrate=16000rtp:parameter name=bitrate value=64000//rtp:payload-typertp:payload-type id=102 name=ILBC clockrate=8000rtp:parameter name=bitrate value=13300//rtp:payload-typertp:payload-type id=108 name=speex clockrate=8000rtp:parameter name=bitrate value=11000//rtp: - --- XMPP received from 'google-cathy' --- payload-typertp:payload-type id=0 name=PCMU clockrate=8000rtp:parameter name=bitrate value=64000//rtp:payload-typertp:payload-type id=8 name=PCMA clockrate=8000rtp:parameter name=bitrate value=64000//rtp:payload-typertp:payload-type id=127 name=red clockrate=8000/rtp:payload-type id=126 name=telephone-event clockrate=8000/rtp:rtcp-mux/rtp:encryptionrtp:crypto crypto-suite=AES_CM_128_HMAC_SHA1_80 key-params=inline:t/ni1bJ62BAh0CYQgH0LebZabWx47cG7iou0/OsJ tag=1/rtp:crypto crypto-suite=AES_CM_128_HMAC_SHA1_32 key-params=inline:ysx82SVYw1H61YGmaV2d0b32zxvRBtf6PvBMlhwR tag=2//rtp:encryption/rtp:descriptionp:transport xmlns:p=http://www.google.com/transport/p2p//jin:content/jin:jingleses:session type=initiate id=c1654741541 initiator= - --- XMPP received from 'google-cathy' --- jeandenis.gir...@gmail.com/gmail.3027C461 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=103 name=ISAC clockrate=16000/pho:payload-type id=104 name=ISAC clockrate=32000/pho:payload-type id=107 name=speex bitrate=22000 clockrate=16000/pho:payload-type id=9 name=G722 bitrate=64000 clockrate=16000/pho:payload-type id=102 name=ILBC bitrate=13300 clockrate=8000/pho:payload-type id=108 name=speex bitrate=11000 clockrate=8000/pho:payload-type id=0 name=PCMU bitrate=64000 clockrate=8000/pho:payload-type id=8 name=PCMA bitrate=64000 clockrate=8000/pho:payload-type id=127 name=red clockrate=8000/pho:payload-type id=126 name=telephone-event clockrate=8000/pho:rtcp-mux/pho:src-id731587560/pho:src-idrtp:encryption xmlns:rtp= - --- XMPP received from 'google-cathy' --- urn:xmpp:jingle:apps:rtp:1rtp:crypto crypto-suite=AES_CM_128_HMAC_SHA1_80 key-params=inline:t/ni1bJ62BAh0CYQgH0LebZabWx47cG7iou0/OsJ tag=1/rtp:crypto crypto-suite=AES_CM_128_HMAC_SHA1_32 key-params=inline:ysx82SVYw1H61YGmaV2d0b32zxvRBtf6PvBMlhwR tag=2/pho:usage//rtp:encryption/pho:description/ses:session/iq - --- XMPP received from 'google-cathy' --- iq type=set to=cathy.fou...@gmail.com/asterisk-x217D1B44 id=7B548BACBF5495D3 from=jeandenis.gir...@gmail.com/gmail.3027C461jin:jingle action=session-terminate sid=c1654741541 xmlns:jin=urn:xmpp:jingle:1ses:reason xmlns:ses=http://www.google.com/session;ses:connectivity-error//ses:reasonpho:call-ended xmlns:pho=http://www.google.com/session/phone//jin:jingleses:session type=terminate id=c1654741541 initiator=jeandenis.gir...@gmail.com/gmail.3027C461 xmlns:ses=http://www.google.com/session;ses:reasonses:connectivity-error//ses:reasonpho:call-ended xmlns:pho=http://www.google.com/session/phone//ses:session/iq - Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEUEARECAAYFAlCzs60ACgkQuu7Rv+oOo/iAvQCYlWFMToLIl3CFtYLhCCpQBbZx WACeJ6xBAn1c/JU+U7kqqlvAZvPr+lk= =DOBH -END PGP SIGNATURE- -- _ -- 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] Google Voice Routing
Jean-Denis Girard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 26/11/2012 04:26, Joshua Colp a écrit : To others using chan_motif - are you experiencing the same issue? I didn't use chan_motif since testing a few weeks ago, so I may I have broke my configuration, but Google Voice seems to be broken now. stripped The signaling you've posted isn't actually from Google Voice, it's from Google Talk. While they both go through the Google XMPP server the signaling is far far different. Just right now I tested both a Gmail client calling into Asterisk and Google Voice calling into Asterisk. Both are working as expected for me. This narrows things down to the following: 1. Configuration issue as has been discussed for both of you 2. Google Talk client changes that chan_motif isn't tolerant of yet 3. Google Voice gateway changes (limited to some) that chan_motif isn't tolerant of yet It's probably #1 _ but I have nothing to immediately suggest, I'll keep thinking and looking. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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
