RE: [asterisk-users] Cisco 7970 behind NAT
Jeremiah wrote: Does anyone have this working? I have a Cisco 7970 with the 8-0-2-SR1S firmware loaded on it. I can get the phone to register with * just fine when I place my asterisk server on the same subnet and do no NAT. When I give my asterisk server a static public IP and put the phone behind a NAT to connect to the server registration fails. I turn on sip debugging and see that the phone is trying to register but it gets 401 Unauthorized. The same phone config is being used with only modifications to the IPs of the proxy and some NAT settings. I've adjusted NAT settings in two places (phone config and sip.conf). The problem is that the 7970 phones by default are listening for replies to their register requests on port 5060. Unfortunately, the phone sends them out from random ports. So, if you have nat=yes on the sip peer in asterisk then the asterisk will send the reply to the port the request came from and not 5060. The only deployment we have done of these phones with NAT involved was for 2 executives at a branch office. In order to get the phones working we had to set the XML configs for the phones to send the external IP address of the firewall (you'll need a static IP for this to work) and to request replies on a custom port other than 5060. We then gave the phones DHCP reservations so they would always get the same private IP and mapped the custom sip ports through the firewall to each of the 2 phones. The sip peers in asterisk then had nat=no. Kind of a kludge but since there were only two 7970 phones it was manageable. The other cisco phones don't seem to have this problem. Perhaps somebody out there knows a way to make the 7970 phones accept SIP responses back to the originating port. I wasted several hours but couldn't figure it out. -Evan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Can you explain why multiple registration isan important (missing) feature ?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Your definition in the sip.conf would be defining devices according to their MAC addresses. Your dial plan would call these devices based on extensions. exten = 100,1,Dial(SIP/MAC) ; where MAC is the MAC address of the phone All right. Then I give to my girlfriend my number 1234567 and she calls me in, how will I know to which MAC address I need to pass call? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: codecs/voicemail/DTMF
On 2006-09-20 10:23:01 -0700, Mr. Jones [EMAIL PROTECTED] said: Hi Eric, I'm confused on where I would put this? I'm also confused on how this would help with external calls (which we want to be g729) vs internal calls to voicemail (which appear to need to be g711)? No, calls to voicemail do not need to be ulaw. You can definitely call voicemail via G729 and use rfc2833 for DTMF. It works depending on your equipment. You are calling using G729 and trying to pass your tones inband, which is impossible due to lack of bandwidth. I think using DTMF=rfc2833 instead of auto is your best bet. Good Luck, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Codecs
This is what I did, i installed an the open source codec. but legally, your supposed to buy a license.1) download 723 and 729 codechttp://kvin.lv/pub/Linux/Asterisk/2) copy both codecs to /usr/lib/asterisk/modules3) restart asterisk/etc/init.d/asterisk restart4) verify its working. as long as 723 and 729 display numbers and not S, it's workingasterisk1*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - 2 2 2 3 2 1 3 8 - 12 gsm 8 - 2 2 3 2 1 3 8 - 12 ulaw 8 2 - 1 3 2 1 3 8 - 12 alaw 8 2 1 - 3 2 1 3 8 - 12 g726 9 3 3 3 - 3 2 4 9 - 13 adpcm 8 2 2 2 3 - 1 3 8 - 12 slin 7 1 1 1 2 1 - 2 7 - 11 lpc10 8 2 2 2 3 2 1 - 8 - 12 g729 8 2 2 2 3 2 1 3 - - 12 speex - - - - - - - - - - - ilbc 8 2 2 2 3 2 1 3 8 - -5) allow the codec in sip.conf under [general]disallow=allallow=g729[EMAIL PROTECTED] wrote: Good Day,I am new to Asterisk and I need help in configuring codecs in Asterisk. I also will need to buy the license for G 729 codec and need to put that key in Asterisk. Can someone please provide the step by step instructions for the codec configurations?Also I am having problems forwarding calls to landline or cellular numbers. As soon as the preson on the forwarded end picks up the call drops - what can cause this and what is the solution?ThanksWyatt___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help in Reloading of Asterisk...
Hi Users,I need help or clues from U, please help me...I'm new Asterisk, I want to do the Asterisk in RealTime ConfiguringMy problem is below one . After every change to the database, the asterisk will need to be reloaded.How to Reload the Asterisk server.Now its simple , stupid doubt to put in mail-list...I don't know it. -- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Uninstalling Trixbox
Thanx all for ur concern...wish me good luck..im starting to do something with it right now On 9/21/06, Leo Ann Boon [EMAIL PROTECTED] wrote: Yeah thats your problem I guess. Using a tarball 'package' certainly does not give you an easy way out or un-install. Best to start again or get your toothpick out :)Or you can try this from / (your root directory) :)rm -rf `tar tzf xxx.tar.gz`Note the backquotes enclosing the tar statement It will remove everything that you installed from the tar ball.Leo___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- RegardsRizwan HishamSoftware Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help in Reloading of Asterisk...
if you are using asterisk realtime no need to reload. If you want to reload type command asterisk -rvvv then you type reloadOn 9/21/06, raviprakash sunkara [EMAIL PROTECTED] wrote: Hi Users,I need help or clues from U, please help me...I'm new Asterisk, I want to do the Asterisk in RealTime ConfiguringMy problem is below one . After every change to the database, the asterisk will need to be reloaded.How to Reload the Asterisk server.Now its simple , stupid doubt to put in mail-list...I don't know it. -- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How much SIP calls can I squeeze from this box
Hi lists, I would like to know how much can i get from the below configuration. I have a machine in my office that I want to use for demo purpose. The features I want to implement are: voicemail (users call the box to get their messages) voicemail to email (some users will the the vm by email) pbx like behavior (music on hold, a simple IVR to select what department to talk to) Full 100% call recording. Software spec: Centos 4.4 Asterisk 1.2.12.1 no sql SIP users with IP hardphones running g711 Hardware: Asterisk Box: Dual core Pentium D at 2.4ghz, 533fsb, Intel 945GNT board,100Mbit intel NIC. Dual 80gbit sata2 disk. A 8-port fxs card (pci in a PCI-X slot) and the FXS will be connected to a Panasonic PBX Protocol: G711 all the way if possible (even in moh) SIP users?: Here it comes my question in terms of: - Registered users - Simultaneous calls (remember full call recording) BTW: What options do I have to minimize disk writes for the call recording part? more ram to make it as a ramdisk? special ramdisk cards? any special format or way to capture/encode/store the recorded stream? During night hours I was thinking of moving the recorded files to another server via NFS. thanks in advance. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTCP and RTP packetization in 1.4
Hi all, I'm so excited about 1.4 coming out soon :) , I was wondering if anyone can comment on the following: 1. Will RTP packetization (5162) committed to trunk (43243) be in 1.4? I have it running here for a while, and its really working well. I have used the patch for 1.2.10 2. Will RTCP (2863) committed to trunk (32230) be in 1.4? There is only a patch for 1.2.4, have used that, but will there be an updated patch. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTCP and RTP packetization in 1.4
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 yusuf wrote: Hi all, I'm so excited about 1.4 coming out soon :) , I was wondering if anyone can comment on the following: 1. Will RTP packetization (5162) committed to trunk (43243) be in 1.4? I have it running here for a while, and its really working well. I have used the patch for 1.2.10 2. Will RTCP (2863) committed to trunk (32230) be in 1.4? There is only a patch for 1.2.4, have used that, but will there be an updated patch. Trunk is the code that will become 1.4 :) So if something has been committed to trunk, then yes, it will be in 1.4. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFEkLQS6d5vy0jeVcRAgu2AJ48xpw8sHHnr1/m0qC/jPoJQQlBngCdEV5W dyvf0jSiY4bkGivb1D4yiqU= =yY0i -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTCP and RTP packetization in 1.4
Matt Riddell (IT) wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 yusuf wrote: Hi all, I'm so excited about 1.4 coming out soon :) , I was wondering if anyone can comment on the following: 1. Will RTP packetization (5162) committed to trunk (43243) be in 1.4? I have it running here for a while, and its really working well. I have used the patch for 1.2.10 2. Will RTCP (2863) committed to trunk (32230) be in 1.4? There is only a patch for 1.2.4, have used that, but will there be an updated patch. Trunk is the code that will become 1.4 :) So if something has been committed to trunk, then yes, it will be in 1.4. - -- Cheers, Matt Riddell ___ Hi, That is usually the case, however, there is a feature freeze some time before stable releases, and since RTP packetization was only committed to trunk on 09-18-06, does that mean that it wont be in 1.4, maybe only 1.4.1 or 1.4.2. Or am I completely wrong (I hope I am :) ) -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTCP and RTP packetization in 1.4
Dose this trunk do just like IAX2 trunk, to reduce bandwidth?___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTCP and RTP packetization in 1.4
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 yusuf wrote: That is usually the case, however, there is a feature freeze some time before stable releases, and since RTP packetization was only committed to trunk on 09-18-06, does that mean that it wont be in 1.4, maybe only 1.4.1 or 1.4.2. Or am I completely wrong (I hope I am :) ) The freeze was to stop things being committed. So if it has been committed, it got through the freeze! :) Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFEkgFS6d5vy0jeVcRAtKjAJ9v9FOrZUj93J/nz+c6JT1bmrGtrgCfVz/j lOdcA02sXLPAQlCxydgfYb0= =c02e -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unexpected delay: problem with outgoing calls
Hi to all. I've registred my Asterisk 1.2.12.1 to a VoIP Service Provider and I've some problem with outgoing calls: there is a big delay for bidirectional audio flow. Here is mean part of an asterisk trace releted to outgoing calls. (canreinvite=no for both peers). Until SIP 180 ringing signaling is correct...bold highlight time for NOTICE _ _ _ _ Sep 18 16:01:43 [1;33;40mNOTICE[0;37;40m[23098]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m9854[0;37;40m [1;37;40mhandle_response_register[0;37;40m: Outbound Registration: Expiry for 10.28.52.74 is 3599 sec (Scheduling reregistration in 3584 s) [1;30;40m -- [0;37;40mSIP/outgoing-08197388 is ringing Transmitting (no NAT) to 10.28.52.244:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.28.52.244;branch=z9hG4bKc39fdf160cf737ac;received=10.28.52.244 From: sip:[EMAIL PROTECTED];user=phone;tag=a82e9be13c882482 To: sip:[EMAIL PROTECTED];user=phone;tag=as2ea0ddd1 Call-ID: [EMAIL PROTECTED] CSeq: 829 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- [1;30;40m -- [0;37;40mSIP/outgoing-08197388 is making progress passing it to SIP/bt102-08190d90 Sep 18 16:02:37 [1;33;40mNOTICE[0;37;40m[23098]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m11613[0;37;40m [1;37;40msip_poke_noanswer[0;37;40m: Peer 'outgoing' is now UNREACHABLE! Last qualify: 4 -- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- -- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- -- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- -- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- -- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- [1;30;40m -- [0;37;40mSIP/outgoing-08197388 answered SIP/bt102-08190d90 We're at 10.28.52.246 port 16274 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 10.28.52.244:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.28.52.244;branch=z9hG4bKc39fdf160cf737ac;received=10.28.52.244 From: sip:[EMAIL PROTECTED];user=phone;tag=a82e9be13c882482 To: sip:[EMAIL PROTECTED];user=phone;tag=as2ea0ddd1 Call-ID: [EMAIL PROTECTED] CSeq: 829 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 184 v=0 o=root 23109 23110 IN IP4 10.28.52.246 s=session c=IN IP4 10.28.52.246 t=0 0 m=audio 16274 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- [1;30;40m -- [0;37;40mAttempting native bridge of SIP/bt102-08190d90 and SIP/outgoing-08197388 Sep 18 16:03:25 [1;33;40mNOTICE[0;37;40m[23098]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m9882[0;37;40m [1;37;40mhandle_response_peerpoke[0;37;40m: Peer 'outgoing' is now REACHABLE! (6ms / 2000ms) -- SIP read from 10.28.52.244:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.28.52.244;branch=z9hG4bK30db550457acdb99 From: sip:[EMAIL PROTECTED];user=phone;tag=a82e9be13c882482 To: sip:06720228.52.246;user=phone;tag=as2ea0ddd1 Contact: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 829 ACK User-Agent: Grandstream BT110 1.0.8.12 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 _ _ _ _ From trace it points out that time gap from 180 Ringing and follow 200 Ok is about 1 minute.. and so from 200 OK and ACK Any suggestions? Moreover..when I attempt to make an outgoing call with option canreinvite=yes, Asterisk notifies the follow message? Sep 20 14:13:42 WARNING[2373]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x819b240', 10 retries! Can anyone tell me what it does mean and how to fix it? Thanks 4 all -- * (o ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * -- * (o ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTCP and RTP packetization in 1.4
Ma Zhiyong wrote: Dose this trunk do just like IAX2 trunk, to reduce bandwidth? RTP packetization is for RTP based channels, like SIP, and it reduces badwidth by putting multiple frames in one packet, so you save *ALOT* on packet headers, and it actually is more efficient. This in not what a trunk in IAX2 is, which is a multiplexed trunk, putting multiple calls in one 'trunk' So with packetization, if packetization=10 and g729 , your 1 packet contain 100ms of audio, instead of 10 packets containing 10ms audio each. However, dont set tooo high level of packetization, as you will be introducing delay. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Analog Modem through ISDN E1 Primay Line
Hi all, I have next scenario: One E1 Line connected to my asterisk through TE412P. One analog modem connected to an PAP2. I want to make internet conecction with this modem through E1 Primary Line but I obtain more thatn 95% of errors in connections. Must I do anything to permit data connection with an analog modem through E1 ISDN Primary Line? Regards, Tron -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de flavio Enviado el: jueves, 21 de septiembre de 2006 10:26 Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] Unexpected delay: problem with outgoing calls Hi to all. I've registred my Asterisk 1.2.12.1 to a VoIP Service Provider and I've some problem with outgoing calls: there is a big delay for bidirectional audio flow. Here is mean part of an asterisk trace releted to outgoing calls. (canreinvite=no for both peers). Until SIP 180 ringing signaling is correct...bold highlight time for NOTICE _ _ _ _ Sep 18 16:01:43 [1;33;40mNOTICE[0;37;40m[23098]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m9854[0;37;40m [1;37;40mhandle_response_register[0;37;40m: Outbound Registration: Expiry for 10.28.52.74 is 3599 sec (Scheduling reregistration in 3584 s) [1;30;40m -- [0;37;40mSIP/outgoing-08197388 is ringing Transmitting (no NAT) to 10.28.52.244:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.28.52.244;branch=z9hG4bKc39fdf160cf737ac;received=10.28.52.244 From: sip:[EMAIL PROTECTED];user=phone;tag=a82e9be13c882482 To: sip:[EMAIL PROTECTED];user=phone;tag=as2ea0ddd1 Call-ID: [EMAIL PROTECTED] CSeq: 829 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- [1;30;40m -- [0;37;40mSIP/outgoing-08197388 is making progress passing it to SIP/bt102-08190d90 Sep 18 16:02:37 [1;33;40mNOTICE[0;37;40m[23098]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m11613[0;37;40m [1;37;40msip_poke_noanswer[0;37;40m: Peer 'outgoing' is now UNREACHABLE! Last qualify: 4 -- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- -- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- -- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- -- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- -- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- [1;30;40m -- [0;37;40mSIP/outgoing-08197388 answered SIP/bt102-08190d90 We're at 10.28.52.246 port 16274 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 10.28.52.244:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.28.52.244;branch=z9hG4bKc39fdf160cf737ac;received=10.28.52.244 From: sip:[EMAIL PROTECTED];user=phone;tag=a82e9be13c882482 To: sip:[EMAIL PROTECTED];user=phone;tag=as2ea0ddd1 Call-ID: [EMAIL PROTECTED] CSeq: 829 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 184 v=0 o=root 23109 23110 IN IP4 10.28.52.246 s=session c=IN IP4 10.28.52.246 t=0 0 m=audio 16274 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- [1;30;40m -- [0;37;40mAttempting native bridge of SIP/bt102-08190d90 and SIP/outgoing-08197388 Sep 18 16:03:25 [1;33;40mNOTICE[0;37;40m[23098]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m9882[0;37;40m [1;37;40mhandle_response_peerpoke[0;37;40m: Peer 'outgoing' is now REACHABLE! (6ms / 2000ms) -- SIP read from 10.28.52.244:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.28.52.244;branch=z9hG4bK30db550457acdb99 From: sip:[EMAIL PROTECTED];user=phone;tag=a82e9be13c882482 To: sip:06720228.52.246;user=phone;tag=as2ea0ddd1 Contact: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 829 ACK User-Agent: Grandstream BT110 1.0.8.12 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 _ _ _ _ From trace it points out that time gap from 180 Ringing and follow 200 Ok is about 1 minute.. and so from 200 OK and ACK Any suggestions? Moreover..when I attempt to make an outgoing call with option canreinvite=yes, Asterisk notifies the follow message? Sep 20 14:13:42 WARNING[2373]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x819b240', 10 retries! Can anyone tell me what it does mean and how to fix it? Thanks 4 all -- * (o ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * -- * (o ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Iax Netstat Output
HiI've * running but I'm other side voice is not so clear and delay. this is my iax netstat output can someone help me where is the problem.here is the iax netstat output Channel RTT Jit Del Lost Drop OOO Kpkts Jit Del Lost Drop OOO Kpkts 3 Traffic from Server to Agent 4 IAX2/2003-29 6 -1 0 -1 0 -1 228 18 89 1294 0 127 226 5 IAX2/2006-18 5 -1 0 -1 0 -1 83 18 87 934 0 1 826 IAX2/2021-11 11 -1 0 -1 0 -1 30 20 77 158 0 1 28 7 IAX2/2021-12 9 -1 0 -1 0 -1 45 20 80 167 0 0 448 IAX2/2021-31 8 -1 0 -1 0 -1 74 18 91 429 0 4 73 9 IAX2/2022-13 6 -1 0 -1 0 -1 123 17 94 740 0 1 12110 IAX2/2023-6 11 -1 0 -1 0 -1 229 19 97 2114 0 46 226 11 IAX2/2024-49 9 -1 0 -1 0 -1 45 18 76 202 0 1 4412 13 Traffic from Server to Minutes Provider 14 IAX2/callaus-15 1000 -1 0 -1 0 -1 0 0 0 0 0 0 0 15 IAX2/callaus-30 1000 -1 0 -1 0 -1 0 0 0 0 0 0 016 IAX2/callaus-34 259 -1 0 -1 0 -1 5 0 40 0 0 0 0 17 IAX2/callaus-4 502 -1 0 -1 0 -1 1 0 40 0 0 0 018 IAX2/callaus-40 260 -1 0 -1 0 -1 2 0 40 0 0 0 0 19 IAX2/callaus-5 1000 -1 0 -1 0 -1 0 0 0 0 0 0 020 IAX2/callaus-7 259 -1 0 -1 0 -1 1 0 40 0 0 0 0 21 IAX2/velilevox-19 256 -1 0 -1 0 -1 14 0 40 0 0 0 0thankarun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTCP and RTP packetization in 1.4
Okay, Thank you. So packetization is a feature of RTP and can work with all of the codecs, isn't it? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iax Netstat Output
Arun Kumar wrote: Hi I've * running but I'm other side voice is not so clear and delay. this is my iax netstat output can someone help me where is the problem. here is the iax netstat output Channel RTT Jit Del Lost DropOOO Kpkts Jit Del LostDrop OOO Kpkts 3 Traffic from Server to Agent 4 IAX2/2003-296 -1 0 -1 0 -1 228 18 89 12940 127 226 5 IAX2/2006-185 -1 0 -1 0 -1 83 18 87 934 0 1 82 6 IAX2/2021-1111 -1 0 -1 0 -1 30 20 77 158 0 1 28 7 IAX2/2021-129 -1 0 -1 0 -1 45 20 80 167 0 0 44 8 IAX2/2021-318 -1 0 -1 0 -1 74 18 91 429 0 4 73 9 IAX2/2022-136 -1 0 -1 0 -1 123 17 94 740 0 1 121 10 IAX2/2023-6 11 -1 0 -1 0 -1 229 19 97 21140 46 226 11 IAX2/2024-499 -1 0 -1 0 -1 45 18 76 202 0 1 44 12 13 Traffic from Server to Minutes Provider 14 IAX2/callaus-15 1000-1 0 -1 0 -1 0 0 0 0 0 0 0 15 IAX2/callaus-30 1000-1 0 -1 0 -1 0 0 0 0 0 0 0 16 IAX2/callaus-34 259 -1 0 -1 0 -1 5 0 40 0 0 0 0 17 IAX2/callaus-4 502 -1 0 -1 0 -1 1 0 40 0 0 0 0 18 IAX2/callaus-40 260 -1 0 -1 0 -1 2 0 40 0 0 0 0 19 IAX2/callaus-5 1000-1 0 -1 0 -1 0 0 0 0 0 0 0 20 IAX2/callaus-7 259 -1 0 -1 0 -1 1 0 40 0 0 0 0 21 IAX2/velilevox-19 256 -1 0 -1 0 -1 14 0 40 0 0 0 0 thank arun -- I have had this same sort of disproportionate stats, where there is huge delay, packet loss and Out Of Order packets only on side, and the other side is fine. So the users hanging off the agent will not be able to hear the other side. One flow seems uneven compared to the other. I dont have a solution, but try playing with your jitterbuffer setttings, and make sure the network is fine, is ping times and the like equal going from the one to the other, and back. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iax Netstat Output
I know what, if I use ZAP-IAX2 ---IAX2, I also got one direction poor. But if I use SIP-IAX2 ---IAX2-, every think is OK. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Habitual set of number
Good afternoon. For an output in city I use such construction:
exten = 9,1,Answer
exten = 9,2,SIPDtmfMode(rfc2833)
exten = 9,3, Set(TIMEOUT(digit)=3)
exten = 9,4,ChanIsAvail(ZAP/g2|j)
exten = 9,5,NoOp(${AVAILCHAN})
exten = 9,6,Playtones(dial)
exten = 9,7,Cut(chan=AVAILCHAN,-,1)
exten = 9,8,NoOp(${chan})
exten = 9,9,waitexten()
exten = _XX,1,Dial(${chan}/${EXTEN},,tT)
exten = _XX,2,Hangup
exten = _XXX,1,Dial(${chan}/${EXTEN},,tT)
exten = _XXX,2,Hangup
exten = 9,105,Playtones(busy)
exten = 9,106,Busy(10)
Like all it is quite good, except for one, hooter goes in a tube at
typing, at that time, while it hammers in number in waitexten. How it
is possible to realize too most, only that with a set of the first
figure hooter interrupted? In advance thanks.
___
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Re: [asterisk-users] Iax Netstat Output
no zap - iax2 - iax2 only iax2 - iax2 - iax2thanksOn 9/21/06, Ma Zhiyong [EMAIL PROTECTED] wrote:I know what, if I use ZAP-IAX2 ---IAX2, I also got one direction poor. But if I use SIP-IAX2 ---IAX2-, every think is OK. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk skills in the philippines
hi all,my apologies for posting it here in a technical mailing list. i need some info on companies that support asterisk deployment in the Philippines. Please send me a note offline.thanks Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk / chan_capi problems
Hi! I have problems with an Asterisk box which was running fine for some time but now causes problems (asterisk restarts, hangs ...). I use asterisk 1.2.7.1 with chan_capi-cm-0.6.5 and divas4linux-melware-3.0.f-106.622-1 In syslog I see lots of the following messages: Sep 20 07:02:41 ast01 kernel: kcapi: appl 1 ncci 0x130104 down Sep 20 07:02:43 ast01 kernel: kcapi: appl 1 ncci 0x60202 up Sep 20 07:02:43 ast01 kernel: kcapi: appl 1 ncci 0x10204 up Sep 20 07:03:12 ast01 kernel: kcapi: appl 1 ncci 0x10204 down Sep 20 07:03:12 ast01 kernel: kcapi: appl 1 ncci 0x60202 down Sep 20 07:06:25 ast01 kernel: kcapi: appl 1 ncci 0x20104 up Sep 20 07:06:40 ast01 kernel: kcapi: appl 1 ncci 0x20104 down Sep 20 07:07:13 ast01 kernel: kcapi: appl 1 ncci 0x70302 up What are the cause of this messages? May they be related with the asterisk crashes ? thanks klaus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Invite issues
Hi all, I am receiving handle_request_invite: Failed to authenticate errors on two VoIP gateway devices connected to my asterisks SIP server. The problem seems to be in my configuration. I will only focus on one of this devices in this mail. On this device I receive the error Sep 21 12:02:23 NOTICE[30782]: chan_sip.c:10468 handle_request_invite: Failed to authenticate user 2006 sip:[EMAIL PROTECTED]:5060;tag=43c7d6ab at the CIL This is the configuration for 2006 in my sip.conf [2006] type=friend username=2006 secret=2006 context=from-sip callerid=Allan 2006 host=dynamic defaultip=192.168.0.100 ;nat=no canreinvite=yes dtmfmode=RFC2833 [12] insecure=very canreinvite=yes type=friend username=12 secret=12 context=from-gsm callerid=Allan 12 host=dynamic ;defaultip=192.168.0.100 dtmfmode=rfc2833 ;register=:@192.168.0.10 ; Local interface ;qualify=no Attached kindly find the SIP communication captured between the device and Asterisk. Allan. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com No. TimeSourceDestination Protocol Info 8 7.567552192.168.0.100 192.168.0.2 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description Frame 8 (860 bytes on wire, 860 bytes captured) Arrival Time: Sep 21, 2006 09:39:48.143706000 Time delta from previous packet: 4.270322000 seconds Time since reference or first frame: 7.567552000 seconds Frame Number: 8 Packet Length: 860 bytes Capture Length: 860 bytes Protocols in frame: eth:ip:udp:sip:sdp Coloring Rule Name: UDP Coloring Rule String: udp Ethernet II, Src: PortechC_00:01:8c (00:03:7e:00:01:8c), Dst: Zioncom_f1:de:eb (00:0e:e8:f1:de:eb) Destination: Zioncom_f1:de:eb (00:0e:e8:f1:de:eb) Address: Zioncom_f1:de:eb (00:0e:e8:f1:de:eb) ...0 = Multicast: This is a UNICAST frame ..0. = Locally Administrated Address: This is a FACTORY DEFAULT address Source: PortechC_00:01:8c (00:03:7e:00:01:8c) Address: PortechC_00:01:8c (00:03:7e:00:01:8c) ...0 = Multicast: This is a UNICAST frame ..0. = Locally Administrated Address: This is a FACTORY DEFAULT address Type: IP (0x0800) Internet Protocol, Src: 192.168.0.100 (192.168.0.100), Dst: 192.168.0.2 (192.168.0.2) Version: 4 Header length: 20 bytes Differentiated Services Field: 0xa0 (DSCP 0x28: Class Selector 5; ECN: 0x00) 1010 00.. = Differentiated Services Codepoint: Class Selector 5 (0x28) ..0. = ECN-Capable Transport (ECT): 0 ...0 = ECN-CE: 0 Total Length: 846 Identification: 0x4111 (16657) Flags: 0x00 0... = Reserved bit: Not set .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 61 Protocol: UDP (0x11) Header checksum: 0xb737 [correct] Good: True Bad : False Source: 192.168.0.100 (192.168.0.100) Destination: 192.168.0.2 (192.168.0.2) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 826 Checksum: 0x2331 [correct] Session Initiation Protocol Request-Line: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Method: INVITE Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.0.100:5060;rport;branch=z9hG4bK63d97bbf1956f9ad23a40b1dcb898fcc From: 2006 sip:[EMAIL PROTECTED]:5060;tag=74f33ef1 SIP Display info: 2006 SIP from address: sip:[EMAIL PROTECTED]:5060 SIP tag: 74f33ef1 To: sip:[EMAIL PROTECTED] SIP to address: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 Contact Binding: sip:[EMAIL PROTECTED]:5060 URI: sip:[EMAIL PROTECTED]:5060 SIP contact address: sip:[EMAIL PROTECTED]:5060 CSeq: 801 INVITE Max-Forwards: 70 Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp User-Agent: CMI CM5K Content-Length: 329 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 56901 0 IN IP4 192.168.0.100 Owner Username: - Session ID: 56901 Session Version: 0 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 192.168.0.100 Session Name (s): SIP CALL Connection Information (c): IN IP4 192.168.0.100 Connection Network Type: IN Connection Address Type: IP4
Re: [asterisk-users] asterisk / chan_capi problems
On Thu, 21 Sep 2006, Klaus Darilion wrote: Hi! I have problems with an Asterisk box which was running fine for some time but now causes problems (asterisk restarts, hangs ...). I use asterisk 1.2.7.1 with chan_capi-cm-0.6.5 and divas4linux-melware-3.0.f-106.622-1 In syslog I see lots of the following messages: Sep 20 07:02:41 ast01 kernel: kcapi: appl 1 ncci 0x130104 down Sep 20 07:02:43 ast01 kernel: kcapi: appl 1 ncci 0x60202 up Sep 20 07:02:43 ast01 kernel: kcapi: appl 1 ncci 0x10204 up Sep 20 07:03:12 ast01 kernel: kcapi: appl 1 ncci 0x10204 down Sep 20 07:03:12 ast01 kernel: kcapi: appl 1 ncci 0x60202 down Sep 20 07:06:25 ast01 kernel: kcapi: appl 1 ncci 0x20104 up Sep 20 07:06:40 ast01 kernel: kcapi: appl 1 ncci 0x20104 down Sep 20 07:07:13 ast01 kernel: kcapi: appl 1 ncci 0x70302 up What are the cause of this messages? May they be related with the asterisk crashes ? No, they are not related. These messages are just info messages from common kernelcapi driver about b-channel up: ncci 0x up b-channel down: ncci 0x down For the problems you have, some logs would be needed. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] HINT problems with SVN-trunk-r43322
The reason is that, at least in the SIP channel in trunk, the structure that keeps track of device state for hinting only gets allocated on peer objects and then only if call-limit is configured to some value. It's been a long time since I've done any development with 1.2 (all my 1.2 systems are waiting for 1.4 to come out so we can add a bunch of features), so I forget how that works there. Rumor has it these restrictions aren't necessary, but I forget. If by '6 months' you mean trunk from that long ago, it's entirely plausible that you got a snapshot during the evolution from where it was in 1.2 to where it is today. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Wednesday, September 20, 2006 10:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] HINT problems with SVN-trunk-r43322 Group Looks like the type=peer call-limit=2 Works. Now the question is why? The sample I sent is working on a system build 6 months ago. Will do some more checking and will report to the list on anything I find... Thanks Bradley for this bit of info you gave!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Wednesday, September 20, 2006 1:36 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] HINT problems with SVN-trunk-r43322 On Wednesday 20 September 2006 12:31, Watkins, Bradley wrote: You will need to change the type=friend to type=peer and also define call-limit to some value (it can be large if you don't care about the actual limit). That should fix hints for you. But if you have it set to 1 then busy status won't work, isn't that the case? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Two phones, same number
I would like to have two phones acting like one phone. Explaining it is a bit difficult, even though what I want is really simple. Assume that I have two phones, a wireless and a wired phone. Both are in the Dial(), so whenever a call comes in, both of the phones ring. If I happen to be at the desk, I pick up the wired phone and talk. If a second call comes in, the wireless phone will be ringing even though I'm busy on the wired phone. I would like the second caller to get a busy tone. I have considered various ways to solve this. One is to make a queue, and only allow one caller in the queue. As far as I can see this won't work, at least not when I am busy because I did an outgoing call. Another way is to use GROUP() to put the calls in a separate group, and return busy when GROUP_COUNT 0. Unfortunately I am already using the GROUP() functionality for something different on those calls -- and it seems a call can't be in two GROUP()'s simultaneously. Ideas welcome... /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk, iaxmodem, hylafax quality problem
Hello, My setup is PRI card, Asterisk, iaxmodem, hylafax or PRI card, Asterisk, channel bank, fax machine. I'm using Fedora Core 4, iaxmodem 0.1.14, hylafax 4.3.0, asterisk 1.2.10. Everything is fine when caller use ECM but when ECM isn't in use I often got unusable incoming faxes (much often that it should be). But when I switch back to fax machine that receive faxes perfectly (at least no visible error). Where should be the problem? Is there any solution for improving quality? Any tuning in Asterisk or Hylafax? As I see some people use slinear for iaxmodem and some user use alaw. Which is better? What config should I post if that needed for ideas? bye, Zsolt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] HINT problems with SVN-trunk-r43322
Brad Thanks for your insight. The info I used to set this up before was from Grandstream http://www.grandstream.com/FAQ/FAQ_and_Example_for_Asterisk_Configuratio n_for_GXP-2000.pdf I will also notify them about the error in the above document. Thanks again!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Watkins, Bradley Sent: Thursday, September 21, 2006 6:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] HINT problems with SVN-trunk-r43322 The reason is that, at least in the SIP channel in trunk, the structure that keeps track of device state for hinting only gets allocated on peer objects and then only if call-limit is configured to some value. It's been a long time since I've done any development with 1.2 (all my 1.2 systems are waiting for 1.4 to come out so we can add a bunch of features), so I forget how that works there. Rumor has it these restrictions aren't necessary, but I forget. If by '6 months' you mean trunk from that long ago, it's entirely plausible that you got a snapshot during the evolution from where it was in 1.2 to where it is today. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Wednesday, September 20, 2006 10:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] HINT problems with SVN-trunk-r43322 Group Looks like the type=peer call-limit=2 Works. Now the question is why? The sample I sent is working on a system build 6 months ago. Will do some more checking and will report to the list on anything I find... Thanks Bradley for this bit of info you gave!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Wednesday, September 20, 2006 1:36 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] HINT problems with SVN-trunk-r43322 On Wednesday 20 September 2006 12:31, Watkins, Bradley wrote: You will need to change the type=friend to type=peer and also define call-limit to some value (it can be large if you don't care about the actual limit). That should fix hints for you. But if you have it set to 1 then busy status won't work, isn't that the case? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] enumlookup - deprecated working - but appreciated one duznt :-(
Got this working.
It seems, ENUMLOOKUP needs arguments, seperated by '|' , instead of ','
as documented.
Check out
http://www.voip-info.org/wiki/view/Asterisk+func+enumlookup
for the tiny patch.
Another observation, the cmd EnumLookup, duz search thru different
domains listed in enum.conf, but the function ENUMLOOKUP doesn't(it just
searches for e164.arpa, and if not found, gives up, if the zone argument
is left empty).
Anyone worked around this one??
cheerz
Ben.
Benjamin Jacob wrote:
Hello ppl,
I had appdata set to use the function ENUMLOOKUP. But it gets me nothing.
| id| context | exten | priority | app
|
appdata
==
48 | pbx1| _011. | 1|
Set |
enumresult=${ENUMLOOKUP(+13015611020,sip,c,enum.info)}
| 49 | pbx1| _011. | 2| SayDigits |
${enumresult}
But, using the application, EnumLookup, I do get back the results.
| 48 | pbx1| _011. | 1| EnumLookup |
+13015611020| 49 | pbx1|
_011. | 2| Dial |
${ENUM}
Another interesting observation, in my enum.conf, I've set only search
= enum.info .
In the tcpdump, I see EnumLookup, the deprecated one looking for the
correct enum.info, but, with the function ENUMLOOKUP, I see
enum.arpa being pinged!!???
Any ideas where I am going wrong?
My enum.info pasted :
===
;
; ENUM Configuration for resolving phone numbers over DNS
;
; Sample config for Asterisk
; This file is reloaded at reload enum in the CLI
;
[general]
;
; The search list for domains may be customized. Domains are searched
; in the order they are listed here.
;
;search = e164.arpa
;
; If you'd like to use the E.164.org public ENUM registery in addition
; to the official e164.arpa one, uncomment the following line
;
;search = e164.org
search = e164.info
;
; As there are more H323 drivers available you have to select to which
; drive a H323 URI will map. Default is H323.
;
h323driver = H323
==
I got the enum.info info, from the site, http://nona.net/features/enum/ .
cheerz
Ben.
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Re: [asterisk-users] asterisk, iaxmodem, hylafax quality problem
Are you use digium card? digium pri card offen cause many problems, check zttest___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ChanIsAvail
I managed to work around my Dialplan. The ChanIsAvail application is great, except it only returns the 1st available channel. Could there be a ChansAreAvail which returns all the channels available instead of just the first. I'm sure it could be implemented as a macro or I guess a rewrite of the code. Anyone want a go? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk, iaxmodem, hylafax quality problem
Artifex Maximus wrote: Hello, Everything is fine when caller use ECM but when ECM isn't in use I often got unusable incoming faxes (much often that it should be). But when I switch back to fax machine that receive faxes perfectly (at least no visible error). Where should be the problem? Is there any solution for improving This belongs on the HylaFAX mailing list. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Can you explain why multiple registration isan important (missing) feature ?
Lacy Moore - Aspendora wrote: On 9/20/06, Craig Guy [EMAIL PROTECTED] wrote: [9580] type=peer auth=000413242fff:[EMAIL PROTECTED] It would be [MAC ADDRESS] type=peer ...etc.. Or at least, that's how I interpreted what Eric said. I think that's an excellent approach. THe phones are devices. An extension calls one or more devices. Makes a lot more sense than multiple extensions calling multiple extensions. Your definition in the sip.conf would be defining devices according to their MAC addresses. Your dial plan would call these devices based on extensions. exten = 100,1,Dial(SIP/MAC) ; where MAC is the MAC address of the phone We only user type=peer / type=user for servers since they commonly require different incoming .vs. outgoing auth. For phones we user type=friend. [0004f201e443-a] callerid=Jay Kresbach 9852461234 [EMAIL PROTECTED] type=friend host=dynamic secret=0004f201e443-a context=toll-access [0004f201e443-b] callerid=Jay Kresbach 9852461234 [EMAIL PROTECTED] type=friend host=dynamic secret=0004f201e443-b context=toll-access ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Can you explain why multiple registration isan important (missing) feature ?
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Your definition in the sip.conf would be defining devices according to their
MAC addresses. Your dial plan would call these devices based on extensions.
exten = 100,1,Dial(SIP/MAC) ; where MAC is the MAC address of the phone
All right. Then I give to my girlfriend my number 1234567 and she calls me in,
how will I know to which MAC address I need to pass call?
Perhaps you are tying to use wildcard destinations in your setup. This
does not scale.
Wildcard:
exten = 1234567,1,Dial(SIP/${EXTEN})
This does not scale.
Each extension should have it's own exten = line and Dial(... line.
exten = 1234567,1,Dial(SIP/[0004f201e443-a) because 0004f201e443-a is
the userid of the phone that you want to send the call to.
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Re: [asterisk-users] asterisk, iaxmodem, hylafax quality problem
Hello, On 9/21/06, Doug Lytle [EMAIL PROTECTED] wrote: Artifex Maximus wrote: Hello, Everything is fine when caller use ECM but when ECM isn't in use I often got unusable incoming faxes (much often that it should be). But when I switch back to fax machine that receive faxes perfectly (at least no visible error). Where should be the problem? Is there any solution for improving This belongs on the HylaFAX mailing list. OK, sorry. That's why I post there as well. I don't really know where is the problem or where should I improve something for better result. Might in Asterisk channel setup might in iaxmodem codec setup or might in Hylafax setup. bye, Zsolt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk, iaxmodem, hylafax quality problem
Hello, On 9/21/06, Ma Zhiyong [EMAIL PROTECTED] wrote: Are you use digium card? digium pri card offen cause many problems, check zttest Yes, it's a T405P. Is zttest disturb the current calls or might works in parallel with calls? Because it's very busy in worktime. And what should I look for on result? bye, Zsolt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two phones, same number
Why don't you simply give them separate extensions and put them in a ring group. Or disable call waiting on this phone, and forward the second call using Call Forward On Busy to a queue, where MoH file will be a busy phone signal. Called will hear a busy phone signal and the second phone will still be ringing. But whats the point to make the second phone ring if caller is hearing a busy tone. He'll hang up anyways. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using Asterisk with IVR connected with legacy pbx via rs-232
Hi,I have some cases that I need to use Asterisk as an IVR/VoiceMail only. It will be connected to legacy pbx using a serial port (R2-232) to exchange integrations and/or messages to allow pbx to send to terminal extensions 'message indications' (a led on in KS). I know Asterisk can do it alone and better, but in some cases isn't possible to change the pbx structure and this protocol via rs-232 is widely used for some big pbx systems.Any direction? Is there already a solution for this? Or I need to do a custom development? Thanks in advance!Paulo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoicemailMain()
Hi! Is this possible to make asterisk follow the dial plan after executing VoicemailMain? Thanks, Michel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Using Asterisk with IVR connected with legacy pbxvia rs-232
HI. Depends the kind of PBX are you using. For example in some cases like Meridian is imposible to integrate the legacy PBX funcinalities like light on the phone for indicate the voicemail sign. So i don´t know other systems but i integrate the voicemail, IVR and ACD module with Meridian Option 11 and its works perfect. So the only problem that a got was the MWI on the existing meridian phone. We resolve the issue using the mail notification. But i got some ideas how to resolve the MWI issue but you need some develop depending of the Legacy PBX. Any. Let me know. Cristian. From: Paulo Garcia [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Using Asterisk with IVR connected with legacy pbxvia rs-232 Date: Thu, 21 Sep 2006 09:13:14 -0300 Hi, I have some cases that I need to use Asterisk as an IVR/VoiceMail only. It will be connected to legacy pbx using a serial port (R2-232) to exchange integrations and/or messages to allow pbx to send to terminal extensions 'message indications' (a led on in KS). I know Asterisk can do it alone and better, but in some cases isn't possible to change the pbx structure and this protocol via rs-232 is widely used for some big pbx systems. Any direction? Is there already a solution for this? Or I need to do a custom development? Thanks in advance! Paulo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Find a local pizza place, music store, museum and more then map the best route! http://local.live.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicemailMain()
Didnt quite get ur question. But, if you mean, you want to, for e.g. play a file, dial out another number, sing a song, dance around, after execution of VoicemailMain, yes, its very much possible. Just add your enhanced dialplan at the next priority of VoicemailMain. cheerz - Ben Michel Zenone wrote: Hi! Is this possible to make asterisk follow the dial plan after executing VoicemailMain? Thanks, Michel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk / chan_capi problems
Armin Schindler wrote: On Thu, 21 Sep 2006, Klaus Darilion wrote: Hi! I have problems with an Asterisk box which was running fine for some time but now causes problems (asterisk restarts, hangs ...). I use asterisk 1.2.7.1 with chan_capi-cm-0.6.5 and divas4linux-melware-3.0.f-106.622-1 In syslog I see lots of the following messages: Sep 20 07:02:41 ast01 kernel: kcapi: appl 1 ncci 0x130104 down Sep 20 07:02:43 ast01 kernel: kcapi: appl 1 ncci 0x60202 up Sep 20 07:02:43 ast01 kernel: kcapi: appl 1 ncci 0x10204 up Sep 20 07:03:12 ast01 kernel: kcapi: appl 1 ncci 0x10204 down Sep 20 07:03:12 ast01 kernel: kcapi: appl 1 ncci 0x60202 down Sep 20 07:06:25 ast01 kernel: kcapi: appl 1 ncci 0x20104 up Sep 20 07:06:40 ast01 kernel: kcapi: appl 1 ncci 0x20104 down Sep 20 07:07:13 ast01 kernel: kcapi: appl 1 ncci 0x70302 up What are the cause of this messages? May they be related with the asterisk crashes ? No, they are not related. These messages are just info messages from common kernelcapi driver about b-channel up: ncci 0x up b-channel down: ncci 0x down Just to make sure: What does it mean if a B channel goes up - is there a call started on this channel? For the problems you have, some logs would be needed. No suspect logs at all. I've increase loglevel now and wait for new crashes. regards klaus Armin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-users] Calls between IAX2 Clients don't work correctly
Hello, I have troubles with calls with IAX2 phones, either sofphones and hardphones. When I make a call between an IAX phone and a normal phone(RTB) the person connected on the RTB phone can hear the caller from the IAX2 phone, but the caller from the IAX phone can't hear what tells the person on the normal phone. If a call is done between two IAX phones, for example one hardphone and a sofphone, nobody hears anything. But the strange is that the calls between servers and with IAX2 works, it is, that if I call from a normal phone connected to the RDSI PBX, that this one is redirectioned to the asterisk PBX and tge call is sended over the IAX2 provider contracted to any location, it works perfectly. In conclusion, the IAX2 works correctly between servers, but between clients, the sound isn't sent correctly. We've tested it with different softphones and a hardphone, and noone has worked correctly, having the problems mentioned above. Regards, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicemailMain()
Michel Zenone wrote:
Hi!
Is this possible to make asterisk follow the dial plan after executing
VoicemailMain?
Happens by default, unless the caller hangs up of course.
; Give voicemail at extension 3509
exten = 3509,1,SetVar(LOOP=1)
exten = 3509,2,Answer
exten = 3509,3,Wait(.5)
exten = 3509,4,GotoIf($[X${RDNIS} = X]?5:10)
exten = 3509,5,VoicemailMain
exten = 3509,6,Wait(.5)
exten = 3509,7,GotoIf($[${LOOP} = 3]?11:8)
exten = 3509,8,SetVar(LOOP=$[${LOOP} + 1])
exten = 3509,9,Goto(5)
exten = 3509,10,VoiceMail(u${RDNIS})
exten = 3509,11,Hangup
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Re: [asterisk-users] asterisk / chan_capi problems
On Thu, 21 Sep 2006, Klaus Darilion wrote: Armin Schindler wrote: On Thu, 21 Sep 2006, Klaus Darilion wrote: Hi! I have problems with an Asterisk box which was running fine for some time but now causes problems (asterisk restarts, hangs ...). I use asterisk 1.2.7.1 with chan_capi-cm-0.6.5 and divas4linux-melware-3.0.f-106.622-1 In syslog I see lots of the following messages: Sep 20 07:02:41 ast01 kernel: kcapi: appl 1 ncci 0x130104 down Sep 20 07:02:43 ast01 kernel: kcapi: appl 1 ncci 0x60202 up Sep 20 07:02:43 ast01 kernel: kcapi: appl 1 ncci 0x10204 up Sep 20 07:03:12 ast01 kernel: kcapi: appl 1 ncci 0x10204 down Sep 20 07:03:12 ast01 kernel: kcapi: appl 1 ncci 0x60202 down Sep 20 07:06:25 ast01 kernel: kcapi: appl 1 ncci 0x20104 up Sep 20 07:06:40 ast01 kernel: kcapi: appl 1 ncci 0x20104 down Sep 20 07:07:13 ast01 kernel: kcapi: appl 1 ncci 0x70302 up What are the cause of this messages? May they be related with the asterisk crashes ? No, they are not related. These messages are just info messages from common kernelcapi driver about b-channel up: ncci 0x up b-channel down: ncci 0x down Just to make sure: What does it mean if a B channel goes up - is there a call started on this channel? No, the call already has started. The b-channel is the voice/data connection of the call. For the problems you have, some logs would be needed. No suspect logs at all. I've increase loglevel now and wait for new crashes. Okay. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Two phones, same number
ZZ == Zeeshan Zakaria [EMAIL PROTECTED] writes: ZZ Why don't you simply give them separate extensions and put them in ZZ a ring group. I'm not quite sure what you mean by ring group. Perhaps you could elaborate? ZZ Or disable call waiting on this phone, and forward the second call ZZ using Call Forward On Busy to a queue, where MoH file will be a ZZ busy phone signal. Called will hear a busy phone signal and the ZZ second phone will still be ringing. I don't want the second phone to ring. ZZ But whats the point to make the second phone ring if caller is ZZ hearing a busy tone. He'll hang up anyways. I want the caller to get the busy tone. Basically, if I'm talking on one phone, I don't want the other phone to ring. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] enumlookup - deprecated working - but appreciated one duznt :-(
On Sep 21, 2006, at 12:28 PM, Benjamin Jacob wrote: Another observation, the cmd EnumLookup, duz search thru different domains listed in enum.conf, but the function ENUMLOOKUP doesn't(it just searches for e164.arpa, and if not found, gives up, if the zone argument is left empty). Anyone worked around this one?? yeah, use multiple calls to the function, all with a different search domain ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting QOS settings in asterisk and/or CentOS?
How would I go about setting the TOS bit to "RTP IP TOS Byte: 18 (hex)" for SIP and IAX traffic at the asterisk server? Also, Do you have a quick reference on how to configure a Cisco switch to prioritize SIP traffic? I check in various Cisco docs, and there are so many references, and none of them seem to relate directly to using the TOS bit for QOS. I am looking into using the TOS bit because that is the only method that my SIP devices use. (Citel Handset Gateway) ref: QOS settings from Citel Handset Gateway: Handset Gateway - QoS Configuration IP Type of Service RTP IP TOS Byte: 18 (hex) Silence Suppression Mute Mode: On, UDP keep-alive every 10 secondsG.711 Voice Activity Detection: Off Codec Preferences G.711u: 1 (Highest priority) G.711a: 2 Thank You, Steven BerkHolz- MCSA - MCSE -Manager of Information SystemsTESCO Group CompaniesFax. 248-836-5101www.TESCOGroup.com Board member ofwww.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LDAP athentication
I have some in house scripts, but it's definitely not real time.. It uses perl to generate various config files to be included. The philosophy behind it is to store the dial plan on the hard drive (and in some sort of rcs) and to generate phone objects into separate config files. The LDAP schema is fairly abstracted -- there is no dial plan steps in it.. I could pass it along if you're interested.On 9/18/06, Andre O. [EMAIL PROTECTED] wrote: Hello, Does anyone have a solution for having SIP users to authenticate against a LDAP server?Best Regards,Andre O. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looped message playback
Hello, I'm trying to play an audio file to a phone an arbitrary number of times. The audio is a five-second segment of a sine wave. I need this to be played repeatedly without gaps between playbacks. I've tried doing this in the dial plan, e.g.: exten = s,1,Playback(tonefile) exten = s,2,Goto(1) but there is too long of a gap between the playbacks. Does anyone know of a way to achieve this? Thanks, Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 2.0.1 Software
Had problems the first night I downloaded and installed, but tracked to very poor net conditions. Reloaded this week and all has been working fine. Nice to finally be able to use all the buttons on the sidecar for blf:) It may be my imagination, but it also seems that it is staying in sync through reloads, or at least resyncing shrtly after one. On Sep 20, 2006, at 10:13 PM, Douglas Garstang wrote: No problems with SIP subscriptions here... -Original Message- From: Lacy Moore - Aspendora [mailto:[EMAIL PROTECTED] Sent: Wed 9/20/2006 8:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] Polycom 2.0.1 Software I couldn't get the hinting to work. Went back to 1.6.7, same config, and it works. I wasn't sure if the config had changed between the two. But, now that you mention it, I did experience a phone rebooting several times. I was half-way paying attention, so I just thought I had done something. On 9/20/06, Douglas Garstang [EMAIL PROTECTED] wrote: Is anyone seeing any weird stuff with the latest Polycom 2.0.1 SIP application software? A few of our phones, after upgrading would come up with a 0x4000 Configuration Error. Rebooting again a couple of times, or doing a 'Format Local Filesystem' seemed to fix it, with no change to the config files on the FTP server. I've also had an instance where a phone was refusing to register after upgrading. It worked fine, first boot, after doing a 'format local filesystem' on the phone, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Aspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Two phones, same number
voip*CLI show application chanisavail
voip*CLI
-= Info about application 'ChanIsAvail' =-
[Synopsis]
Check channel availability
[Description]
ChanIsAvail(Technology/resource[Technology2/resource2...][|options]):
This application will check to see if any of the specified channels are
available. The following variables will be set by this application:
${AVAILCHAN} - the name of the available channel, if one exists
${AVAILORIGCHAN} - the canonical channel name that was used to create the
channel
${AVAILSTATUS} - the status code for the available channel
Options:
s - Consider the channel unavailable if the channel is in use at all
j - Support jumping to priority n+101 if no channel is available
Use chanisavail to check if one or both phones is busy - if either is busy,
redirect to voicemail/busy/whatever.
Wes Baehr
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Benny Amorsen
Sent: Thursday, September 21, 2006 9:22 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Two phones, same number
ZZ == Zeeshan Zakaria [EMAIL PROTECTED] writes:
ZZ Why don't you simply give them separate extensions and put them in
ZZ a ring group.
I'm not quite sure what you mean by ring group. Perhaps you could
elaborate?
ZZ Or disable call waiting on this phone, and forward the second call
ZZ using Call Forward On Busy to a queue, where MoH file will be a
ZZ busy phone signal. Called will hear a busy phone signal and the
ZZ second phone will still be ringing.
I don't want the second phone to ring.
ZZ But whats the point to make the second phone ring if caller is
ZZ hearing a busy tone. He'll hang up anyways.
I want the caller to get the busy tone. Basically, if I'm talking on
one phone, I don't want the other phone to ring.
/Benny
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Re: [asterisk-users] Re: Two phones, same number
set group/check group On Sep 21, 2006, at 8:22 AM, Benny Amorsen wrote: ZZ == Zeeshan Zakaria [EMAIL PROTECTED] writes: ZZ Why don't you simply give them separate extensions and put them in ZZ a ring group. I'm not quite sure what you mean by ring group. Perhaps you could elaborate? ZZ Or disable call waiting on this phone, and forward the second call ZZ using Call Forward On Busy to a queue, where MoH file will be a ZZ busy phone signal. Called will hear a busy phone signal and the ZZ second phone will still be ringing. I don't want the second phone to ring. ZZ But whats the point to make the second phone ring if caller is ZZ hearing a busy tone. He'll hang up anyways. I want the caller to get the busy tone. Basically, if I'm talking on one phone, I don't want the other phone to ring. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asteisk plays music on hold starting from randompoint
Giorgio, This format works just how you want it to - it will play the files starting at the beginning. If you convert your files to .wav or some other format - you will get a cleaner sound. From what I have read, Fedora opens a single instance of the file regardless of the number accessing it, using Format_MP3 (the native MOH player in Asterisk 1.2 that replaced mpg123) - so you should save on CPU usage using this program instead of mpg123. From the post that I read, the person was testing it using a .wav file. Here is the copied musiconhold.conf setup [default] mode=files directory=/var/lib/asterisk/moh-native random=yes ; Play the files in a random order Robert Chadwell 800-330-7704 toll free 813-343-0181 ph 813-413-8195 fx Please feel free to IM me as well AOL Screenname: cmgrobert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo Sent: Wednesday, September 20, 2006 12:19 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asteisk plays music on hold starting from randompoint Hi, I'm using mpg123 to play music on hold but it seems that Asterisk does play the music from a random point: is there a way to make my music on hold always starting from beginning? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looped message playback
Earle Clubb wrote: Hello, I'm trying to play an audio file to a phone an arbitrary number of times. The audio is a five-second segment of a sine wave. I need this to be played repeatedly without gaps between playbacks. I've tried doing this in the dial plan, e.g.: exten = s,1,Playback(tonefile) exten = s,2,Goto(1) but there is too long of a gap between the playbacks. Does anyone know of a way to achieve this? You have a long gap in your tone file. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk with IVR connected with legacy pbxvia rs-232
Hi Cristianthank you for your reply.There are two main issues that I need to provide:1 - Lights on/off when alerting for a new voicemail message. I think I need to develop some new exten application to allow me send rs-232 message to pbx turns on/off the KS light. If someone has another approach , please, tell me. 2 - Some pbx, when a user try to call a busy extensions (or no answer) he will be transfer to another extension (connect in asterisk in this case) and simultaneouly send a rs-232 message telling to voicemail which mailbox need to be started. Asterisk need to receive this serial message, parse the command and mailbox and then answer the extension, redirecting directly to correct mailbox. AFAIK, Asterisk doesn't have built in functions to support this kind of operation then may be I need to create custom applications to do that. Is it correct?Thanks in advance!Paulo On 9/21/06, kritikus Araklidas [EMAIL PROTECTED] wrote: HI.Depends the kind of PBX are you using. For example in some cases likeMeridian is imposible to integrate the legacy PBX funcinalities like lighton the phone for indicate the voicemail sign. So i don´t know other systems but i integrate the voicemail, IVR and ACD module with Meridian Option 11and its works perfect. So the only problem that a got was the MWI on theexisting meridian phone. We resolve the issue using the mail notification. But i got some ideas how to resolve the MWI issue but you need some developdepending of the Legacy PBX.Any.Let me know.Cristian.From: Paulo Garcia [EMAIL PROTECTED]Reply-To: Asterisk Users Mailing List - Non-CommercialDiscussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.comSubject: [asterisk-users] Using Asterisk with IVR connected with legacypbxvia rs-232Date: Thu, 21 Sep 2006 09:13:14 -0300 Hi,I have some cases that I need to use Asterisk as an IVR/VoiceMail only. Itwill be connected to legacy pbx using a serial port (R2-232) to exchangeintegrations and/or messages to allow pbx to send to terminal extensions 'message indications' (a led on in KS).I know Asterisk can do it alone and better, but in some cases isn'tpossibleto change the pbx structure and this protocol via rs-232 is widely used for some big pbx systems.Any direction? Is there already a solution for this? Or I need to do acustom development?Thanks in advance!Paulo___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users_Find a local pizza place, music store, museum and more…then map the bestroute! http://local.live.com___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Format_MP3, Streaming, File Formats, MOH
I agree, using mpg123 for streaming from Shoutcast-type servers hasn't worked well for me either. I would prefer to use SlimServer as you can, through LAME, drop the bitrate and file quality down to the point where you wouldn't necessarily need further conversion, but this requires /stream.mp3 to be added to the end of the URL (which I haven't been able to get to work). If Shoutcast streaming HAS worked well for folks, maybe you could provide us with some insight. Robert Chadwell 800-330-7704 toll free 813-343-0181 ph 813-413-8195 fx Please feel free to IM me as well AOL Screenname: cmgrobert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Raphaël Jacquot Sent: Wednesday, September 20, 2006 10:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Format_MP3, Streaming, File Formats, MOH Asterisk [Submusic] wrote: musiconhold.conf [shoutcast] mode=custom application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 http://stream128.submusic.ch:8004/ ; The '/' in the stream URL is important ! I tried this. however it doesn't work. apparently, asterisk doesn't read from the mpg123 when no one is listening to MOH, and stuff appear to be stacking inside a pipe of some sort. when the next caller gets the MOH, he gets the music from 5 minutes ago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk / chan_capi problems
Armin Schindler wrote: On Thu, 21 Sep 2006, Klaus Darilion wrote: Armin Schindler wrote: On Thu, 21 Sep 2006, Klaus Darilion wrote: Hi! I have problems with an Asterisk box which was running fine for some time but now causes problems (asterisk restarts, hangs ...). I use asterisk 1.2.7.1 with chan_capi-cm-0.6.5 and divas4linux-melware-3.0.f-106.622-1 In syslog I see lots of the following messages: Sep 20 07:02:41 ast01 kernel: kcapi: appl 1 ncci 0x130104 down Sep 20 07:02:43 ast01 kernel: kcapi: appl 1 ncci 0x60202 up Sep 20 07:02:43 ast01 kernel: kcapi: appl 1 ncci 0x10204 up Sep 20 07:03:12 ast01 kernel: kcapi: appl 1 ncci 0x10204 down Sep 20 07:03:12 ast01 kernel: kcapi: appl 1 ncci 0x60202 down Sep 20 07:06:25 ast01 kernel: kcapi: appl 1 ncci 0x20104 up Sep 20 07:06:40 ast01 kernel: kcapi: appl 1 ncci 0x20104 down Sep 20 07:07:13 ast01 kernel: kcapi: appl 1 ncci 0x70302 up What are the cause of this messages? May they be related with the asterisk crashes ? No, they are not related. These messages are just info messages from common kernelcapi driver about b-channel up: ncci 0x up b-channel down: ncci 0x down Just to make sure: What does it mean if a B channel goes up - is there a call started on this channel? No, the call already has started. The b-channel is the voice/data connection of the call. So this means, that somewhere in between SETUP and CONNECT the B channel goes up, and goes down with RELEASE? klaus For the problems you have, some logs would be needed. No suspect logs at all. I've increase loglevel now and wait for new crashes. Okay. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Format_MP3, Streaming, File Formats, MOH
Frederic, Did this work for you under Asterisk 1.2x? If it did, did you receive Warning Flexible rate not heavily tested notices in the Asterisk CLI? Robert Chadwell 800-330-7704 toll free 813-343-0181 ph 813-413-8195 fx Please feel free to IM me as well AOL Screenname: cmgrobert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk [Submusic] Sent: Wednesday, September 20, 2006 9:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Format_MP3, Streaming, File Formats, MOH Hi, This config is working for me: _ musiconhold.conf [shoutcast] mode=custom application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 http://stream128.submusic.ch:8004/ ; The '/' in the stream URL is important ! _ extensions.conf exten = 17,1,Answer exten = 17,2,MusicOnHold(shoutcast) _ Regards Frederic De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Robert Chadwell Envoyé: mardi, 19. septembre 2006 14:47 À: asterisk-users@lists.digium.com Objet: [asterisk-users] Format_MP3, Streaming, File Formats, MOH Format_MP3 appears to play MOH files starting at the beginning of each file, using the .wav file format, making for some repetitive hold music unless you alter the file itself to begin somewhere in the middle. Solution: One stream that all users connect to giving dynamic hold music (tried and tested in A1.0x using mpg123 with some success, and Icecast or Slimserver or Shoutcast) Format_MP3 doesnt seem to stream, and the wiki is wrong about streamplayer being used to play streams, as it is only used to play raw TCP streams. There are many questions in forums on the web with no answers about how to solve this dilemma, How do you get users connected to a constantly-changing stream of music instead of streams starting from the beginning (regardless of whether Linux counts them as one stream or not where the processor is concerned)? Hopefully, at the end of this thread, I will have enough information to go back to these web-forums and post the answer. To get it started here is what I have tried that hasnt worked. In most all cases the response is Music on hold started, immediately followed by Music on hold stopped with no sound in any case. ;[classes] ;mode=custom ;application=/usr/bin/streamplayer 194.158.114.67 8000 ;format=ulaw --- Straight From The Music On Hold Wiki ;default = quietmp3:/var/lib/asterisk/mohmp3-dummy -@,http://www.shoutcast.com/sbin/tunein-station.pls?id=7733filename=playlist.pls --- From the Nerd Vittles Tutorial with the -@ added because mpg123 seemed to ask for it since the file was a .pls ;default = mp3:http://127.0.0.1:9000/stream.mp3 -- From a forum of someone using mpg123 to stream SlimServer (installed mpg123 v0.60 with no success here) [default] mode=files directory= /var/lib/asterisk/mohmp3-dummy,http://127.0.0.1:9000/stream.mp3 -- Tried a 1.2 format ;default = quietmp3:/var/lib/asterisk/mohmp3-dummy,http://193.251.154.243:8000/ -- Thought maybe it was SlimServer so tried to stream the top Shoutcast station ;default = quietmp3:/var/lib/asterisk/mohmp3-dummy,http://127.0.0.1:9000/stream.mp3 -- Tried to stream Slimserver using the old format Thank you in advance I have been at this for a week now. How did you make it work in Asterisk 1.2x? Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Looped message playback
Why not just merge the file together a few times using an audio program and make a longer file? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Earle Clubb Sent: Thursday, September 21, 2006 9:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Looped message playback Hello, I'm trying to play an audio file to a phone an arbitrary number of times. The audio is a five-second segment of a sine wave. I need this to be played repeatedly without gaps between playbacks. I've tried doing this in the dial plan, e.g.: exten = s,1,Playback(tonefile) exten = s,2,Goto(1) but there is too long of a gap between the playbacks. Does anyone know of a way to achieve this? Thanks, Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Looped message playback
DOH I just read that you said an arbitrary number of times no wonder you asked this question Please ignore me. :) -Original Message- From: Bill Gibbs Sent: Thursday, September 21, 2006 10:11 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Looped message playback Why not just merge the file together a few times using an audio program and make a longer file? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Earle Clubb Sent: Thursday, September 21, 2006 9:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Looped message playback Hello, I'm trying to play an audio file to a phone an arbitrary number of times. The audio is a five-second segment of a sine wave. I need this to be played repeatedly without gaps between playbacks. I've tried doing this in the dial plan, e.g.: exten = s,1,Playback(tonefile) exten = s,2,Goto(1) but there is too long of a gap between the playbacks. Does anyone know of a way to achieve this? Thanks, Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco 7970 behind NAT
Shortly after I sent this e-mail I got it figured out. In sip.conf I had to put nat=no. The phone config also need to have all NAT features turned off. It was strange because I was sniffing the packets for the registration and saw no authentication information coming from the phone (with a really high source port number I might add), then I turned off NAT in sip.conf and did a reload and all of a sudden the phone was registered. This is the opposite of what I do for my 7960s running the 7.4 SIP image. After I got the 7970 working I had a 7961 running the 8.0.2SR1 unified image and had to do the same thing. The config files and settings for phones running the newer Cisco SIP software all require these parameters. Just an F.Y.I. Jeremiah The problem is that the 7970 phones by default are listening for replies to their register requests on port 5060. Unfortunately, the phone sends them out from random ports. So, if you have nat=yes on the sip peer in asterisk then the asterisk will send the reply to the port the request came from and not 5060. The only deployment we have done of these phones with NAT involved was for 2 executives at a branch office. In order to get the phones working we had to set the XML configs for the phones to send the external IP address of the firewall (you'll need a static IP for this to work) and to request replies on a custom port other than 5060. We then gave the phones DHCP reservations so they would always get the same private IP and mapped the custom sip ports through the firewall to each of the 2 phones. The sip peers in asterisk then had nat=no. Kind of a kludge but since there were only two 7970 phones it was manageable. The other cisco phones don't seem to have this problem. Perhaps somebody out there knows a way to make the 7970 phones accept SIP responses back to the originating port. I wasted several hours but couldn't figure it out. -Evan -- __ Rock River InternetJeremiah Millay 202 W. State St, 8th Floor [EMAIL PROTECTED] Rockford, IL 61101 815-968-9888 Ext. 2202 USA fax 968-6888 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems with Polycom 500 boot up
Interesting, it shouldn't work according to Polycom, but I guess go with it :)I really apologize, I haven't worked with a IP 500 in such a long time and most people who mention the 500 really end up meaning the 501, I guess I assumed wrong. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 20, 2006, at 7:14 PM, Forum wrote: Like a charm From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dean Collins Sent: Wednesday, September 20, 2006 5:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] problems with Polycom 500 boot up Lol, does it work thought? Cheers,Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Forum Expansive Sent: Wednesday, 20 September 2006 7:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] problems with Polycom 500 boot up I just updated my 500 to the latest - 3.2.2. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Azfhasterisk Sent: Wednesday, 20 September 2006 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] problems with Polycom 500 boot up I am not positive but I thought that the 2.6.2 bootrom was the highest you could put on the ip500. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jessee J Holmes Sent: Wednesday, September 20, 2006 2:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] problems with Polycom 500 boot up Get 3.2.2 from your reseller or installer. Bootrom 3.2.2 and firmware 2.0.1 is the latest available from Polycom. Your phone installer, service provider, or reseller should be able to provide you with this firmware. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 20, 2006, at 2:53 PM, Forum wrote: I think that's the problem the bootROM version on the phone is 2.02 Apr 02 16.33. Does anyone have this version and the corresponding Sip? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Eric "ManxPower" Wieling Sent: Wednesday, September 20, 2006 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] problems with Polycom 500 boot up Forum wrote: Thanks for your response. Unfortunately I still receive the same error - 'Error updating bootrom' - no matter what version of sip and the bootROM I upload to the ftp site. I have even used the latest release of the fimware - could I have somehow broke the phone with a corrupted flash. How do I do a full format when it can not update the bootROM? Check the EXISTING BootROM on the phone. You can't usually downgrade versions. Also check the password configured for the FTP user on the phone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems with Polycom 500 boot up
Dear Dean,The difference between these two phones is simply that the IP 501 phone has extra memory on it which was added to the phone ONLY to provide HTTPS provisioning options. (this is what Polycom told us anyways)Funny thing is, not many people know how to use the HTTPS options because it's not well documented by the manufacturer supposedly. *shakes head* I've been working pretty closely with Polycom right now to get this FINALLY documented, hopefully well. We'll post it to our knowledge-bases when I get something put together on this topic (no idea when this will happen yet since I'm waiting on Polycom engineers to do some more in-lab testing). Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 20, 2006, at 7:33 PM, Dean Collins wrote: Speaking of which, can anyone tell me the differences between the IP500 and the IP501? Cheers,Dean From: Dean Collins Sent: Wednesday, 20 September 2006 8:10 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] problems with Polycom 500 boot up Lol, does it work thought? Cheers,Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Forum Expansive Sent: Wednesday, 20 September 2006 7:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] problems with Polycom 500 boot up I just updated my 500 to the latest - 3.2.2. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Azfhasterisk Sent: Wednesday, 20 September 2006 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] problems with Polycom 500 boot up I am not positive but I thought that the 2.6.2 bootrom was the highest you could put on the ip500. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jessee J Holmes Sent: Wednesday, September 20, 2006 2:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] problems with Polycom 500 boot up Get 3.2.2 from your reseller or installer. Bootrom 3.2.2 and firmware 2.0.1 is the latest available from Polycom. Your phone installer, service provider, or reseller should be able to provide you with this firmware. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 20, 2006, at 2:53 PM, Forum wrote: I think that's the problem the bootROM version on the phone is 2.02 Apr 02 16.33. Does anyone have this version and the corresponding Sip? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Eric "ManxPower" Wieling Sent: Wednesday, September 20, 2006 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] problems with Polycom 500 boot up Forum wrote: Thanks for your response. Unfortunately I still receive the same error - 'Error updating bootrom' - no matter what version of sip and the bootROM I upload to the ftp site. I have even used the latest release of the fimware - could I have somehow broke the phone with a corrupted flash. How do I do a full format when it can not update the bootROM? Check the EXISTING BootROM on the phone. You can't usually downgrade versions. Also check the password configured for the FTP user on the phone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Format_MP3, Streaming, File Formats, MOH
Robert Chadwell wrote: I agree, using mpg123 for streaming from Shoutcast-type servers hasn't worked well for me either. I would prefer to use SlimServer as you can, through LAME, drop the bitrate and file quality down to the point where you wouldn't necessarily need further conversion, but this requires /stream.mp3 to be added to the end of the URL (which I haven't been able to get to work). If Shoutcast streaming HAS worked well for folks, maybe you could provide us with some insight. Has anyone got any config info for using slimserver ? Julian Robert Chadwell 800-330-7704 toll free 813-343-0181 ph 813-413-8195 fx Please feel free to IM me as well AOL Screenname: cmgrobert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Raphaël Jacquot Sent: Wednesday, September 20, 2006 10:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Format_MP3, Streaming, File Formats, MOH Asterisk [Submusic] wrote: musiconhold.conf [shoutcast] mode=custom application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 http://stream128.submusic.ch:8004/ ; The '/' in the stream URL is important ! I tried this. however it doesn't work. apparently, asterisk doesn't read from the mpg123 when no one is listening to MOH, and stuff appear to be stacking inside a pipe of some sort. when the next caller gets the MOH, he gets the music from 5 minutes ago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI Problems
Hi! Today I had problems with an E1 link to a Siemens PBX (Sangoma 2xE1 card). Although everything looked fine (wanrouter status connected, pri show status was also fine) there were no messages from the PBX to Asterisk. The log file is below. I reloaded asterisk and the kernel module several times - disconnect and reconnected the cables but nothing worked. Finally I rebootet the server and than everything was fine again. Does someone knows the reason for this problem? thanks klaus 12:23:58== Primary D-Channel on span 1 up 12:24:00== Primary D-Channel on span 1 up 12:24:01== Primary D-Channel on span 1 up 12:24:02== Primary D-Channel on span 1 up 12:24:02 -- B-channel 0/5 successfully restarted on span 2 12:24:03== Primary D-Channel on span 1 up 12:24:05 [ 00 01 7f ] 12:24:05 Unnumbered frame: 12:24:05 SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 12:24:05 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data 12:24:05 -- Got SABME from cpe peer. 12:24:05 Sending Unnumbered Acknowledgement 12:24:05 [ 00 01 73 ] 12:24:05 Unnumbered frame: 12:24:05 SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 12:24:05 M3: 3 P/F: 1 M2: 0 11: 3 [ UA (unnumbered acknowledgement) ] 0 bytes of data 12:24:05 -- Restarting T203 counter 12:24:05 -- Restarting T203 counter 12:24:05== Primary D-Channel on span 1 up 12:24:06 [ 00 01 7f ] 12:24:06 Unnumbered frame: 12:24:06 SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 12:24:06 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data 12:24:06 -- Got SABME from cpe peer. 12:24:06 Sending Unnumbered Acknowledgement 12:24:06 [ 00 01 73 ] 12:24:06 Unnumbered frame: 12:24:06 SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 12:24:06 M3: 3 P/F: 1 M2: 0 11: 3 [ UA (unnumbered acknowledgement) ] 0 bytes of data 12:24:06 -- Restarting T203 counter 12:24:06 -- Restarting T203 counter 12:24:06== Primary D-Channel on span 1 up ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looped message playback
Eric ManxPower Wieling wrote: Earle Clubb wrote: Hello, I'm trying to play an audio file to a phone an arbitrary number of times. The audio is a five-second segment of a sine wave. I need this to be played repeatedly without gaps between playbacks. I've tried doing this in the dial plan, e.g.: exten = s,1,Playback(tonefile) exten = s,2,Goto(1) but there is too long of a gap between the playbacks. Does anyone know of a way to achieve this? You have a long gap in your tone file. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Eric, Thanks for the reply. There is no gap in the tone file. The file begins with the sine wave going positive from the zero-crossing and ends with the wave at the zero-crossing from negative. Also, I can loop the file on my PC and there are no gaps in the audio. Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 2.0.1 Software
Is anyone seeing any weird stuff with the latest Polycom 2.0.1 SIP application software? My experiences with 2.01 have been mostly good. Definitely haven't seen any rebooting or error messages. I am, however, having problems getting the speeddial-to-key remapping to work. I waited for this feature for so long, but now I can only get the phone to remap one speed dial. If I try more than one, all the remapped buttons will dial only the first speed dial number listed in the directory. - Noah On 9/21/06, Jerry Jones [EMAIL PROTECTED] wrote: Had problems the first night I downloaded and installed, but tracked to very poor net conditions. Reloaded this week and all has been working fine. Nice to finally be able to use all the buttons on the sidecar for blf:) It may be my imagination, but it also seems that it is staying in sync through reloads, or at least resyncing shrtly after one. On Sep 20, 2006, at 10:13 PM, Douglas Garstang wrote: No problems with SIP subscriptions here... -Original Message- From: Lacy Moore - Aspendora [mailto:[EMAIL PROTECTED] Sent: Wed 9/20/2006 8:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] Polycom 2.0.1 Software I couldn't get the hinting to work. Went back to 1.6.7, same config, and it works. I wasn't sure if the config had changed between the two. But, now that you mention it, I did experience a phone rebooting several times. I was half-way paying attention, so I just thought I had done something. On 9/20/06, Douglas Garstang [EMAIL PROTECTED] wrote: Is anyone seeing any weird stuff with the latest Polycom 2.0.1 SIP application software? A few of our phones, after upgrading would come up with a 0x4000 Configuration Error. Rebooting again a couple of times, or doing a 'Format Local Filesystem' seemed to fix it, with no change to the config files on the FTP server. I've also had an instance where a phone was refusing to register after upgrading. It worked fine, first boot, after doing a 'format local filesystem' on the phone, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Aspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looped message playback
Bill Gibbs wrote: Why not just merge the file together a few times using an audio program and make a longer file? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Earle Clubb Sent: Thursday, September 21, 2006 9:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Looped message playback Hello, I'm trying to play an audio file to a phone an arbitrary number of times. The audio is a five-second segment of a sine wave. I need this to be played repeatedly without gaps between playbacks. I've tried doing this in the dial plan, e.g.: exten = s,1,Playback(tonefile) exten = s,2,Goto(1) but there is too long of a gap between the playbacks. Does anyone know of a way to achieve this? Thanks, Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The duration of the tone can vary at runtime and I have no way of knowing beforehand what the duration will be. Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Format_MP3, Streaming, File Formats, MOH
Hi, Yes , I use Asterisk 1.2.10 But , I don't have Warning Flexible rate not heavily tested notices in the Asterisk CLI The Shoutcast server is in the same box as Asterisk, and the stream source is in the same LAN Regards Fred ___ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert ChadwellSent: jeudi, 21. septembre 2006 15:58To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] Format_MP3, Streaming, File Formats, MOH Frederic, Did this work for you under Asterisk 1.2x? If it did, did you receive Warning Flexible rate not heavily tested notices in the Asterisk CLI? Robert Chadwell800-330-7704 toll free813-343-0181 ph813-413-8195 fxPlease feel free to IM me as wellAOL Screenname: cmgrobert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk [Submusic]Sent: Wednesday, September 20, 2006 9:58 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] Format_MP3, Streaming, File Formats, MOH Hi, This config is working for me: _ musiconhold.conf [shoutcast] mode=custom application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 http://stream128.submusic.ch:8004/ ; The '/' in the stream URL is important ! _ extensions.conf exten = 17,1,Answer exten = 17,2,MusicOnHold(shoutcast) _ Regards Frederic De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Robert ChadwellEnvoyé: mardi, 19. septembre 2006 14:47À: asterisk-users@lists.digium.comObjet: [asterisk-users] Format_MP3, Streaming, File Formats, MOH Format_MP3 appears to play MOH files starting at the beginning of each file, using the .wav file format, making for some repetitive hold music unless you alter the file itself to begin somewhere in the middle. Solution: One stream that all users connect to giving dynamic hold music (tried and tested in A1.0x using mpg123 with some success, and Icecast or Slimserver or Shoutcast) Format_MP3 doesnt seem to stream, and the wiki is wrong about streamplayer being used to play streams, as it is only used to play raw TCP streams. There are many questions in forums on the web with no answers about how to solve this dilemma, How do you get users connected to a constantly-changing stream of music instead of streams starting from the beginning (regardless of whether Linux counts them as one stream or not where the processor is concerned)? Hopefully, at the end of this thread, I will have enough information to go back to these web-forums and post the answer. To get it started here is what I have tried that hasnt worked. In most all cases the response is Music on hold started, immediately followed by Music on hold stopped with no sound in any case. ;[classes] ;mode=custom ;application=/usr/bin/streamplayer 194.158.114.67 8000 ;format=ulaw --- Straight From The Music On Hold Wiki ;default = quietmp3:/var/lib/asterisk/mohmp3-dummy -@,http://www.shoutcast.com/sbin/tunein-station.pls?id=7733filename=playlist.pls --- From the Nerd Vittles Tutorial with the -@ added because mpg123 seemed to ask for it since the file was a .pls ;default = mp3:http://127.0.0.1:9000/stream.mp3 -- From a forum of someone using mpg123 to stream SlimServer (installed mpg123 v0.60 with no success here) [default] mode=files directory= /var/lib/asterisk/mohmp3-dummy,http://127.0.0.1:9000/stream.mp3 -- Tried a 1.2 format ;default = quietmp3:/var/lib/asterisk/mohmp3-dummy,http://193.251.154.243:8000/ -- Thought maybe it was SlimServer so tried to stream the top Shoutcast station ;default = quietmp3:/var/lib/asterisk/mohmp3-dummy,http://127.0.0.1:9000/stream.mp3 -- Tried to stream Slimserver using the old format Thank you in advance I have been at this for a week now. How did you make it work in Asterisk 1.2x? Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Setting QOS settings in asterisk and/or CentOS?
Hello, For the Cisco QOS: Based on a Cisco Router all you need is a simple access-list. class-map match-any voip-class match ip rtp 10001 match access-group 150 ! ! policy-map voip-policy class voip-class priority xxx (in Kbits) access-list 150 permit udp any any eq 5060 access-list 150 permit udp any any eq 4569 Voila! Redouane De: BerkHolz, Steven [mailto:[EMAIL PROTECTED] Envoyé: jeudi 21 septembre 2006 15:33À: asterisk-users@lists.digium.comObjet: [asterisk-users] Setting QOS settings in asterisk and/or CentOS? How would I go about setting the TOS bit to "RTP IP TOS Byte: 18 (hex)" for SIP and IAX traffic at the asterisk server? Also, Do you have a quick reference on how to configure a Cisco switch to prioritize SIP traffic? I check in various Cisco docs, and there are so many references, and none of them seem to relate directly to using the TOS bit for QOS. I am looking into using the TOS bit because that is the only method that my SIP devices use. (Citel Handset Gateway) ref: QOS settings from Citel Handset Gateway: Handset Gateway - QoS Configuration IP Type of Service RTP IP TOS Byte: 18 (hex) Silence Suppression Mute Mode: On, UDP keep-alive every 10 secondsG.711 Voice Activity Detection: Off Codec Preferences G.711u: 1 (Highest priority) G.711a: 2 Thank You, Steven BerkHolz- MCSA - MCSE -Manager of Information SystemsTESCO Group CompaniesFax. 248-836-5101www.TESCOGroup.com Board member ofwww.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM2400P
Hi all, I have a TDM2400P w/ echo cancellation, 8 FXSs and 8 FXOs. They are installed respectively on banks 1,2,5 and 6. The problem I am having is that when I make a call using the ZAP channel, I can hear perfectly but the person on the other end is hearing my voice with lots of ticks. It would seem I am making this call over a very bad bandwidth which is not the case since this is the PSTN. My configuration files are below, I have the latest versions of Zaptel, Libpri and Asterisk. I am using Polycoms IP301 and IP430 Phones. I would appreciate help since I have to put this in production on Saturday. # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # loadzone = us defaultzone=us fxsks=1-4 fxsks=5-8 fxoks=17-20 fxoks=21-24 Zapata.conf [channels] language=en context=default ;switchtype=national echocancel=64 echocancelwhenbridged=no echotraining=800 toneduration=200 busydetect=yes signalling = fxs_ks rxgain=5.0 txgain=-10.0 channel = 1-4 channel = 5-8 signalling = fxo_ks channel = 17-20 channel = 21-24 Best Regards, Robson Ribeiro MSN: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] problems with Polycom 500 boot up
Thankyou for that valuable information Jessee, as all my handsets are on a local lan and I dont use HTTPS I guess picking up some second hand IP500s for $90-100 is the way to go. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jessee J Holmes Sent: Thursday, 21 September 2006 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] problems with Polycom 500 boot up Dear Dean, The difference between these two phones is simply that the IP 501 phone has extra memory on it which was added to the phone ONLY to provide HTTPS provisioning options. (this is what Polycom told us anyways) Funny thing is, not many people know how to use the HTTPS options because it's not well documented by the manufacturer supposedly. *shakes head* I've been working pretty closely with Polycom right now to get this FINALLY documented, hopefully well. We'll post it to ourknowledge-bases when I get something put together on this topic (no idea when this will happen yet since I'm waiting on Polycom engineers to do some more in-lab testing). Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 20, 2006, at 7:33 PM, Dean Collins wrote: Speaking of which, can anyone tell me the differences between the IP500 and the IP501? Cheers, Dean From: Dean Collins Sent: Wednesday, 20 September 2006 8:10 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] problems with Polycom 500 boot up Lol, does it work thought? Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Forum Expansive Sent: Wednesday, 20 September 2006 7:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] problems with Polycom 500 boot up I just updated my 500 to the latest - 3.2.2. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Azfhasterisk Sent: Wednesday, 20 September 2006 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] problems with Polycom 500 boot up I am not positive but I thought that the 2.6.2 bootrom was the highest you could put on the ip500. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jessee J Holmes Sent: Wednesday, September 20, 2006 2:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] problems with Polycom 500 boot up Get 3.2.2 from your reseller or installer. Bootrom 3.2.2 and firmware 2.0.1 is the latest available from Polycom. Your phone installer, service provider, or reseller should be able to provide you with this firmware. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 20, 2006, at 2:53 PM, Forum wrote: I think that's the problem the bootROM version on the phone is 2.02 Apr 02 16.33. Does anyone have this version and the corresponding Sip? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Eric ManxPower Wieling Sent: Wednesday, September 20, 2006 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] problems with Polycom 500 boot up Forum wrote: Thanks for your response. Unfortunately I still receive the same error - 'Error updating bootrom' - no matter what version of sip and the bootROM I upload to the ftp site. I have even used the latest release of the fimware - could I have somehow broke the phone with a corrupted flash. How do I do a full format when it can not update the bootROM? Check the EXISTING BootROM on the phone. You can't usually downgrade versions. Also check the password configured for the FTP user on the phone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
[asterisk-users] Re: Uninstalling Trixbox
Leo Ann Boon [EMAIL PROTECTED]wrote: Message: 2 Date: Thu, 21 Sep 2006 06:51:29 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Uninstalling Trixbox To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Yeah thats your problem I guess. Using a tarball 'package' certainly does not give you an easy way out or un-install. Best to start again or get your toothpick out :) Or you can try this from / (your root directory) :) rm -rf `tar tzf xxx.tar.gz` Note the backquotes enclosing the tar statement It will remove everything that you installed from the tar ball. Unfortunately the install script for Trixbox also carries out a number of rpm installs which you would need to uninstall. It should be possible to go through the install script line by line and produce and uninstall script for everything except the pure asterisk components but I strongly suspect it would be easier just to wipe the slate clean and start again. The moral of this story is that if you just want to have a play with Trixbox then install the VMWare version. -- Nic ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CURL
Ok, after requesting information to digium (no answer yet) and being informed that asterisk-dev is *NOT* a support hot line, I am trying in this list to see if someone has information on this regard.I know this is not a support area so I am only trying to get some clues.I have asterisk be and I am trying to use the CURL function (or application?). It is not available when I try it even though it is documented. Does anyone knows if there is a way to load it as a function/application inside asterisk? if so, is there code to download/compile to get it working inside asterisk?Any clue will be highly appreciated. (I keep trying digium support).Elpidio Elpidio Ramos PresidentRM International ServicesSA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax+52 (55)1755-6601 CellUSA: +1 (801) 494-1415 Office +1 (240) 250-8264 Fax +1 (801) 938-4740Direct ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asteisk plays music on hold starting from randompoint
Hi Robert, cpu is no problem..but if you say the sound is cleaner..well I can try Format_MP3...the problem is I do not know how to install it. I read it is inside asterisk-addons (there is a directory named format_mp3 inside) and I always install asterisk-addons on PBXs but...how can I invoke this command?? Is it automatic?? TIA Giorgio Incantalupo Robert Chadwell wrote: Giorgio, This format works just how you want it to - it will play the files starting at the beginning. If you convert your files to .wav or some other format - you will get a cleaner sound. From what I have read, Fedora opens a single instance of the file regardless of the number accessing it, using Format_MP3 (the native MOH player in Asterisk 1.2 that replaced mpg123) - so you should save on CPU usage using this program instead of mpg123. From the post that I read, the person was testing it using a .wav file. Here is the copied musiconhold.conf setup [default] mode=files directory=/var/lib/asterisk/moh-native random=yes ; Play the files in a random order Robert Chadwell 800-330-7704 toll free 813-343-0181 ph 813-413-8195 fx Please feel free to IM me as well AOL Screenname: cmgrobert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo Sent: Wednesday, September 20, 2006 12:19 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asteisk plays music on hold starting from randompoint Hi, I'm using mpg123 to play music on hold but it seems that Asterisk does play the music from a random point: is there a way to make my music on hold always starting from beginning? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CURL
You can always use the System() command in asterisk to call the curl executable. jerry --- Ok, after requesting information to digium (no answer yet) and being informed that asterisk-dev is *NOT* a support hot line, I am trying in this list to see if someone has information on this regard. I know this is not a support area so I am only trying to get some clues. I have asterisk be and I am trying to use the CURL function (or application?). It is not available when I try it even though it is documented. Does anyone knows if there is a way to load it as a function/application inside asterisk? if so, is there code to download/compile to get it working inside asterisk? Any clue will be highly appreciated. (I keep trying digium support). Elpidio Elpidio Ramos President RM International Services SA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax +52 (55) 1755-6601 Cell USA: +1 (801) 494-1415 Office +1 (240) 250-8264 Fax +1 (801) 938-4740 Direct ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Problems
Hi Klaus, may sound stoopid but...we had a similar problem then we discover the cable we made had wrong pinout..and asterisk gave me no messages. Hope may help. ::) Giorgio Incantalupo Klaus Darilion wrote: Hi! Today I had problems with an E1 link to a Siemens PBX (Sangoma 2xE1 card). Although everything looked fine (wanrouter status connected, pri show status was also fine) there were no messages from the PBX to Asterisk. The log file is below. I reloaded asterisk and the kernel module several times - disconnect and reconnected the cables but nothing worked. Finally I rebootet the server and than everything was fine again. Does someone knows the reason for this problem? thanks klaus 12:23:58== Primary D-Channel on span 1 up 12:24:00== Primary D-Channel on span 1 up 12:24:01== Primary D-Channel on span 1 up 12:24:02== Primary D-Channel on span 1 up 12:24:02 -- B-channel 0/5 successfully restarted on span 2 12:24:03== Primary D-Channel on span 1 up 12:24:05 [ 00 01 7f ] 12:24:05 Unnumbered frame: 12:24:05 SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 12:24:05 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data 12:24:05 -- Got SABME from cpe peer. 12:24:05 Sending Unnumbered Acknowledgement 12:24:05 [ 00 01 73 ] 12:24:05 Unnumbered frame: 12:24:05 SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 12:24:05 M3: 3 P/F: 1 M2: 0 11: 3 [ UA (unnumbered acknowledgement) ] 0 bytes of data 12:24:05 -- Restarting T203 counter 12:24:05 -- Restarting T203 counter 12:24:05== Primary D-Channel on span 1 up 12:24:06 [ 00 01 7f ] 12:24:06 Unnumbered frame: 12:24:06 SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 12:24:06 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data 12:24:06 -- Got SABME from cpe peer. 12:24:06 Sending Unnumbered Acknowledgement 12:24:06 [ 00 01 73 ] 12:24:06 Unnumbered frame: 12:24:06 SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 12:24:06 M3: 3 P/F: 1 M2: 0 11: 3 [ UA (unnumbered acknowledgement) ] 0 bytes of data 12:24:06 -- Restarting T203 counter 12:24:06 -- Restarting T203 counter 12:24:06== Primary D-Channel on span 1 up ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400P
On Thu, Sep 21, 2006 at 12:28:16PM -0400, Robson Ribeiro wrote: Hi all, I have a TDM2400P w/ echo cancellation, 8 FXSs and 8 FXOs. They are installed respectively on banks 1,2,5 and 6. The problem I am having is that when I make a call using the ZAP channel, I can hear perfectly but the person on the other end is hearing my voice with lots of ticks. It would seem I am making this call over a very bad bandwidth which is not the case since this is the PSTN. My configuration files are below, I have the latest versions of Zaptel, Libpri and Asterisk. I am using Polycomâs IP301 and IP430 Phones. I would appreciate help since I have to put this in production on Saturday. Well, if that weren't an analog card, I'd say it sounded like clock slip. It could be digital clipping/overdrive; you might check your gains. You have FXS, FXO, and SIP channels, there; which combinations cause the clicking in the transmit audio? Does it happen from FXS to FXO? SIP to FXO? SIP to SIP? How frequently, and how regularly, are the ticks? How loud? How sharp? Can you call someone with audio experience to describe them to you? (If no one else, feel free to call me; I'm good at this stuff... ;-) Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looped message playback
Depending on the format of your audio file, you could generate a
one-second sample of audio and then something like the following
#!/bin/bash
NUM=$1
CUR=0
rm -f bigtonefile;
[ $CUR != $NUM ] {
cat tonefile bigtonefile;
CUR = $CUR+1;
}
System(generator 7)
Then Playback(bigtonefile) ; for seven seconds of audio
You could use half-second tones or less if you wanted finer granularity.
Just a suggestion, think outside the box, they say :)
Earle Clubb wrote:
Bill Gibbs wrote:
Why not just merge the file together a few times using an audio program
and make a longer file?
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Earle
Clubb
Sent: Thursday, September 21, 2006 9:45 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Looped message playback
Hello,
I'm trying to play an audio file to a phone an arbitrary number of
times. The audio is a five-second segment of a sine wave. I need this
to be played repeatedly without gaps between playbacks. I've tried
doing this in the dial plan, e.g.:
exten = s,1,Playback(tonefile)
exten = s,2,Goto(1)
but there is too long of a gap between the playbacks. Does anyone know
of a way to achieve this?
Thanks,
Earle
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The duration of the tone can vary at runtime and I have no way of
knowing beforehand what the duration will be.
Earle
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!DSPAM:500,4512b29352141804284693!
--
Mojo [EMAIL PROTECTED]
Office Manager, Horan Company, LLC
(907) 747- x112
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Re: [asterisk-users] asterisk, iaxmodem, hylafax quality problem
Artifex Maximus wrote: Everything is fine when caller use ECM but when ECM isn't in use I often got unusable incoming faxes (much often that it should be). But when I switch back to fax machine that receive faxes perfectly (at least no visible error). The fax machine itself uses ECM, undoubtedly. If callers that have quality problems with IAXmodem+HylaFAX don't have problems with the fax machine, then that strongly indicates that something is wrong with your Asterisk setup... corrupting the audio. Usually this is due to resource constriction of the Zap device, zttest scores less than 99.98% is indicative of that situation. Where should be the problem? Is there any solution for improving quality? Any tuning in Asterisk or Hylafax? As I see some people use slinear for iaxmodem and some user use alaw. Which is better? There is no functional difference between using uLaw, alaw, or slinear... except that using slinear reduces the need for conversion... and thus possibly lessens CPU usage very slightly. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] notransfer local channel on redirect
Hello! On redirects(Got SIP response 302 Moved Temporarily) calls are sent to a local-channel in the sip-context. The billsec/duration is written to the parent record. however, i would like it to write them to the record of the local-channel. Is there a way to tell asterisk to add the notransfer-option(\n) on redirected calls? thanks Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call is dead after featuredigittimeout
For testing purposes, I have a Billion USB adapter connected to our PBX (P2P) and a cheap SIP phone (BT 101). Most things work, but I have a problem with the * key on any phone that may transfer calls because of the t or T option in extensions.conf (now try to google for an answer for a problem with * in asterisk :-)): If I press the * on a phone that might transfer a call, the call is dead after featuredigittimeout passes, no side can hear the other side, and no dtmf-codes have any effect. The only thing you can still do is to hang up. If I call from mISDn to SIP and then hang up the ISDN phone, I get Sep 21 17:41:05 WARNING[15656]: res_features.c:1384 ast_bridge_call: Bridge failed on channels mISDN/1-1 and SIP/bt101-081c3830 If I hang up the SIP phone instead, I get Sep 21 17:41:28 WARNING[15668]: indications.c:150 playtones_generator: Can't generate that much data! Sep 21 17:41:28 WARNING[15668]: res_features.c:1384 ast_bridge_call: Bridge failed on channels mISDN/1-1 and SIP/bt101-081f4fb0 If I press 2 fast enough after *, I get an attended transfer, and if I press any other digit within the timeout, nothing happens and the call can continue. This seems not to be a driver issue, it happens on calls misdn - SIP misdn - misdn SIP - misdn SIP - IAX2 misdn - IAX2 IAX2 - SIP I use asterisk SVN-branch-1.2-r43314M. The features.conf is trivial: - [general] language=de parkext = 700 ; What extension to dial to park parkpos = 701-720 ; What extensions to park calls on. context = parkedcalls ; Which context parked calls are in [featuremap] blindxfer = # ; Blind transfer atxfer = *2; Attended transfer [applicationmap] -- (If I change blindxfer to #2 and atxfer to *, I get the same problems with #.) misdn.log contains something like Thu Sep 21 15:09:21 2006: P[ 1] Transmitting 128 samples 2 misdn Thu Sep 21 15:09:21 2006: P[ 1] writing 128 bytes 2 asterisk Thu Sep 21 15:09:21 2006: P[ 0] misdn_jb_empty: read:128 | Bufferstatus:20 p:8137390 Thu Sep 21 15:09:21 2006: P[ 1] Transmitting 128 samples 2 misdn Thu Sep 21 15:09:21 2006: P[ 1] writing 128 bytes 2 asterisk Thu Sep 21 15:09:21 2006: P[ 1] Jitterbuffer Underrun. Thu Sep 21 15:09:21 2006: P[ 1] Transmitting 20 samples 2 misdn Thu Sep 21 15:09:21 2006: P[ 1] writing 128 bytes 2 asterisk Thu Sep 21 15:09:21 2006: P[ 0] misdn_jb_empty: Wait...requested:128 p:8137390 Thu Sep 21 15:09:21 2006: P[ 1] Transmitting 128 samples 2 misdn Thu Sep 21 15:09:21 2006: P[ 1] writing 128 bytes 2 asterisk Thu Sep 21 15:09:21 2006: P[ 0] misdn_jb_empty: Wait...requested:128 p:8137390 Thu Sep 21 15:09:21 2006: P[ 1] Transmitting 128 samples 2 misdn Thu Sep 21 15:09:21 2006: P[ 1] writing 128 bytes 2 asterisk when * is pressed, and then eight seconds later Thu Sep 21 15:09:29 2006: P[ 0] misdn_jb_empty: Wait...requested:128 p:8137390 Thu Sep 21 15:09:29 2006: P[ 1] Transmitting 128 samples 2 misdn Thu Sep 21 15:09:29 2006: P[ 1] writing 128 bytes 2 asterisk Thu Sep 21 15:09:29 2006: P[ 0] misdn_jb_empty: Wait...requested:128 p:8137390 Thu Sep 21 15:09:29 2006: P[ 1] Transmitting 128 samples 2 misdn Thu Sep 21 15:09:29 2006: P[ 1] writing 128 bytes 2 asterisk Thu Sep 21 15:09:29 2006: P[ 0] misdn_jb_empty: Wait...requested:128 p:8137390 Thu Sep 21 15:09:29 2006: P[ 1] Transmitting 128 samples 2 misdn Thu Sep 21 15:09:29 2006: P[ 1] Select Timed out Thu Sep 21 15:09:29 2006: P[ 0] misdn_jb_empty: Wait...requested:128 p:8137390 Thu Sep 21 15:09:29 2006: P[ 1] Transmitting 128 samples 2 misdn Thu Sep 21 15:09:29 2006: P[ 1] Select Timed out Any hints? Yours, Florian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: notransfer local channel on redirect
2006/9/21, Benko [EMAIL PROTECTED]: notransfer-option(\n) on redirected calls? sorry, it is called no release quote:(the n stands for no release) so is there a way to tell asterisk to not release a local channel on a redirect so the billsec and duration is written to it? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 650 Question
Hi I just read about the new Polycom 650 being released in a few months. http://www.vonmag.com/webexclusives/2006/09/20_IP_Phones_Get_Boosted.asp and integration with Microsoft Office Communicator IM client. This caught my eye, anyone know what functionality is available via this link? Or even better how it may related to Asterisk and what functionality we can drive with it? Regards, Dean Collins Cognation Pty Ltd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CURL
Jerry,Using system, is there a way to read into an asterisk variable the content of the response (i.e. a text in an http response) ?. If this is possible then I can use this approach.ThanksJerry Geis [EMAIL PROTECTED] wrote: You can always use the System() command in asterisk to call the curl executable.jerry---Ok, after requesting information to digium (no answer yet) and being informed that asterisk-dev is *NOT* a support hot line, I am trying in this list to see if someone has information on this regard.I know this is not a support area so I am only trying to get some clues.I have asterisk be and I am trying to use the CURL function (or application?). It is not available when I try it even though it is documented. Does anyone knows if there is a way to load it as a function/application inside asterisk? if so, is there code to download/compile to get it working inside asterisk?Any clue will be highly appreciated. (I keep trying digium support).ElpidioElpidio Ramos President RM International Services SA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office+52 (55) 5116-9805 Fax +52 (55) 1755-6601 CellUSA: +1 (801) 494-1415 Office+1 (240) 250-8264 Fax+1 (801) 938-4740 Direct___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Elpidio Ramos PresidentRM International ServicesSA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax+52 (55)1755-6601 CellUSA: +1 (801) 494-1415 Office +1 (240) 250-8264 Fax +1 (801) 938-4740Direct ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys SPA400
Does anyone know if the Linksys SPA400 is compatible with Asterisk or is it only for the SPA9000 system? It is interesting because it is a 4 FXO ATA at a reasonable price. -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Panasonic D500
I am integrating an Asterisk server with a Panasonic D500 system. We have it set up so Asterisk receives an E1 from the phone company and also several SIP lines. All these come into several menus depending on DID. The E1 to the phone company is using MFC/R2 and so is the E1 connecting the Panasonic and Asterisk. We basically have everything working except that whenever I dial from the Asterisk to the Panasonic using the E1 the only extension that ever rings is the operator. No matter what DID I send to the Panasonic only that extension will ring. If I dial an invalid extension I get a busy tone. I am assuming that either Asterisk is not sending something to the Panasonic or that the Panasonic is not interpreting the DID correctly and it defaults to the operator. Anyone have any experience integrating Asterisk and Panasonic? -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400P
Robson Ribeiro wrote: Hi all, I have a TDM2400P w/ echo cancellation, 8 FXSs and 8 FXOs. They are installed respectively on banks 1,2,5 and 6. The problem I am having is that when I make a call using the ZAP channel, I can hear perfectly but the person on the other end is hearing my voice with lots of ticks. It would seem I am making this call over a very bad bandwidth which is not the case since this is the PSTN. My configuration files are below, I have the latest versions of Zaptel, Libpri and Asterisk. I am using Polycom's IP301 and IP430 Phones. I would appreciate help since I have to put this in production on Saturday. Sounds to me like an the Digium card is sharing an IRQ with something else. cat /proc/interrupts ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX or SIP termination provider that reaches 6421xxxxxxx?
Is there anybody else out there that can terminate to 6421xxx? It's mobile, New Zealand Vodafone. I'd like to consider all options. Thanks Mojo with Horan Company, LLC wrote: Hi, my asterisk is set up with a pay-as-you-go Teliax account, and can dial out just fine to most numbers, but this cell phone number in New Zealand, 6421xxx, just rings and rings. Teliax support says: Unfortunately, not all International Cell numbers can be dialed by Teliax users. There is a problem from the receiving side. They have restricted us. We may appear as a solicitor to them and that is the way they take the call. If this works from a land line or pots line, that may be the case. This does work from a pots line. Do any list members know of a SIP or IAX termination providers that can call this country/city code combination? the city code, assigned to Vodafone (mobile/wireless?) is 21. Thanks! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM2400P
Dear Jay, maybe I would better describe the sound as breaking and not skipping. It is a constant thing so the person on the other side can't understand a word. It's like when you are in a bad cellphone connection. It ONLY happens and this is the weird part, when I call OUT of the TDM. When someone call IN nothing happens. The call is originating as a ZAP call on a FXSs channel and going directly to the PSTN. Now, I tried working with TX/RX But it didn’t make any difference as the issue doesn’t seem to matter if gain is higher or lower. If I was calling from a VOIP provider I could understand this as being a bandwidth issue. But from the PSTN to another PSTN it is very strange indeed. I tried calling you but noone answered. Will try later. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth Sent: Thursday, September 21, 2006 1:23 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] TDM2400P On Thu, Sep 21, 2006 at 12:28:16PM -0400, Robson Ribeiro wrote: Hi all, I have a TDM2400P w/ echo cancellation, 8 FXSs and 8 FXOs. They are installed respectively on banks 1,2,5 and 6. The problem I am having is that when I make a call using the ZAP channel, I can hear perfectly but the person on the other end is hearing my voice with lots of ticks. It would seem I am making this call over a very bad bandwidth which is not the case since this is the PSTN. My configuration files are below, I have the latest versions of Zaptel, Libpri and Asterisk. I am using Polycom’s IP301 and IP430 Phones. I would appreciate help since I have to put this in production on Saturday. Well, if that weren't an analog card, I'd say it sounded like clock slip. It could be digital clipping/overdrive; you might check your gains. You have FXS, FXO, and SIP channels, there; which combinations cause the clicking in the transmit audio? Does it happen from FXS to FXO? SIP to FXO? SIP to SIP? How frequently, and how regularly, are the ticks? How loud? How sharp? Can you call someone with audio experience to describe them to you? (If no one else, feel free to call me; I'm good at this stuff... ;-) Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looped message playback
Mojo with Horan Company, LLC wrote:
Depending on the format of your audio file, you could generate a
one-second sample of audio and then something like the following
#!/bin/bash
NUM=$1
CUR=0
rm -f bigtonefile;
[ $CUR != $NUM ] {
cat tonefile bigtonefile;
CUR = $CUR+1;
}
System(generator 7)
Then Playback(bigtonefile) ; for seven seconds of audio
You could use half-second tones or less if you wanted finer granularity.
Just a suggestion, think outside the box, they say :)
Earle Clubb wrote:
Bill Gibbs wrote:
Why not just merge the file together a few times using an audio program
and make a longer file?
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Earle
Clubb
Sent: Thursday, September 21, 2006 9:45 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Looped message playback
Hello,
I'm trying to play an audio file to a phone an arbitrary number of
times. The audio is a five-second segment of a sine wave. I need
this to be played repeatedly without gaps between playbacks. I've
tried doing this in the dial plan, e.g.:
exten = s,1,Playback(tonefile)
exten = s,2,Goto(1)
but there is too long of a gap between the playbacks. Does anyone
know of a way to achieve this?
Thanks,
Earle
___
Great idea. Unfortunately I may never know the duration of the tone
until after it is turned off. For example if switch is turned on, the
tone should begin. It should keep playing until the switch is turned off.
Earle
___
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[asterisk-users] RE: Setting QOS settings in asterisk and/or CentOS?
I use the following in rc.local for setting tos bits using iptables:iptables -A POSTROUTING -t mangle -p udp -m udp --sport 1:2 -j DSCP --set-dscp 0x2eWorks like a champ! 1.RE:SettingQOSsettingsinasteriskand/orCentOS? (RedouaneDoumer) De:BerkHolz,Steven[mailto:[EMAIL PROTECTED] Envoyé:jeudi21septembre200615:33 À:asterisk-users@lists.digium.com Objet:[asterisk-users]SettingQOSsettingsinasteriskand/orCentOS? HowwouldIgoaboutsettingtheTOSbitto"RTPIPTOSByte:18(hex)"forSIPandIAXtrafficattheasteriskserver? Also, DoyouhaveaquickreferenceonhowtoconfigureaCiscoswitchtoprioritizeSIPtraffic? IcheckinvariousCiscodocs,andtherearesomanyreferences,andnoneofthemseemtorelatedirectlytousingtheTOSbitforQOS. IamlookingintousingtheTOSbitbecausethatistheonlymethodthatmySIPdevicesuse.(CitelHandsetGateway) ref: QOSsettingsfromCitelHandsetGateway: HandsetGateway-QoSConfiguration IPTypeofService RTPIPTOSByte:18(hex) SilenceSuppression MuteMode:On,UDPkeep-aliveevery10seconds G.711VoiceActivityDetection:Off CodecPreferences G.711u:1(Highestpriority) G.711a:2ThankYou, StevenBerkHolz -MCSA-MCSE- ManagerofInformationSystems TESCOGroupCompanies Fax.248-836-5101 www.TESCOGroup.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asteisk plays music on hold startingfrom randompoint
Giorgio, I copied this snippet from the web: cd asterisk-addons/format_mp3 if your asterisk is running: make autoload if not: make install Robert Chadwell 800-330-7704 toll free 813-343-0181 ph 813-413-8195 fx Please feel free to IM me as well AOL Screenname: cmgrobert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo Sent: Thursday, September 21, 2006 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asteisk plays music on hold startingfrom randompoint Hi Robert, cpu is no problem..but if you say the sound is cleaner..well I can try Format_MP3...the problem is I do not know how to install it. I read it is inside asterisk-addons (there is a directory named format_mp3 inside) and I always install asterisk-addons on PBXs but...how can I invoke this command?? Is it automatic?? TIA Giorgio Incantalupo Robert Chadwell wrote: Giorgio, This format works just how you want it to - it will play the files starting at the beginning. If you convert your files to .wav or some other format - you will get a cleaner sound. From what I have read, Fedora opens a single instance of the file regardless of the number accessing it, using Format_MP3 (the native MOH player in Asterisk 1.2 that replaced mpg123) - so you should save on CPU usage using this program instead of mpg123. From the post that I read, the person was testing it using a .wav file. Here is the copied musiconhold.conf setup [default] mode=files directory=/var/lib/asterisk/moh-native random=yes ; Play the files in a random order Robert Chadwell 800-330-7704 toll free 813-343-0181 ph 813-413-8195 fx Please feel free to IM me as well AOL Screenname: cmgrobert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo Sent: Wednesday, September 20, 2006 12:19 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asteisk plays music on hold starting from randompoint Hi, I'm using mpg123 to play music on hold but it seems that Asterisk does play the music from a random point: is there a way to make my music on hold always starting from beginning? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Format_MP3, Streaming, File Formats, MOH
Fred, A glimmer of hope! What version of mpg123 do you have running? I am guessing that you control the bitrate on your internal Shoutcast server, is that right? Robert Chadwell 800-330-7704 toll free 813-343-0181 ph 813-413-8195 fx Please feel free to IM me as well AOL Screenname: cmgrobert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Frédéric Marti Sent: Thursday, September 21, 2006 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Format_MP3, Streaming, File Formats, MOH Hi, Yes , I use Asterisk 1.2.10 But , I don't have Warning Flexible rate not heavily tested notices in the Asterisk CLI The Shoutcast server is in the same box as Asterisk, and the stream source is in the same LAN Regards Fred ___ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Chadwell Sent: jeudi, 21. septembre 2006 15:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Format_MP3, Streaming, File Formats, MOH Frederic, Did this work for you under Asterisk 1.2x? If it did, did you receive Warning Flexible rate not heavily tested notices in the Asterisk CLI? Robert Chadwell 800-330-7704 toll free 813-343-0181 ph 813-413-8195 fx Please feel free to IM me as well AOL Screenname: cmgrobert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk [Submusic] Sent: Wednesday, September 20, 2006 9:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Format_MP3, Streaming, File Formats, MOH Hi, This config is working for me: _ musiconhold.conf [shoutcast] mode=custom application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 http://stream128.submusic.ch:8004/ ; The '/' in the stream URL is important ! _ extensions.conf exten = 17,1,Answer exten = 17,2,MusicOnHold(shoutcast) _ Regards Frederic De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Robert Chadwell Envoyé : mardi, 19. septembre 2006 14:47 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Format_MP3, Streaming, File Formats, MOH Format_MP3 appears to play MOH files starting at the beginning of each file, using the .wav file format, making for some repetitive hold music unless you alter the file itself to begin somewhere in the middle. Solution: One stream that all users connect to - giving dynamic hold music (tried and tested in A1.0x using mpg123 with some success, and Icecast or Slimserver or Shoutcast) Format_MP3 doesn't seem to stream, and the wiki is wrong about streamplayer being used to play streams, as it is only used to play raw TCP streams. There are many questions in forums on the web with no answers about how to solve this dilemma, How do you get users connected to a constantly-changing stream of music instead of streams starting from the beginning (regardless of whether Linux counts them as one stream or not where the processor is concerned)? Hopefully, at the end of this thread, I will have enough information to go back to these web-forums and post the answer. To get it started - here is what I have tried that hasn't worked. In most all cases the response is Music on hold started, immediately followed by Music on hold stopped with no sound in any case. ;[classes] ;mode=custom ;application=/usr/bin/streamplayer 194.158.114.67 8000 ;format=ulaw --- Straight From The Music On Hold Wiki ;default = quietmp3:/var/lib/asterisk/mohmp3-dummy -@,http://www.shoutcast.com/sbin/tunein-station.pls?id=7733filename=playlist.pls --- From the Nerd Vittles Tutorial with the -@ added because mpg123 seemed to ask for it since the file was a .pls ;default = mp3:http://127.0.0.1:9000/stream.mp3 -- From a forum of someone using mpg123 to stream SlimServer (installed mpg123 v0.60 with no success here) [default] mode=files directory= /var/lib/asterisk/mohmp3-dummy,http://127.0.0.1:9000/stream.mp3 -- Tried a 1.2 format ;default = quietmp3:/var/lib/asterisk/mohmp3-dummy,http://193.251.154.243:8000/ -- Thought maybe it was SlimServer - so tried to stream the top Shoutcast station ;default = quietmp3:/var/lib/asterisk/mohmp3-dummy,http://127.0.0.1:9000/stream.mp3 -- Tried to stream Slimserver using the old format Thank you in advance - I have been at this for a week now. How did you make it work in Asterisk 1.2x? Rob ___
RE: [asterisk-users] CURL
You need to use AGI to do this. You would put a shell script yourscript.agi in /var/lib/asterisk/agi-bin If you want an HTTP response dumped into your dialplan as a variable, you would use wget: myagi.agi: #!/bin/bash TMPFILE=/tmp/$$.$RANDOM wget -q -t3 --output-document=$TMPFILE http://mysite.com/myhtml.html MYVAR=$(cat $TMPFILE) echo "SET VARIABLE MYASTERISKVAR \"$MYVAR\"" rm $TMPFILE You would call the AGI like this: exten = foo,1,AGI(/var/lib/asterisk/agi-bin/myagi.agi) Optionally you can pass a parameter to the AGI: exten = foo,1,AGI(/var/lib/asterisk/agi-bin/myagi.agi foo) and you would retrieve the parameter in the AGI using the $1, $2, $3 variables in the AGI for parameter 1,2,3 etc. hth -Original Message-From: Elpidio Ramos [mailto:[EMAIL PROTECTED]Sent: Thursday, September 21, 2006 10:52 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] CURL Jerry, Using system, is there a way to read into an asterisk variable the content of the response (i.e. a text in an http response) ?. If this is possible then I can use this approach. ThanksJerry Geis [EMAIL PROTECTED] wrote: You can always use the System() command in asterisk to call the curl executable.jerry---Ok, after requesting information to digium (no answer yet) and being informed that asterisk-dev is *NOT* a support hot line, I am trying in this list to see if someone has information on this regard.I know this is not a support area so I am only trying to get some clues.I have asterisk be and I am trying to use the CURL function (or application?). It is not available when I try it even though it is documented. Does anyone knows if there is a way to load it as a function/application inside asterisk? if so, is there code to download/compile to get it working inside asterisk?Any clue will be highly appreciated. (I keep trying digium support).ElpidioElpidio Ramos President RM International Services SA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office+52 (55) 5116-9805 Fax +52 (55) 1755-6601 CellUSA: +1 (801) 494-1415 Office+1 (240) 250-8264 Fax+1 (801) 938-4740 Direct___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Elpidio Ramos PresidentRM International ServicesSA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax +52 (55)1755-6601 CellUSA: +1 (801) 494-1415 Office +1 (240) 250-8264 Fax +1 (801) 938-4740Direct ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400P
Indeed there is something very strange here: look how the PC is recognizing the Digium boardIs this normal?also i have noticed that both the IVR and Musiconhold seem to be acceleratedafter lspci i get: :04:09.0 Ethernet controller: Digium, Inc.: Unknown device 2400 (rev 11)On 9/21/06, Robson Ribeiro [EMAIL PROTECTED] wrote:Dear Jay, maybe I would better describe the sound as breaking and not skipping. It is a constant thing so the person on the other side can't understand a word. It's like when you are in a bad cellphone connection. It ONLY happens and this is the weird part, when I call OUT of the TDM. When someone call IN nothing happens. The call is originating as a ZAP call on a FXSs channel and going directly to the PSTN. Now, I tried working with TX/RX But it didn't make any difference as the issue doesn't seem to matter if gain is higher or lower. If I was calling from a VOIP provider I could understand this as being a bandwidth issue. But from the PSTN to another PSTN it is very strange indeed. I tried calling you but noone answered. Will try later. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Jay R. AshworthSent: Thursday, September 21, 2006 1:23 PMTo: asterisk-users@lists.digium.comSubject: Re: [asterisk-users] TDM2400P On Thu, Sep 21, 2006 at 12:28:16PM -0400, Robson Ribeiro wrote:Hi all, I have a TDM2400P w/ echo cancellation, 8 FXSs and 8 FXOs.They are installed respectively on banks 1,2,5 and 6. The problem I am having is that when I make a call using the ZAP channel, Ican hear perfectly but the person on the other end is hearing myvoice with lots of ticks. It would seem I am making this call over a very bad bandwidth which is not the case since this is thePSTN. My configuration files are below, I have the latest versionsof Zaptel, Libpri and Asterisk. I am using Polycom’s IP301 and IP430 Phones. I would appreciate help since I have to put this inproduction on Saturday.Well, if that weren't an analog card, I'd say it sounded like clockslip.It could be digital clipping/overdrive; you might check your gains. You have FXS, FXO, and SIP channels, there; which combinations causethe clicking in the transmit audio?Does it happen from FXS to FXO?SIP to FXO?SIP to SIP?How frequently, and how regularly, are the ticks?How loud?How sharp?Can you call someone with audio experience to describe them toyou?(If no one else, feel free to call me; I'm good at this stuff...;-)Cheers,-- jra--Jay R. Ashworth [EMAIL PROTECTED]DesignerBaylink RFC 2100Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USAhttp://baylink.pitas.com +1 727 647 1274That's women for you; you divorce them, and 10 years later,they stop having sex with you.-- Jennifer Crusie; _Fast_Women_ ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Linksys SPA400
It's only designed for use with the SPA-9000 (LVS-9000) product ecosystem. Cory Andrews -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez Sent: Thursday, September 21, 2006 1:00 PM To: Asterisk Subject: [asterisk-users] Linksys SPA400 Does anyone know if the Linksys SPA400 is compatible with Asterisk or is it only for the SPA9000 system? It is interesting because it is a 4 FXO ATA at a reasonable price. -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
