Re: [Asterisk-Users] unsubscribe
Read The Manual Before Asking!! Indeed: To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QUICK QUESTION
How can I have asterisk ignore incoming rings so it doesn't answer a specific line. I tried setting up an empty context section but that didn't work. Make a long delay the first line of the phone's context. This can even be turned on and off using a few more lines. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Quad Span Cards
On Wed, 27 Apr 2005, Adam Goryachev wrote: Just wondering, but does the AMD multi CPU architecture improve the interrupt handling? My understanding of that architecture is that each CPU can deal with it's own PCI bus/interrupts/etc independently of each other, and also with their own memory/etc? Would this improve the scalability? In fact, would a multi-PCI bus system by itself 'solve' the problem? Beware that not all multi cpu Opteron motherboards are created equal. Quite a few connect all their pci busses to one cpu. A good motherboard will distribute the pci busses across the cpus. Read http://www.samag.com/documents/s=9408/sam0411b/0411b.htm for good and bad examples and a list of things to watch out for when purchasing an Opteron system. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call a peer over the asterisk manager with a php script
Hello,
I want to call a peer over the Asterisk Manager with this php-script:
htmlbodyPRE?$socket = fsockopen("192.168.204.44","5038", $errno, $errstr, $timeout);fputs($socket, "Action: Login\r\n");fputs($socket, "UserName: test\r\n");fputs($socket, "Secret: test\r\n\r\n");//fputs($socket, "Action: ListCommands\r\n\r\n");fputs($socket, "Action: Originate\r\n");fputs($socket, "Channel: 6159bfb47b9\r\n\r\n");fputs($socket, "Exten: 1009\r\n\r\n");fputs($socket, "Context: test\r\n\r\n");fputs($socket, "Priority: 1\r\n\r\n");fputs($socket, "Action: Logoff\r\n\r\n");while (!feof($socket)) {$wrets .= fread($socket, 8192);}fclose($socket);echo ASTERISKMANAGERENDASTERISK MANAGER OUTPUT:$wretsASTERISKMANAGEREND;?/pre
I got this resulat:
ASTERISK MANAGER OUTPUT:
Asterisk Call Manager/1.0
Response: Success
Message: Authentication accepted
Response: Error
Message: Invalid channel
Response: Error
Message: Missing action in request
Response: Error
Message: Missing action in request
Response: Error
Message: Missing action in request
Response: Goodbye
Message: Thanks for all the fish.
I tried many diffrent SIP/Channels but nothing works
THX
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[Asterisk-Users] RTP vs cRTP vs IAX
Hi List, I have seen this: http://www.convergence.com.pk/iax2/trunked.html According to this table, using trunking, you can have 16 channels with 171.7 kbps bandwith using g.729 + IAX2 trunking? Sounds too good to be true... Any comments on this? Cheers, Jean-Michel. -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap/PRI: received AOC-E charging
On 4/26/05, Matthew Boehm [EMAIL PROTECTED] wrote: Trying to make a call via our PRI: (CVS everything, CVS-NHEAD-04/23/05-16:08:12) -- Executing Dial(IAX2/[EMAIL PROTECTED], Zap/R2/2815699900|30) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called R2/2815699900 -- Channel 0/19, span 2 got hangup -- Channel 0/19, span 2 received AOC-E charging 0 units Apr 26 09:06:49 WARNING[10040]: chan_zap.c:7457 zt_pri_error: PRI: Call From Your debug Ext: 1 Cause: Temporary failure (41), class = Network Congestion (2) ] Looks like either a number problem or no route to destination. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP, Asterisk and NAT
Irakli Natsvlishvili wrote: 100k question - does asterisk correctly handle following situations: There are plenty of good documents on Asterisk, SIP and NAT on the voip-info.org wiki. Please look them up. There are also information within the configs/sip.conf.sample file within Asterisk. 1. Asterisk is on a public IP Two SIP clients on separate networks, each of them are behind dynamic NAT gateway. Nat gateway does not have ALG. Media stream SHOULD NOT go thought asterisk. If the media stream SHOULD NOT go through Asterisk, then it's up to the phones to support NAT traversal properly and handle this, it's not an Asterisk problem. From Asterisk's point of view, we should not see that they are in fact behind NAT. Modern phones in combination with STUN and a decent NAT device supports this. 2. Even worst case - three clients, two of them on one site, second is on another site. For example extensions 500 and 600 are on the same site and in the same subnet and extension 1000 is on another site/network. There are PAT FW/gateways with dynamic public IP in front of clients and those are symmetric NAT/FW. The task - clients registering on Asterisk server, calling each other and RTP should not go via asterisk. So, media stream should go directly from one client to another. If Asterisk is on a public IP, again: it's up to the phones. It's still not an Asterisk problem. I want to know: 1. Is it possible? - yes/no. Implementation should involve asterisk and SIP clients and not involving third party hardware products - ALG, session border controllers or so on. Yes, but you need to pick the right phone, the right NAT/FW and have a lot of patience :-) 2. If it is possible, what are requirements for SIP clients. Good NAT traversal support. 3. What configuration changes should be done on Asterisk server and on a sip clients. From Asterisk's point of view, all of these phones are on a public IP and we do not give them any NAT traversal support. If you want detailed configurations, there are several consultants available that can help you with that (including my company). And final question - if it is NOT possible with Asterisk, do you know an open source product which works in above stated scenarios and you've actually tested it. It is possible with Asterisk and every other SIP server. With your requirements, it's completely a client-side problem. Best regards, /Olle ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fail over solutions
On 4/26/05, snacktime [EMAIL PROTECTED] wrote: On 4/26/05, Sean Kennedy [EMAIL PROTECTED] wrote: Hi folks, I'm curious; What does everyone do for failover? I have two servers, same os/compilation. I designate one the master, the other the slave, and I rsync the config files once an hour and trigger a restart when convenient command on the console. These two servers are setup in the dns in a round robin fashion. What is everyone else doing? That's kind of a loaded question... Do you plan on expanding? What is your budget? What are your uptime requirements? Are you serving customers or is this just for internal use? The biggest problem with that solution is voicemail it could get left on one server and not be on the other for one hour. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call a ldap result via my x-lite
Hello, Question: I use x-lite as softphone and I want to call with my softphone a peer who is the result of my ldap search. Has someone an idea how I can fixe this problem?? THX Gesendet von Yahoo! Mail - Jetzt mit 250MB kostenlosem Speicher___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queue Management and Command Execution
You can record queue conversations, check out the configs in queue.conf |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Daniel Salama |Sent: Martes, 26 de Abril de 2005 05:53 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: [Asterisk-Users] Queue Management and Command Execution | |Is there a way to execute a command prior to sending a queued |call to an agent? | |What I'm trying to do is record agent's conversation at the |server. I could put the Monitor command when the call is |answered by *, but if the caller has to wait on hold for some |time, I wouldn't want to record that. | |What I would prefer is to be able to put the caller in a queue |and once an agent is ready, for the Monitor command to kick |in. Even better would be to know the agent id where the call |is going to be sent to, so I can use it as part of the file |name of the Monitor command. | |Any clues on how to do this, if at all possible? | |Thanks, |Daniel | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] YAC and IPs
Not a bad idea at all! |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |William Suffill |Sent: Martes, 26 de Abril de 2005 07:02 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] YAC and IPs | |Why not DBGet from the SIP.Registry in ASTDB? Wouldn't that be |cleaner since it would only return the 1 you want instead of |parsing what could be a load of sip peers? | |-- William |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP vs cRTP vs IAX
Go have a look at http://www.asteriskguru.com/tool2.php and calculate it for yourself. This is without the signalling frames for call setup / teardown. (bandwidth used by those is very small). Greetz, /Z Incoming Bandwidth Outgoing Bandwidth Calls: 16 Calls: 16 RTP: 2.34 Kbps RTP: 2.34 Kbps UDP: 3.13 Kbps UDP: 3.13 Kbps IP: 4.69 Kibps IP: 4.69 Kibps Protocol: IAX2 TRUNKED Protocol: IAX2 TRUNKED Audio Codec: 8.00 Kbps Audio Codec: 8.00 Kbps *IAX2 TRUNKED is not using RTP or RTCP! *IAX2 TRUNKED is not using RTP or RTCP! Incoming bandwidth used is: *139.72 Kbps* *0.14 Mbps* *17.47 KBps* *0.02 MBps* Outgoing bandwidth used is: *139.72 Kibps* *0.14 Mbps* *17.47 KBps* *0.02 MBps* Total bidirectional bandwidth used (incoming and outgoing) is: *279.44 Kbps* *0.27 Mbps* *34.93 KBps* *0.03 MBps* Jean-Michel Hiver wrote: Hi List, I have seen this: http://www.convergence.com.pk/iax2/trunked.html According to this table, using trunking, you can have 16 channels with 171.7 kbps bandwith using g.729 + IAX2 trunking? Sounds too good to be true... Any comments on this? Cheers, Jean-Michel. signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP, Asterisk and NAT
Im trying to write some tutorial for these ever recurring SIP + NAT questions. Its far from ready, and its without layout, but the draft can be found at: http://www.asteriskguru.com/natut.php it has all most of the situations explained, and explains all the options you need to look at in the asterisk config files. /Z Olle E. Johansson wrote: Irakli Natsvlishvili wrote: 100k question - does asterisk correctly handle following situations: There are plenty of good documents on Asterisk, SIP and NAT on the voip-info.org wiki. Please look them up. There are also information within the configs/sip.conf.sample file within Asterisk. 1. Asterisk is on a public IP Two SIP clients on separate networks, each of them are behind dynamic NAT gateway. Nat gateway does not have ALG. Media stream SHOULD NOT go thought asterisk. If the media stream SHOULD NOT go through Asterisk, then it's up to the phones to support NAT traversal properly and handle this, it's not an Asterisk problem. From Asterisk's point of view, we should not see that they are in fact behind NAT. Modern phones in combination with STUN and a decent NAT device supports this. 2. Even worst case - three clients, two of them on one site, second is on another site. For example extensions 500 and 600 are on the same site and in the same subnet and extension 1000 is on another site/network. There are PAT FW/gateways with dynamic public IP in front of clients and those are symmetric NAT/FW. The task - clients registering on Asterisk server, calling each other and RTP should not go via asterisk. So, media stream should go directly from one client to another. If Asterisk is on a public IP, again: it's up to the phones. It's still not an Asterisk problem. I want to know: 1. Is it possible? - yes/no. Implementation should involve asterisk and SIP clients and not involving third party hardware products - ALG, session border controllers or so on. Yes, but you need to pick the right phone, the right NAT/FW and have a lot of patience :-) 2. If it is possible, what are requirements for SIP clients. Good NAT traversal support. 3. What configuration changes should be done on Asterisk server and on a sip clients. From Asterisk's point of view, all of these phones are on a public IP and we do not give them any NAT traversal support. If you want detailed configurations, there are several consultants available that can help you with that (including my company). And final question - if it is NOT possible with Asterisk, do you know an open source product which works in above stated scenarios and you've actually tested it. It is possible with Asterisk and every other SIP server. With your requirements, it's completely a client-side problem. Best regards, /Olle ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fail over solutions
One thing that could be done is to have a disk array for voicemail and all with dual controllers. Then plug that into each of two servers. Bind the IP components to a IP that is transportable between machines. When one fails ifconfig the failover machine to use that IP (could be a virtual interface). Veritas HA works similiarly that way. Via a serial cable there are 'global atomic broadcasts' basically a ping. If the ping fails to occur the machine marked backup assumes the IP for all services of the primary. Because it has access to the same disks it can mount them and carry on like nothing happened. Veritas seperates services from the machine. If you have say a web server, mail, and SIP you would have each one on a seperate IP so that if any one single service fails that one and only that one can be moved to the backup server. With asterisk this may be overkill. MAC addresses are the only other problem. Veritas accomplishes this by MAC spoofing. Cisco PIX do as well. You might, depending on specific ethernet driver, be able to ifconfig eth0 headdr 00:00:de:ca:fb:ad. Just a thought. On Wed, 2005-04-27 at 08:33 +0100, Jason Williams wrote: On 4/26/05, snacktime [EMAIL PROTECTED] wrote: On 4/26/05, Sean Kennedy [EMAIL PROTECTED] wrote: Hi folks, I'm curious; What does everyone do for failover? I have two servers, same os/compilation. I designate one the master, the other the slave, and I rsync the config files once an hour and trigger a restart when convenient command on the console. These two servers are setup in the dns in a round robin fashion. What is everyone else doing? That's kind of a loaded question... Do you plan on expanding? What is your budget? What are your uptime requirements? Are you serving customers or is this just for internal use? The biggest problem with that solution is voicemail it could get left on one server and not be on the other for one hour. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fail over solutions
On Wed, 2005-04-27 at 00:52 -0700, trixter http://www.0xdecafbad.com wrote: One thing that could be done is to have a disk array for voicemail and all with dual controllers. Then plug that into each of two servers. Bind the IP components to a IP that is transportable between machines. When one fails ifconfig the failover machine to use that IP (could be a virtual interface). Veritas HA works similiarly that way. Via a serial cable there are 'global atomic broadcasts' basically a ping. If the ping fails to occur the machine marked backup assumes the IP for all services of the primary. Because it has access to the same disks it can mount them and carry on like nothing happened. Veritas seperates services from the machine. If you have say a web server, mail, and SIP you would have each one on a seperate IP so that if any one single service fails that one and only that one can be moved to the backup server. With asterisk this may be overkill. MAC addresses are the only other problem. Veritas accomplishes this by MAC spoofing. Cisco PIX do as well. You might, depending on specific ethernet driver, be able to ifconfig eth0 headdr 00:00:de:ca:fb:ad. Just a thought. I forgot to add that if you have T1/E1/J1s you would want a hunt group defined so that calls from one goto the other if the card is nonresponsive. Analogue lines can forward to a seperate machine on a 'no answer' basis. Of course if you are doing failover odds are you arent doing analogue lines. All in all this shouldnt be a terribly difficult solution to implement, and could even be done on 1U boxes or whatever. Basically a 'brain dead' add on package that requires little configuration, and then distributed by whatever means someone chooses (if they choose unwisely someone else will just write something similar that is distributed differently :) Due to the cost of asterisk this could be a feature that normal PBX systems do not have, or do not have for anything 'reasonably' priced. Giving yet another advantage to asterisk. The disk array would be the only expensive add on, more than a normal asterisk system. It all depends on how important voicemail is in your application, although there are cheaper alternatives (NFS for example, but then your NFS server becomes a single point of failure, depending on the disk array that same issue could be true there as well). -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fail over solutions
Could you explain me some more how i could use dual controllers ? Is this done with special harddisks ? What hardware do i need to do this ? /Z. trixter http://www.0xdecafbad.com wrote: One thing that could be done is to have a disk array for voicemail and all with dual controllers. Then plug that into each of two servers. Bind the IP components to a IP that is transportable between machines. When one fails ifconfig the failover machine to use that IP (could be a virtual interface). Veritas HA works similiarly that way. Via a serial cable there are 'global atomic broadcasts' basically a ping. If the ping fails to occur the machine marked backup assumes the IP for all services of the primary. Because it has access to the same disks it can mount them and carry on like nothing happened. Veritas seperates services from the machine. If you have say a web server, mail, and SIP you would have each one on a seperate IP so that if any one single service fails that one and only that one can be moved to the backup server. With asterisk this may be overkill. MAC addresses are the only other problem. Veritas accomplishes this by MAC spoofing. Cisco PIX do as well. You might, depending on specific ethernet driver, be able to ifconfig eth0 headdr 00:00:de:ca:fb:ad. Just a thought. On Wed, 2005-04-27 at 08:33 +0100, Jason Williams wrote: On 4/26/05, snacktime [EMAIL PROTECTED] wrote: On 4/26/05, Sean Kennedy [EMAIL PROTECTED] wrote: Hi folks, I'm curious; What does everyone do for failover? I have two servers, same os/compilation. I designate one the master, the other the slave, and I rsync the config files once an hour and trigger a restart when convenient command on the console. These two servers are setup in the dns in a round robin fashion. What is everyone else doing? That's kind of a loaded question... Do you plan on expanding? What is your budget? What are your uptime requirements? Are you serving customers or is this just for internal use? The biggest problem with that solution is voicemail it could get left on one server and not be on the other for one hour. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fail over solutions
On Wed, 2005-04-27 at 11:17 +0300, Zoa wrote: Could you explain me some more how i could use dual controllers ? Is this done with special harddisks ? What hardware do i need to do this ? We used a winchester drive array, which is not cheap, and way overkill for asterisk. EMC makes similar boxes. The one we had was a 19 inch cabinet and all drives were RAID. It came with integrated controllers each was dual ported so the machine could do 2x SCSI speeds, and there were 2 controllers integrated into the rack so both systems could benefit from this (ie 4 ports). I am unsure if there are smaller cheaper solutions, a multi-terabyte raid array would be underused for just voicemail unless you get a TON of voicemail, and I cant imagine asterisk being able to handle the clients that would require that. I would suggest googling multiport drive array I have not seen any ability to connect multiple controllers to the same disk, so you have to get a special controller that allows for this type of connectivity. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No music on hold when transferring call
MOH is working in that a defined extension works just fine:exten = 6000,1,Answerexten = 6000,2,MusicOnHold()musiconhold.conf is as per the default:[classes]default = quietmp3:/var/lib/asterisk/mohmp3,-zand zapata.conf and sip.conf havemusiconhold=default and musicclass=default respectively.However when I put a call on hold for transfer or just pressing the hold button there is no music. Normally I would expect to see something like the following, but nothing appears in the trace. --Started music on hold, class 'default', on SIP/4101-5ea9If a blind transfer is initiated the original caller gets hold music while the blind transfer is setup so I fear that something is back to front. All help gratefully received. Message sent using UebiMiau 2.7.2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] do I configure ISDN in zapata.conf?
I'm new to asterisk and still learning it. I wanted to ease my efforts a bit and use AMP (Asterisk Management Portal), and see what changed in the config files when I use it. However, I realized that I can only add SIP, IAX2 and ZAP extensions - I didn't see an option to configure an ISDN extension etc. So my conclusion was, that ZAP (zapata.conf) allows configuring ISDN extensions / numbers, too? Or am I totally wrong? If someone could make sme clarification about this, I'd be glad. Searching this list, wiki and google didn't bring me a definite answer. I have an Eicon DIVA 2.01 PCI ISDN card. Tomek ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Using Asterisk to dial a number and then wait to dial the extension
Andrew Elchuk [EMAIL PROTECTED] wrote: I did some searching and haven't found a solution to my problem. But right now we are performing a transition from an old system to a new system using asterisk and only a few people are on the new system and testing it out. Anyways, I was wondering if it is possible to dial a phone number with asterisk, and then after that callee picks up to dial an extension? In one forum message I found that you could use 'w' in the dial string to act as a half second wait. I tried doing: exten = 109,1,Dial(ZAP/g1/6525798ww109) This would dial the other phone system, but would not wait 3 seconds til the other system answered and then dial the extension. I also tried using: exten = 109,1,Dial(ZAP/g1/6525798|D(109)) But this did the same thing as the above. Is there another way to dial a number then on the same channel send 3 more digits after the other party answers? Thanks. The D() option is the correct way to do it, but only works if your Zap interface can tell when the remote party answers. Typically, digital lines (ISDN, T1, E1) can tell, but analogue lines can't. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] b0rked hfc config
I have 2 Billion cards and I can't get the hfc driver to work. I get this error: ZT_CHANCONFIG failed on channel 9: No such device or address (6) What am I doing wrong ? ztcfg -vvv gives me this: 88--8- Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: D-channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: D-channel (Default) (Slaves: 14) 6 channels configured. ZT_CHANCONFIG failed on channel 9: No such device or address (6) 88--8- This is my /etc/zaptel.conf: span=1,1,3,ccs,ami bchan=9-10 dchan=11 span=2,1,3,ccs,ami bchan=12-13 dchan=14 loadzone = us defaultzone=us (I'm using 1.07 of the zaptel driver with bristuff-0.2.0-RC8 and the patches from that package) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
S.NASROLLAHI hi i am a new member i want to learn what is TOS and LOG command in the access list and what are they doing? what is their advantage ? when i should use them? thank u ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7.4 SIP firmware
Hello; Does anybody know how can I get the Cisco 7.4 SIP firmware? Many Thanks Betul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No music on hold when transferring call
Bam wrote: MOH is working in that a defined extension works just fine: exten = 6000,1,Answer exten = 6000,2,MusicOnHold() I had some similar problems with asterisk v1.0.6, 1.0.7 solved this. (it had something to do with SIP and MOH) Cheers. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] b0rked hfc config
On Wed, Apr 27, 2005 at 11:08:46AM +0200, Thomas Andrews wrote: I have 2 Billion cards and I can't get the hfc driver to work. I get this error: ZT_CHANCONFIG failed on channel 9: No such device or address (6) What am I doing wrong ? This is my /etc/asterisk/zaptel.conf: [channels] switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=yes group = 1 context = incoming channel = 9 channel = 10 channel = 12 channel = 13 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Supervised transfer problem.
Hi all.
I am new in the list and i believe i have read enough to run an asterisk
pbx good, but i have a problem.
My instalation is enterely SIP based and i am trying now with
grandstream budge tone 102 because with x-lite softphone i cannot get
transfer, supervised or not, be fine.
Few question:
Is supervised transfer supported by SIP channel in 1.0.7 stable release?
Why i cannot obtain results with the hot keys listed in featuresmap?.
[featuremap]
blindxfer = #1; Blind transfer
disconnect = *0 ; Disconnect
automon = *1 ; One Touch Record
atxfer = *2 ; Attended transfer
i dont obtain results with this hotkeys, but pickup key *8 is ok.
dtmf is inband
Thanks to all in advance and for this great work¡¡¡
this is my sip.conf and extensions.conf
sip.conf
[general]
port=5060
bindaddr=0.0.0.0
context=default
srvlookup=yes
dtmfmode=inband
disallow=all
allow=all
language=es
[u0001]
type=friend
username=u0001
secret=xx
auth=md5
callerid=Cesar Garcia 0001
host=dynamic
callgroup=1
pickupgroup=1
nat=yes
canreinvite=no
--
extensions.conf
[default]
exten = ,1,Dial(SIP/u0001SIP/u0004,20)
exten = _0XXX,1, Dial(SIP/u${EXTEN},20)
exten = 828112070,1,Dial(SIP/u0001,20)
exten = 828112071,1,Dial(SIP/u0004,20)
--
César García.
Director de Sistemas, IdecNet S.A.
Centro de Gestión de Red.
Edificio IdecNet. C/Juan XXIII 44.
E-35004, Las Palmas de Gran Canaria,
Islas Canarias - España.
Tfn: +34 828 111 000 Ext: 340
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RE: [Asterisk-Users] Cisco 7.4 SIP firmware
Hi, The legal way is to buy a smartnet (support contract) for the soft. That way you can download it from Cisco's web site. Try to contact your reseller. Regards, Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] -Original Message- From: Betül Gözlükoglu [mailto:[EMAIL PROTECTED] Sent: mercredi 27 avril 2005 11:31 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7.4 SIP firmware Hello; Does anybody know how can I get the Cisco 7.4 SIP firmware? Many Thanks Betul image001.jpg___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Static and echo on PRI
Mark Johnson wrote: If you don't have any facts to share, please don't bother. I am desperate and don't have alot of time left and am begging for the list's advice. I left probably the largest post this month with EXACTLY what I have tried, the results, debug information, etc... I have removed drivers, swapped cards, changed IRQ's... I am open to any suggestions. If you tell me to go buy a different card, I will do that. You guys know more about than I do. What do you suggest, exactly? What *I* suggest is trying a different motherboard. This has solved my problems with Digium cards and Asterisk on at least three occasions. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Variable names in dial plans
/path/src/asterisk/doc/README.variables isn't what you are looking for? Jason Walker wrote: I second this. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Tuesday, April 26, 2005 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Variable names in dial plans Is there any documentation for all the variables available in extensions.conf? Every day I read this list, I read of at least a new variable name that I wasn't aware of so I go out and read bits and pieces about it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk doesn't disconnect when I hang up SIP (SIP - PSTN call)
I'm trying to learn Asterisk.
So far I'm using kphone and an ISDN line (with Eicon DIVA 2.01 PCI card).
I have created that extension following The Asterisk Handbook (page 36):
[ext-local-custom]
exten = _0.,1,Dial(Modem/ttyI0:${EXTEN:1})
exten = _0.,2,Dial(Modem/ttyI1:${EXTEN:1})
exten = _0.,3,Congestion
So whenever I call 055 from kphone, Asterisk connects me to an internal
55 number, and I can talk to myself (wohoo!) when I pick up the phone.
However, when I call 055 from kphone, and *don't* pick up the phone on
the other side, and then disconnect kphone (or even quit it), asterisk
keeps ringing 55.
I'd like to add, that Asterisk detects kphone disconnecting when the
phone is already established.
Any clue?
Tomek
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Re: [Asterisk-Users] Alternatives to SpanDSP??
I can't vouch for the image quality personally, but I have yet to hear of any complaints regarding quality from the end users. Craig - Original Message - From: Matthew Boehm [EMAIL PROTECTED] To: Asterisk Users asterisk-users@lists.digium.com Sent: Wednesday, April 27, 2005 11:16 AM Subject: Re: [Asterisk-Users] Alternatives to SpanDSP?? Damn. I'm using spandsp0.0.2pre15 and asterisk 1.0.7 with a single span card (US PRI) and I can get it to work about 85% of the time on a single 1 paged fax. I count a failed fax if any of the tiff images don't look like the original. I tried sending thru a 15 page fax. All 15 pages were received in the tiff image, but every 2 or 3 pages, it would seem as if the image skipped an inch. So instead of being 8.5 x 11, it was 8.5 x 10 (or 9). -Matthew From: Craig Guy [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 27 Apr 2005 07:51:22 +0800 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Alternatives to SpanDSP?? I agree with Steve on this, am piloting Spandsp 0.0.2pre15 on asterisk 1.0.7 with a TE405p, euroisdn. Fedora Core 2, kernel 2.6.9. Running on an old Dell Optiplex desktop PIII 450mhz with 256mb ram. Takes on average 350 faxes / day with just under 1% failed faxes. I define a failed fax as one with a filesize of 8bytes or won't render to pdf. On the strength of the pilot I am planning to install it to production at another site that takes approx 800 faxes per day. Craig - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 26, 2005 8:12 PM Subject: Re: [Asterisk-Users] Alternatives to SpanDSP?? Why would you expect a bunch of fax modems to work any better than spandsp? If spandsp doesn't work reliably your system is very likely broken. I have had hundreds of complaints about spandsp reliability. I have analysed at least 50 or 60 audio logs. I have found maybe 5 or 6 which has real spandsp problems. The rest had frame slips. Of the 5 or 6 with real problems, most have been fixed in the latest version. I have one weird audio log from a new HP combination printer and fax machine that i haven't sorted out yet. These HP machines really are total crap. I have workarounds in spandsp for several blatently wrong things they do. I don't yet know who is at fault with this latest problem. Regards, Steve Jeremy Melanson wrote: More like, I already have enough Digium cards, and I don't want purchase a bunch of fax/modems and more Digium cards than I alrady have. I have a PRI line that I'd like to support high-volume faxing on. I've gotten SpanDSP to work with * over the PRI, but I need a more reliability. That, and I guess I'm probably just being cheap too :-) - Jeremy On Mon, 2005-04-25 at 13:15 -0500, Anton Krall wrote: Maybe I started the day slow :) but let me see if I undertood correctly. You say that you don't want to rely on having to buy Digums or any other type of cards in oder to tie everything into spandsp and * but you would rather have dedicated PSTN lines with faxes on them? |-Original Message- |From: [EMAIL PROTECTED] | |I guess I didn't word this right. |It's not that SpanDSP ties up extensions, as it definitely |doesn't. I was more referring to the standard hardware-based |solutions out there that need to have a dedicated line for an |incoming fax. I need the ability to send and receive faxes |with a good amount of reliability, and would love to integrate |it with Asterisk. I'm just not keen on needing to buy a bunch |of Digium TDM cards just to support such a solution. | |Don't get me wrong, SpanDSP is great! I'm just looking for |something a little more enterprise-ready. | |On Mon, 2005-04-25 at 12:07 -0500, Eric Wieling aka ManxPower wrote: | I wasn't aware that SpanDSP tied up a bunch of extensions. | | Jeremy Melanson wrote: |I'm trying to see if anyone knows of an alternative solution, | commercial or non-commercial, to SpanDSP. I'm specifically looking |for another software-based, DSP fax that doesn't require me to add a tie up a | bunch of extensions on my PBX. | | Has anyone ever seen such an animal, or gotten such it to play nice | with Asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
RE: [Asterisk-Users] Good FXO for UK use.
Title: Message I had a look at it and...yes it seems to be the same card and it costs much less then I payed for it :( - j - From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RazzaSent: 26 April 2005 17:16To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Good FXO for UK use. Got a feeling that the same card I bought from goods2world.co.uk, which gave me terrible echo problems, due to the impedance mismatch of the US telco network (600ohm) versus the BT network. When using that card all seemed to disconnect fine, I assume the issue lies with the TDM400/FXO daughter board. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Johan AkerstromSent: 26 April 2005 15:39To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Good FXO for UK use. I just bought a DigitNetworks card called "DigitNetworks X100P - FXO PCI card" which supposedly is compatible with the discontinued Digium X100P card. This is a single port FXO card. Tell me how to test forthe TDM400 problem and I'll perform a test and post my results back to the list. The card is dead cheap $25 (but $36 :-( for shipping ). Regards Johan. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RazzaSent: 26 April 2005 14:25To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Good FXO for UK use. This maybe an out of place comment but it would appear Digium show little to no interest in non-North Americanimplementations, do we know if they are ever going to resolve this issue? or indeed how much it would cost? Based on my experience I'm sure there are a number of UK based people who could jointly fund such a development for a reasonable FXO product? Patrick Lidstone wrote:No the TDM400 does not work, it does not detect calling party termination correctly, so IVR and voicemail do not see the caller hang up on BT lines. Digium are aware of the problem, but fixing it doesn't seem to be a high priority, despite the fact that they have been supplied with detailed technical information regarding BT line behaviour :-(. Ian D. Willoughbywrote :Patrick is right about this , I get 20 or so seconds of solid tone at the end of all my voicemails,but I can live with this for the sake of no echo. ** Please note: The e-mail accompanying this disclaimer is confidential and may also be privileged. Please notify us immediately if you are not the intended recipient. You should not copy it, forward it, or use it for any purpose or disclose the contents to any person. This email has been swept for viruses using tools from our preferred suppliers. Telamon Systems actively supply both mail-scanning and anti-virus products in addition to supplying a range of security, infrastructure and business solutions to our customers. For further details please see our web site at www.telamon.co.uk, email [EMAIL PROTECTED] or call our sales team on +44 (0)870 607 4747 ** ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk doesn't disconnect when I hang up SIP (SIP - PSTN call)
try putting
exten = _0.,4,Hangup
like
[ext-local-custom]
exten = _0.,1,Dial(Modem/ttyI0:${EXTEN:1})
exten = _0.,2,Dial(Modem/ttyI1:${EXTEN:1})
exten = _0.,3,Congestion
exten = _0.,4,Hangup
regards,
Umair bari
Tomasz Chmielewski wrote:
I'm trying to learn Asterisk.
So far I'm using kphone and an ISDN line (with Eicon DIVA 2.01 PCI card).
I have created that extension following The Asterisk Handbook (page 36):
[ext-local-custom]
exten = _0.,1,Dial(Modem/ttyI0:${EXTEN:1})
exten = _0.,2,Dial(Modem/ttyI1:${EXTEN:1})
exten = _0.,3,Congestion
So whenever I call 055 from kphone, Asterisk connects me to an
internal 55 number, and I can talk to myself (wohoo!) when I pick up
the phone.
However, when I call 055 from kphone, and *don't* pick up the phone on
the other side, and then disconnect kphone (or even quit it), asterisk
keeps ringing 55.
I'd like to add, that Asterisk detects kphone disconnecting when the
phone is already established.
Any clue?
Tomek
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Re: [Asterisk-Users] Cisco 7960 earpiece speaker echo question
NBX does it due to the proprietary protocol. - Original Message - From: Jeremy Koski [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 26, 2005 11:56 PM Subject: Re: [Asterisk-Users] Cisco 7960 earpiece speaker echo question Hmm..We currently have the 3com NBX system with VoIP which do echo back in the earpiece. I thought it might just be the Cisco phone itself. Hopefully soon I can test with some more Cisco phones to see if it is the phone itself, or something else. On Tue, 26 Apr 2005, Henry Devito wrote: This echo is known as side tone it happens naturally on analog lines, IP phones usually do not provide this, Henry - Original Message - From: Jeremy Koski [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 26, 2005 1:53 PM Subject: [Asterisk-Users] Cisco 7960 earpiece speaker echo question Normally, when you speak into the receiver of a phone, you can hear yourself in the earpiece at a very low volume. I have a Cisco 7960 phone that I'm using with asterisk and I don't get that echo back on the earpiece speaker. I only have one Cisco 7960 phone, so I can't test it on others right now. My question is...Is this normal, do I have a bad handset? Is a way I can fix it? Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial Tone
Hi, I try to generate a dial tone (tone you hear when you pick up the hook). The tone should be stopped as soon the user dials a single digit. Unfortunately Playtones(dial) don't stop until another extension is completely dialed. DISA doesn't work either with our Siemens Phones. The scenario looks like this: User wants to call the number 12345 1. User picks up the hook 2. User dials 0 - hears dial tone 3. User dials 1 - dial tone stops 4. User dials 2345 - phone 12345 is ringing Is there any solution for this? Regards, Henry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] agent monitor filename
is there anyway of changing the default filename of the monitor file if using the record option in agents.conf. The ChangeMonitor command seems to work only for a channel if it's using the Monitor command. Julian. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk doesn't disconnect when I hang up SIP (SIP - PSTN call)
Umair Bari wrote:
try putting
exten = _0.,4,Hangup
like
[ext-local-custom]
exten = _0.,1,Dial(Modem/ttyI0:${EXTEN:1})
exten = _0.,2,Dial(Modem/ttyI1:${EXTEN:1})
exten = _0.,3,Congestion
exten = _0.,4,Hangup
no, still does not hang up :(
I have to pick up the phone and hang up manually (or kill asterisk).
Tomek
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[Asterisk-Users] noload res_musiconhold.so breaksa IAX
In response to a previous question about disabling music on hold, I was advised to do: noload = res_musiconhold.so Unfortunately, this keeps Asterisk (1.0.5) from running: [chan_iax2.so]Apr 27 04:11:34 WARNING[19654]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_moh_stop Apr 27 04:11:34 WARNING[19654]: loader.c:440 load_modules: Loading module chan_iax2.so failed! Ideas please? Is this a bug or a configuration problem. Thanks, /edg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Tone
On Wednesday 27 April 2005 12:12, Henry Jensen wrote:
Hi,
User wants to call the number 12345
1. User picks up the hook
2. User dials 0 - hears dial tone
3. User dials 1 - dial tone stops
4. User dials 2345 - phone 12345 is ringing
We're using chan_capi and had this same problem... The following really hacky
solution works OK with Asterisk 1.0.7, but not with CVS - I don't know why :)
[default]
exten = _120.,1,Goto(s,1) ; fax extensions are 1201 - 1208
exten = s,1,NoOp( incoming call from ISDN )
exten = s,2,Answer
exten = s,3,PlayTones(dial); Give the caller a familiar noise.
exten = s,4,DigitTimeout(0.1)
exten = s,5,WaitExten(0.1)
; next section captures the next digit and stops the dialtone
exten = _X,1,NoOp( Got a digit! It was ${EXTEN})
exten = _X,2,StopPlaytones()
exten = _X,3,SetVar(Predigits=${EXTEN}) ; Put that digit aside for
use later...
exten = _X,4,Goto(s-gathermoredigits,1)
exten = s-gathermoredigits,1,NoOp( Now looking for the rest of the number)
exten = s-gathermoredigits,2,DigitTimeout,3
exten = s-gathermoredigits,3,WaitExten(8) ; and give the caller 8
seconds overall to do their thing
; log + dial the composite number of Predigits + the remainder
exten = _X.,1,NoOp(${TIMESTAMP} ok, now we're going to dial
${Predigits}${EXTEN})
exten = _X.,2,Goto(outbound,${Predigits}${EXTEN},1)
exten = t,1,Goto(#,1) ; If they take too long, give up
exten = i,1,Playback(invalid) ; That's not valid, try again
The [outbound] context is jsut full of the normal exten = _01.,1,Dial(blaaah)
call routing
If someone has a better way of doing this, I'd be interested to hear it!
Cheers,
Gavin.
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Re: [Asterisk-Users] Queue Management and Command Execution
Yes, I read them. But, then my question is: how can I make the file name include the agent that will get the call once it's distributed? Thanks, Daniel On Apr 27, 2005, at 3:40 AM, Anton Krall wrote: You can record queue conversations, check out the configs in queue.conf |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Daniel Salama |Sent: Martes, 26 de Abril de 2005 05:53 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: [Asterisk-Users] Queue Management and Command Execution | |Is there a way to execute a command prior to sending a queued |call to an agent? | |What I'm trying to do is record agent's conversation at the |server. I could put the Monitor command when the call is |answered by *, but if the caller has to wait on hold for some |time, I wouldn't want to record that. | |What I would prefer is to be able to put the caller in a queue |and once an agent is ready, for the Monitor command to kick |in. Even better would be to know the agent id where the call |is going to be sent to, so I can use it as part of the file |name of the Monitor command. | |Any clues on how to do this, if at all possible? | |Thanks, |Daniel | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] noload res_musiconhold.so breaksa IAX
Ed Greenberg wrote: In response to a previous question about disabling music on hold, I was advised to do: noload = res_musiconhold.so Unfortunately, this keeps Asterisk (1.0.5) from running: [chan_iax2.so]Apr 27 04:11:34 WARNING[19654]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_moh_stop Apr 27 04:11:34 WARNING[19654]: loader.c:440 load_modules: Loading module chan_iax2.so failed! Ideas please? Is this a bug or a configuration problem. The advice was wrong. res_musiconhold.so is required by many modules. Remove /etc/asterisk/musiconhold.conf instead. --Eric ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Static and echo on PRI
On April 27, 2005 12:42 am, Matt Klein wrote: Most likely, they can give you Echo Can for free. Bell Canada will not put echo cans on their PRIs unless you specifically ask (and pay) for the service. Indeed, the line techs were surprised to know that echo could even exist on PRI; thse are very smart people, just not well schooled in T1/PRI. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Static and echo on PRI
On April 27, 2005 12:03 am, Mark Johnson wrote: day now. What I find strange is this... If I speak at a normal tone, it sounds OK. I still get static noise when the other person speaks. If I talk louder, I start to get what sounds like a partial echo. If I yell, I get a definite echo. It almost sounds like the card has an odd gain problem on receive. The echo you hear of yourself at high volumes I'd call normal with the software echo cans. Oh yes -- Digium also has TE405Ps and TE410Ps *with* onboard hardware echo cans now. I have one coming in. I forgot about the loopback test; use zttool and throw the span into loopback, do you see errors? When monkeying with the echo cancel, I never really noticed a difference. I would even reboot the machine between changes to see if it made a difference. echocan's a finicky little thing. echotraining does work but I never use it because the delay at the start of the call is unacceptable to me (most of us have headsets so the delay between picking up the phone and getting it to your ear is almost zero). The agressive echo canceller basically turns the phone into a half-duplex system. It's a brute force way of eliminating echo. :-) I am running this on Fedora Core 1. I will try any OS you recommend, but I have always had great luck with RH type distro's. I keep 400 and 500 day uptimes on those machines and they run many, many services. Uptimes would be higher but it seems whenever I find a good place to work, they close up or I move. Admittedly, I don't use RPM's for the core services, I typically compile those myself. I also shut down every module and service I don't need. I did alot of reading and it seemed like Digium cards were the real deal and I also found many users that had luck with the same setup. Should I try a different approach/OS/system? Stability isn't the issue here; it's interrupt latency and kernel delays. Personally I run Slackware for everything but I am certain there are many people here running FC1 for their systems without any issue, so at this point I am not suggesting dropping it. I'm not sure whether those who are running FC1 are running it with Fedora's kernel or with a stock kernel, but I *have* seen people with distro-specific kernels have problems that disappear when they use a stock kernel. Which motherboard are you currently using? Which have you switched to for the test? The great thing about Linux is that in most cases you can (and I have on *many* occasions) pull the hard drive out of one system and install it on a totally different one and it just works. Perhaps some minor tweaking of ethernet drivers but for the most part there are no hassles. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] All lines are busy
Hi all. I ma having a problem with pstn/tdm lines. After the system has been in use for a while, it seems that I can only use one line in the system. I have three PSTN attached to Digium TDM04B/400P and they are grouped into g0. I tried using callprogress and busydetect in the zapata.conf but then my SIP phones can't place calls on hold anymore. (Thought that was pretty strange). Once I restart the asterisk server and the zaptel service it works normally for a while and then goes back. No matter what happens, one line is always available to dial in or out on. Any suggestions would be most appreciated. Thanks CM ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 earpiece speaker echo question
On 4/26/05, Jeremy Koski [EMAIL PROTECTED] wrote: Normally, when you speak into the receiver of a phone, you can hear yourself in the earpiece at a very low volume. I have a Cisco 7960 phone that I'm using with asterisk and I don't get that echo back on the earpiece speaker. I only have one Cisco 7960 phone, so I can't test it on others right now. My question is...Is this normal, do I have a bad handset? Is a way I can fix it? On my 7960 if I blow accross the mouthpiece I can hear it quietly in the earpiece (at least when dialtone is heard) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Does anyone know what the [WARNING: . Changethread: Can't change device '**Unknown**'] line means below.. I just set verbosity to level 5, and noticed that error everytime a voicemail is left.. Everything seems to work ok, and I have no idea how long that error has been there, but I'm just curious if it is something important :-) Thanks, - Andre --- -- Executing Dial(SIP/PhonePort1-4b1b, SIP/PhonePort3|30|tr) in new stack -- Called PhonePort3 -- SIP/PhonePort3-cc9f is ringing -- Nobody picked up in 3 ms -- Executing VoiceMail(SIP/PhonePort1-4b1b, u7600) in new stack -- Playing '/var/spool/asterisk/voicemail/default/7600/unavail' (language 'en') -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/7600/INBOX/msg0012 format: wav49, 0x8485088 -- x=1, open writing: /var/spool/asterisk/voicemail/default/7600/INBOX/msg0012 format: gsm, 0x8407fa8 -- x=2, open writing: /var/spool/asterisk/voicemail/default/7600/INBOX/msg0012 format: wav, 0x846ce30 -- User hung up Apr 27 08:01:57 WARNING[21497]: app_queue.c:375 changethread: Can't change device '**Unknown**' with no technology! == Spawn extension (LocalSIP, 7600, 2) exited non-zero on 'SIP/PhonePort1-4b1b' == Spawn extension (Analog_In2, s, 8) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Busy Tone
Title: Busy Tone Hello, I've connected an Panasonic KX-TD 1232 PBX to an Asterisk PBX through an ISDN-line. I use an AVM Fritz! ISDN PCI card on the Asterisk PBX and connect it to the S0 bus of the Panasonic. When I make a call from a softphone to a phone that is connected to the Panasonic, there isn't a problem. But when I try to make a call from a phone on the Panasonic to a softphone, I get a busy tone. If I keep trying, then after a few times, it works. Does anyone have any ideas to solve this problem? Thanks in advance Grtz, Dennie __ This mail has been scanned for viruses by an AXS Web Firewall, powered by SecuTeam NV. _ This mail has been scanned for viruses by AXS Mail, powered by SecuTeam NV. Register for AXS Mail at http://www.secuteam.com! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] noload res_musiconhold.so breaksa IAX
Works, thanks. --On Wednesday, April 27, 2005 6:36 AM -0500 Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Ed Greenberg wrote: In response to a previous question about disabling music on hold, I was advised to do: noload = res_musiconhold.so Unfortunately, this keeps Asterisk (1.0.5) from running: [chan_iax2.so]Apr 27 04:11:34 WARNING[19654]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_moh_stop Apr 27 04:11:34 WARNING[19654]: loader.c:440 load_modules: Loading module chan_iax2.so failed! Ideas please? Is this a bug or a configuration problem. The advice was wrong. res_musiconhold.so is required by many modules. Remove /etc/asterisk/musiconhold.conf instead. --Eric ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
On 4/27/05, Andre Normandin [EMAIL PROTECTED] wrote: Does anyone know what the [WARNING: . Changethread: Can't change device '**Unknown**'] line means below.. I just set verbosity to level 5, and noticed that error everytime a voicemail is left.. Everything seems to work ok, and I have no idea how long that error has been there, but I'm just curious if it is something important :-) Looks like the call is coming out of voicemail and then going somewhere else or you have an exten _. defined that is catching a hangup, post your extensions.conf for further analysis. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialing out from remote.
I'm attempting the following set up. During Hours - Receptionist Takes the call (no problem works great) After hours I would like to add a item to the receptionist to transfer the call to my cell, any direction would be a great help. I have 4 PSTN incoming lines as backup and Voicepulse. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alternatives to SpanDSP??
On April 26, 2005 10:57 am, Eric Wieling aka ManxPower wrote: We have terrible problems sending faxes via the TDM cards. Not even using SpanDSP. Just TE110P for the telco side and TDM400P for the fax machine. Yes there is a timing issue that crept in somewhere in the last 12-15 months; I believe it's related to the CPU use spiking every few seconds. I would sort of disagree with the spiking thingie (now). If you modify the zttest app to provide timing output in terms of seconds and microseconds, you don't see the spiking impacting those measurements. Rather, you see 8,192 bytes arriving in something greater then 1.000 seconds on a very consistent basis. In my case, that timing is right at 1.02 seconds (about 20,000 microseconds late), which translates into a missed/slipped frame for about one of fifty frames. Not cool with spandsp at all, but not noticed for pure voice use. The design of the card (and asterisk) is 100% oriented around receiving 8,192 bytes from the card every 1. seconds exactly. Any significant variation from 1.000 seconds will result in a missed frame (1024 bytes) sooner or later. What I've not been able to figure out is why the delay. I'm 95% sure it has more to do with asterisk code (including drivers) then it does with other system interrupt handlers, interrupt latency, etc. Those _other_ things certainly can impact it, but there is definitely something within asterisk that is directly related to the TDM card and its drivers. (Its almost consistent enough to look closer at the clocking on the TDM itself. That assumes a clock on the TDM card is responsible for raising the interrupt to the O/S via the pci bus.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Re] Re: [Asterisk-Users] Alternatives to SpanDSP??
Hi Cyril,
Good work. process_baud is a fairly big routine, and your backtrace
doesn't give the actual line number at which things fall over. However,
studying the code I see that I do not protect against the possibility of
a divide by zero during the initial coarse carrier estimation of any of
the fast modems.
I just created 0.0.2pre16, which should eliminate this. Can you try it,
and tell me what happens?
Regards,
Steve
Cyril VELTER wrote:
If you can catch one of these events, and get a traceback of the stack,
I will take a look. This is not happening to most users, so it must be
some specific combination of things on your machine. I have reports of
high volume faxing running for extended periods from some users.
Hi steve,
I use spandsp on one production machine (for receiving fax only) and have
experienced some crash. It's pretty rare, and seem to be related to a
particular fax machine trying to send a fax. When I get a crash, I ususally get
three or for at several minutes interval). I've not been able to identify the
sender fax.
I've some asterisk core dump files. All the crashes occur in libspandsp.so in
the process_baud function.
You'll find bt and bt full output at the end of this email. If you need more
informations, please contact me.
I'm running spandsp 0.0.2pre11 and asterisk CVS as of march 28.
Cyril
Here's the bt result :
#0 0x010025ab in process_baud () from /usr/local/lib/libspandsp.so.0
#1 0x01001bd6 in v27ter_rx () from /usr/local/lib/libspandsp.so.0
#2 0x00ff6334 in fax_rx_process () from /usr/local/lib/libspandsp.so.0
#3 0x006a9aa1 in rxfax_exec (chan=0x9af63b8, data=0xac9f5410) at
app_rxfax.c:274
#4 0x0808407d in pbx_extension_helper (c=0x9af63b8, con=0x0, context=0x9af6500
fax, exten=0x9af65f4 s, priority=2, label=0x0,
callerid=0xac9fb700
/var/spool/asterisk/faxin/467738570-20050414-121811.tif, action=0) at
pbx.c:482
#5 0x0807c19a in ast_pbx_run (c=0x9af63b8) at pbx.c:1875
#6 0x08084891 in pbx_thread (data=0x0) at pbx.c:2120
#7 0x00660dec in start_thread () from /lib/tls/libpthread.so.0
#8 0x003b3a2a in clone () from /lib/tls/libc.so.6
Here's the bt full :
(gdb) bt full
#0 0x010025ab in process_baud () from /usr/local/lib/libspandsp.so.0
No symbol table info available.
#1 0x01001bd6 in v27ter_rx () from /usr/local/lib/libspandsp.so.0
No symbol table info available.
#2 0x00ff6334 in fax_rx_process () from /usr/local/lib/libspandsp.so.0
No symbol table info available.
#3 0x006a9aa1 in rxfax_exec (chan=0x9af63b8, data=0xac9f5410) at
app_rxfax.c:274
res = 0
count = 0
percentflag = 0
fil =
/var/spool/asterisk/faxin/467738570-20050414-121811.tif\000 [EMAIL PROTECTED]@\000
[EMAIL PROTECTED] [EMAIL PROTECTED]@\000xt\237¬\033k3\000 [EMAIL PROTECTED]
\000\000 [EMAIL PROTECTED]
;30;40m--
\033[0;37;[EMAIL PROTECTED]@[EMAIL PROTECTED]
[EMAIL PROTECTED]@\000§\000\000\000àD\017\b...
tmp = /var/spool/asterisk/faxin/467738570-20050414-121811.tif, '\0'
repeats 200 times, ·
x = 0x0
i = 0
fax = {local_ident = LODGIS, '\0' repeats 14 times, far_ident =
0467738570\000\000\000\000\000\000\000\000\000\000,
sub_address = '\0' repeats 20 times, password = '\0' repeats 20 times,
vendor = 0x0, model = 0x0, verbose = 0, phase_b_handler = 0,
phase_b_user_data = 0x0, phase_d_handler = 0x6a9648 phase_d_handler,
phase_d_user_data = 0x9af63b8, phase_e_handler = 0x6a93f8 phase_e_handler,
phase_e_user_data = 0x9af63b8, t30_flush_handler = 0, t30_flush_user_data =
0x0, options = 0, phase = 5, next_phase = 0, state = 6, mode = 0,
msgendtime = 32000, samplecount = 0, dtc_frame = '\0' repeats 14 times,
dtc_len = 0, dcs_frame = '\0' repeats 14 times, dcs_len = 0,
dis_frame = \200\000Îô\200\200\201\200\200\200\030\000\000\000, dis_len =
11, in_message = 0, tone_gen = {v2_1 = 1005.99878, v3_1 = -6413.77002,
fac_1 = -0.156918198, v2_2 = 0, v3_2 = 0, fac_2 = 0, duration = {20800,
600, 0, 0}, repeat = 0, current_section = -1, current_position = 0}, hdlcrx = {
crc_bytes = 2, frame_handler = 0xff389c process_rx_crp+28, user_data =
0xac9f5620, report_bad_frames = 0, rx_state = 1, bitbuf = 2332973030,
byteinprogress = 223, numbits = 3, buffer =
ÿ\023\203\000\212 \200\200\200\200\200\200\020\r§¸\003, '\0' repeats
376 times, len = 0,
rx_bytes = 36, rx_frames = 2, rx_crc_errors = 0, rx_length_errors = 1,
rx_aborts = 1}, hdlctx = {crc_bytes = 2,
underflow_handler = 0xff2338 fast_getbit+284, user_data = 0xac9f5620,
numbits = 4, idle_byte = 231, len = 0,
buffer = '~' repeats 44 times, ûà\000²¤¸\210¼\214\201, '\001' repeats
13 times,
)\207\237\237\237\237¾ø\200\020\a2ð\020\030\020\020\020\021\214oGç, '\0'
repeats 311 times, pos = 0, byte = 7392, bits = 3, underflow_reported = 1},
v21tx = {baud_rate = 300, get_bit = 0xfead04 hdlc_tx_getbyte+88,
user_data = 0xac9f58fc, phase_rates = {993211187, 885837004}, scaling =
7218, current_phase_rate =
Re: [Asterisk-Users] TE405P w/ Intel SE7210TP1_E Motherboard
We've just had problems with a range of Intel boards (Bukner and Avalon) that have the PCI-Express technology - http://www.digium.com/index.php?menu=compatibility lists the Intel SE7525GP2 but we've had problems with the Intel SE7221BK1-E. Digium say Firmware release 10 fixes this issue. All new cards from May 2005 ship with version 10. The link you give below http://www.gtweb.net/support/pdf/SE7210TP1-E_Product_Brief.pdf discusses the PCI-X adapter slot which Intel are pushing the boundaries of the PCI specifications. Rob On 4/26/05, Greg Boehnlein [EMAIL PROTECTED] wrote: On Sat, 29 Jan 2005, Greg Boehnlein wrote: Hello, I'm looking at building a couple new PRI Gateway boxes using TE405P cards, and was wondering if anyone has had any experiences (good or bad) with the Intel SE7210TP1_E motherboards from Intel. General Technics builds some really nice (and cost effective) 1U servers based on the board: Server: http://www.gtweb.net/gt637.html Specs: http://www.gtweb.net/support/pdf/SE7210TP1-E_Product_Brief.pdf Comments? Per my other costs, I've settled on the following box from General Technics for my PRI gateway boxes. http://www.gtweb.net/gt637.html I had a pretty detailed conversation with Chris from GT about the board and how it is laid out, and it appears that the unit has several PCI busses in it and they are separated nicely; The only stuff on the 5V PCI slot is a 10/100 NIC + the Video. Since there is a Gig-E that shares the same bus w/ the SATA drive I can just use that and keep the bus free for the TE405. On most of my boxes, I disable the Video and use serial console. I'll keep people up to speed on how this works out. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rob Lith Connection Telecom cc Mobile: +27 (82) 3893332 Tel:+27 (21) 6572770 DDI:+27 (21) 6572774 Fax:+27 (21) 6572775 Email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Good FXO for UK use.
Message: 5 Date: Wed, 27 Apr 2005 12:04:30 +0100 From: Johan Akerstrom [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Good FXO for UK use. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii I had a look at it and...yes it seems to be the same card and it costs much less then I payed for it :( - j - From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Razza Sent: 26 April 2005 17:16 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Good FXO for UK use. Got a feeling that the same card I bought from goods2world.co.uk, which gave me terrible echo problems, due to the impedance mismatch of the US telco network (600ohm) versus the BT network. When using that card all seemed to disconnect fine, I assume the issue lies with the TDM400/FXO daughter board. The problem is indeed unique to the TDM400 FXO daughter board. I can confirm that the X100P and clones do correctly detect hangup on the BT network, but are plagued by echo problems due the impedance mismatch with the UK phone network. Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to use dialparties.agi
Hi! I looking for an example how to use the dialparties.agi from Asterisk Management Portal 1.10.007a. I tried to understand it by reading the extensions.conf of AMP, but without success. Is anybody out there, who can give me a more easy example or an explanation. Thanks, Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Checking for a sound file
Just as a reminder for those using Outlook, a large percentage of us that receive html postings to the list simply delete them. If you want to see responses from a larger group, stop the html stuff. From: Wiley Siler [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Checking for a sound file Date: Tue, 26 Apr 2005 12:43:22 -0700 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Look at this code ; ; IVR RECORDER ; ; Record voice file to /tmp directory exten = 205,1,Wait(2) ; Call 205 to Record new Sound Files exten = 205,2,Record(/tmp/asterisk-recording:gsm) exten = 205,3,Wait(2) exten = 205,4,Playback(/tmp/asterisk-recording) exten = 205,5,wait(2) exten = 205,6,Hangup Now if I call in on my * and dial 205 I can record a message to the path described above As long as your IVR settings are playing that same file to them, you should be fine I would of course customize the file names and possible the locations... Cheers, Wiley - From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marc Sent: Tuesday, April 26, 2005 11:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Checking for a sound file Hi, At this moment I'm configuring my first asterisk pbx, and am running into the following problem: I would like to create a phonenumber for my customers, which they can call to hear if there are any problems with the servers. In case of a problem, I would like to be able to call that number, authenticate myself and record a new message. From that moment that message must be played when customers call. When the problem is solved, I would like to call the same number again, authenticate, and remove that message, so the original message is again played to customers that call. I've read the wiki pages, but I'm not able to create this configuration. Can somebody please give me some tips how to do this? Regards, Marc ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] b0rked hfc config
Shouldn't it be: ? bchannel = 9,10 dchannel = 11 bchannel = 12-13 dchannel = 14 Julian J. M. On 4/27/05, Thomas Andrews [EMAIL PROTECTED] wrote: On Wed, Apr 27, 2005 at 11:08:46AM +0200, Thomas Andrews wrote: I have 2 Billion cards and I can't get the hfc driver to work. I get this error: ZT_CHANCONFIG failed on channel 9: No such device or address (6) What am I doing wrong ? This is my /etc/asterisk/zaptel.conf: [channels] switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=yes group = 1 context = incoming channel = 9 channel = 10 channel = 12 channel = 13 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alternatives to SpanDSP??
On April 27, 2005 09:04 am, Rich Adamson wrote: I would sort of disagree with the spiking thingie (now). If you modify the zttest app to provide timing output in terms of seconds and microseconds, you don't see the spiking impacting those measurements. Rather, you see 8,192 bytes arriving in something greater then 1.000 seconds on a very consistent basis. Do you have a copy of this patch? I'd like to work on this problem with you (in my ample spare time, ha!). The design of the card (and asterisk) is 100% oriented around receiving 8,192 bytes from the card every 1. seconds exactly. Any significant variation from 1.000 seconds will result in a missed frame (1024 bytes) sooner or later. *nod* What I've not been able to figure out is why the delay. I'm 95% sure it has more to do with asterisk code (including drivers) then it does with other system interrupt handlers, interrupt latency, etc. Those _other_ things certainly can impact it, but there is definitely something within asterisk that is directly related to the TDM card and its drivers. (Its almost consistent enough to look closer at the clocking on the TDM itself. That assumes a clock on the TDM card is responsible for raising the interrupt to the O/S via the pci bus.) Well the clock on the TDM400P is the same as what is used in the T100P, X100P (or is it X101P?) and TE110P. It's just a cheap crystal oscillator within the TJ320 so at least in theory the same problem should exist with those cards if it were an oscillator issue. Even cheap oscillators are more accurate than this though. :-) I'm curious though if the CPU spiking in the wctdm driver has something to do with it (causing the time to stretch), especially since this isn't seen on the other cards, only within that driver, and it's only that card that seems to have it. (I'll reply to your original post about the zttest stuff in -dev and we can continue this there.) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] return a value from dial macro
I would really appreciate any insight here. I have seen a number of
posts in the past regarding implementation of a voicemail detection
scheme using silence detection as well as the machine detect, but
without MACRO_RESULT, there doesn't appear to be any way to actually
implement this.
Thanks
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Dolloff
Sent: Tuesday, April 26, 2005 8:43 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] return a value from dial macro
Does anyone know of a way to pass a value back to the dial plan after
calling a macro from the dial app in the 1.0 release? I think this
should be pretty simple, but I can't quite figure out how.
The example would work except that the modified value of found is not
usable when Dial ends. I think that the MACRO_RESULT would do this,
but
it doesn't appear to have made it into 1.0
I want to stop going through the priorities after completion of a
successful dial, but only if MachineDetect returns 0. If it returns 1
I
want to hangup on the called party and goto the next priority
exten = 223,3,SetVar(__found=0)
exten = 223,4,Dial(SIP/[EMAIL PROTECTED],48,rtgM(md))
exten = 223,5,GotoIf($[${found} = 1]?7)
exten = 223,6,Voicemail(u${EXTEN})
exten = 223,7,Hangup
[macro-md]
exten = s,1,MachineDetect(700,2,2200)
exten = s,2,GotoIf($[${MACHINE} = 1]?3:5)
exten = s,3,SoftHangup(${CHANNEL})
exten = s,4,Goto(6)
exten = s,5,SetVar(found=1)
exten = s,6,NoOp
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Re: [Asterisk-Users] Checking for a sound file
Rich Adamson wrote: Just as a reminder for those using Outlook, a large percentage of us that receive html postings to the list simply delete them. If you want to see responses from a larger group, stop the html stuff. Mozilla Mail, at least, lets you do View / Message Body As / Plain Text. It doesn't follow the torture stupid people way of thinking, but it does keep me from seeing all that HTML crap. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SetGroup on dialed calls?
hi folks, I looked through the list archives and the wiki, but couldn't find an answer to this. Apologies if I just missed something obvious. I want to only have call waiting for certain calls (i.e., those that are dialed directly to a user rather than going through a queue). It seems that the way to do this is to call SetGroup() on all incoming calls and CheckGroup() only on non-call-waiting calls, combined with Local/ channels when needed. However, I can't figure out how to do the SetGroup() properly on outgoing calls (i.e., those that the internal user dials). Is there some obvious way that I'm missing to call some commands before proceeding to the rest of the dialplan? Any thoughts -- including alternate ways to achieve the same goals -- would be much appreciated. (I tried using the incominglimit parameter in sip.conf, but it seems not to be very flexible.) Many thanks, mike -- mike castleman network / systems administrator democracy now! mailto:[EMAIL PROTECTED] tel:+1-(212)-431-9090 signature.asc Description: Digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco SIP Firmware Price Increase
I've heard a few times that the firmware for Cisco Phones to use them with SIP is going to increase $150. Is this true? - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extensions / Contexts
Good points. I stand corrected. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Wade Sent: Tuesday, April 26, 2005 2:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extensions / Contexts Wiley Siler wrote: The short answer is No. Wrong, the short answer is maybe. The method you describe is intrinsicly illogical. Assuming there is an IVR, how will I know which extension 2000 I am calling if that were possible? I might get company A instead of company B. If you get this behavior in your IVR's then you need to redesign them. If what you say were true, I couldn't have a multiple level IVR where 1 takes to you another menu which has another 1 in it that takes me to another menu with another 1 in it! Also, remember that a DEVICE is NOT an EXTENSION! You can create two * servers with identical dial plans, link them over IAX, and allow users to call each other if they use a rpefix lke 7. Example: Comp-A user 2000 calls comp-B user 2000 by dialing 72000. This would work, but why? Use contexts, that's what they are there for - dialplan 'partitioning'. Now if you want to use one server only, then just use 200x for one company and 300x for the other and segment the dial plans. W Again, this would work, but why? And now you've got a higher potential of CompanyA customers reaching CompanyB employees - not good. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Silva Sent: Tuesday, April 26, 2005 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Extensions / Contexts Hi everybody, snip [2000] username=companyA_2000 context=contextCompanyA [2000] username=companyB_2000 context=contextCompanyB Any help will be appreciated. Sebas Your example there should work with one minor change. Make the part in the brackets unique. Meaning [a2000] and [b2000] or similar. Then, from within your dialplan configure extension = 2000 in context [CompanyA] to dial SIP/a2000 and similar for CompanyB, making exten = 2000 in context [CompanyB] dial SIP/b2000. This entire topic has been discussed multiple times on the list. Please read the archives, use google and the 'site:' argument. -Chris PS: Remember that a DEVICE is NOT an EXTENSION! PPS: Remember that an EXTENSION need NOT be a DEVICE! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP4000 Conference Phone
Yep. I have this working now. Thank you! Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul HalesSent: Tuesday, April 26, 2005 4:31 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Polycom IP4000 Conference Phone The 1.41 on the website is fine. PaulH From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley SilerSent: Tuesday, 26 April 2005 11:40 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Polycom IP4000 Conference Phone I was afraid you would say that. Does anyone out there have the latest firmware for the Soundpoint IP 4000? Thanks, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul HalesSent: Monday, April 25, 2005 7:45 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Polycom IP4000 Conference Phone You need to have a very new firmware... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley SilerSent: Tuesday, 26 April 2005 6:33 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Polycom IP4000 Conference Phone Can someone verify that this phone uses the same configs and sip.ld and other files as the IP 500 ? I jus tgot one and I cannot get it provisioned yet. Thanks, Wiley CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you.CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you.CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Checking for a sound file
Causes me to wonder a couple of things. Why does ANYONE use Outhouse or Outhouse Express? There are many much more friendly Windows E-mail clients, from Mozilla on down Don't know nor care about Linux E-mail. For me Linux is a means to an end , not a religion. Even more of a question, why doesn't mailman convert everything to plain text? It seems to be an option that the list owner/operator could easily turn on and give list members ONE less thing to carp about. John Novack Rich Adamson wrote: Just as a reminder for those using Outlook, a large percentage of us that receive "html" postings to the list simply delete them. If you want to see responses from a larger group, stop the html stuff. From: Wiley Siler [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Checking for a sound file Date: Tue, 26 Apr 2005 12:43:22 -0700 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Look at this code ; ; IVR RECORDER ; ; Record voice file to /tmp directory exten = 205,1,Wait(2) ; Call 205 to Record new Sound Files exten = 205,2,Record(/tmp/asterisk-recording:gsm) exten = 205,3,Wait(2) exten = 205,4,Playback(/tmp/asterisk-recording) exten = 205,5,wait(2) exten = 205,6,Hangup Now if I call in on my * and dial 205 I can record a message to the path described above As long as your IVR settings are playing that same file to them, you should be fine I would of course customize the file names and possible the locations... Cheers, Wiley - From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Marc Sent: Tuesday, April 26, 2005 11:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Checking for a sound file Hi, At this moment I'm configuring my first asterisk pbx, and am running into the following "problem": I would like to create a phonenumber for my customers, which they can call to hear if there are any problems with the servers. In case of a problem, I would like to be able to call that number, authenticate myself and record a new message. From that moment that message must be played when customers call. When the problem is solved, I would like to call the same number again, authenticate, and remove that message, so the original message is again played to customers that call. I've read the wiki pages, but I'm not able to create this configuration. Can somebody please give me some tips how to do this? Regards, Marc ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco SIP Firmware Price Increase
Not that I know of I am a Cisco partner and the Category 1 contract is still at least half that or less. - Original Message - From: Dan Levine To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, April 27, 2005 8:49 AM Subject: [Asterisk-Users] Cisco SIP Firmware Price Increase I've heard a few times that the firmware for Cisco Phones to use them with SIP is going to increase $150. Is this true? - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extensions / Contexts
Thanks a lot, I thought this is possible because I don't need to link companies, also, I can solve the problem of the IVR depending on the channel of the PSTN that originates the call. thanks again Sebas Wiley Siler wrote: The short answer is No. The method you describe is intrinsicly illogical. Assuming there is an IVR, how will I know which extension 2000 I am calling if that were possible? I might get company A instead of company B. You can create two * servers with identical dial plans, link them over IAX, and allow users to call each other if they use a rpefix lke 7. Example: Comp-A user 2000 calls comp-B user 2000 by dialing 72000. -- Sebastian Silva G R U P O G A U S S Depto. Sistemas Av. Libertador 6250 4 piso Tl.: 4 706- (int. 121) [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No audio playback after upgrade from 1.0.1
Hello, I'm having some problems upgrading my system from 1.0.1 to 1.0.7. After the upgrade the Playback and Background dial commands don't produce any audio or extremely distorted. I've tried custom recordings and the prepackaged ones with the same result. Calls work fine using ulaw through sip and iax2 channels. Any ideas? Robert J Derr Weatherflow, Inc. begin:vcard fn:Robert Derr n:Derr;Robert org:WeatherFlow, Inc.;IT Florida office adr:;;120 Canal St;New Smyrna Beach;FL;32168;USA email;internet:[EMAIL PROTECTED] title:Software Developer tel;work:386-423-1516 tel;fax:386-409-5178 url:http://www.iwindsurf.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] All lines are busy
Sounds like a possible disconnect issue, When the lines are not available try doing a 'zap show channel X' with X being the channel number 1,2,or 3 and see if asterisk thinks the line is onhook or offhook. Just a thought. Henry - Original Message - From: Colin E. McDonald [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, April 27, 2005 7:02 AM Subject: [Asterisk-Users] All lines are busy Hi all. I ma having a problem with pstn/tdm lines. After the system has been in use for a while, it seems that I can only use one line in the system. I have three PSTN attached to Digium TDM04B/400P and they are grouped into g0. I tried using callprogress and busydetect in the zapata.conf but then my SIP phones can't place calls on hold anymore. (Thought that was pretty strange). Once I restart the asterisk server and the zaptel service it works normally for a while and then goes back. No matter what happens, one line is always available to dial in or out on. Any suggestions would be most appreciated. Thanks CM ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Checking for a sound file
Just as a reminder for those using Outlook, a large percentage of us that receive html postings to the list simply delete them. If you want to see responses from a larger group, stop the html stuff. Mozilla Mail, at least, lets you do View / Message Body As / Plain Text. It doesn't follow the torture stupid people way of thinking, but it does keep me from seeing all that HTML crap. This may sound really stupid, but I'm using an email viewer written in 1996 that doesn't understand html, doesn't open attachments, etc. Never have a virus issue, buffer overflow attempts, etc. :) But, there seems to be a fair number of newbies that wonder why their questions aren't answered, and html posting is at least partially to blame. Just a reminder to those folks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream BT101 Firmware
I've been trying to upgrade a grandstream BT-101 phone, whatever I do, it doesn't seem to want to upgrade. I've pointed it at the grandstream TFTP IP listed on the wiki, I've also tried pointing it at my own TFTP server (after putting all the firmware images downloaded from grandstream website into /tftpboot). Is there any specific trick I should know (eg, upgrade in some sequance, or hack my tftp server in some way, special files/etc..) Also, any comments on what the overall best firmware is for these handsets currently would be interesting... Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue Management and Command Execution
Look at the agents.conf. There is an option there to record calls. Maybe this will point you in the right direction. - Original Message - From: Daniel Salama [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 27, 2005 6:32 AM Subject: Re: [Asterisk-Users] Queue Management and Command Execution Yes, I read them. But, then my question is: how can I make the file name include the agent that will get the call once it's distributed? Thanks, Daniel On Apr 27, 2005, at 3:40 AM, Anton Krall wrote: You can record queue conversations, check out the configs in queue.conf |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Daniel Salama |Sent: Martes, 26 de Abril de 2005 05:53 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: [Asterisk-Users] Queue Management and Command Execution | |Is there a way to execute a command prior to sending a queued |call to an agent? | |What I'm trying to do is record agent's conversation at the |server. I could put the Monitor command when the call is |answered by *, but if the caller has to wait on hold for some |time, I wouldn't want to record that. | |What I would prefer is to be able to put the caller in a queue |and once an agent is ready, for the Monitor command to kick |in. Even better would be to know the agent id where the call |is going to be sent to, so I can use it as part of the file |name of the Monitor command. | |Any clues on how to do this, if at all possible? | |Thanks, |Daniel | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Determinating Phone status
Hi, how can I determine the status (busy, offline, ringing, duration of current call) of an SIP phone? Elmar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Checking for a sound file
Causes me to wonder a couple of things. Why does ANYONE use Outhouse or Outhouse Express? There are many much more friendly Windows E-mail clients, from Mozilla on down Tight integration with Exchange 2003. Find me an alternative client that is as stable and that has such tight integration and I'll jump ship immediately. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extensions / Contexts
Perfect, that's exactly what I need. I will try that, thanks a lot. Sebas Matt Riddell wrote: Sebastian Silva wrote: Hi everybody, I am writing here because I can't find the solution to my problem (my asterisk configuration). I hope somebody can give me a hand with it: I need to provide a PBX service to several companies (extensions with softphones and Digium hardware to manage the analog lines), my problem is that I don't know how to configure the contexts to have, for instance, the following scenario: Company A ext 2000 ext 2001 ext 2002 Company B ext 2000 ext 2001 ext 2002 Company A must not to see extensions of company B and viceversa. ok. So, use: extension.conf [incomingline1] include = companya exten = s,1,Answer() exten = s,2,Background(welcome_to_companya) [incomingline2] include = companyb exten = s,1,Answer() exten = s,2,Background(welcome_to_companyb) [internalcompanya] include = company_a include = voicemail_a include = utils_a include = dialout_a ... [internalcompanya] include = company_b include = voicemail_b include = utils_b include = dialout_b ... [companya] exten = 2000,1,Dial(SIP/2000a) exten = 2001,1,Dial(SIP/2001a) exten = 2002,1,Dial(SIP/2002a) [companyb] exten = 2000,1,Dial(SIP/2000b) exten = 2001,1,Dial(SIP/2001b) exten = 2002,1,Dial(SIP/2002b) sip.conf [2000a] username=companyA_2000 context=Companya [2000b] username=companyB_2000 context=Companyb -- Sebastian Silva G R U P O G A U S S Depto. Sistemas Av. Libertador 6250 4 piso Tl.: 4 706- (int. 121) [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to use dialparties.agi
Christian As I understand it After a user dials an extension number, Asterisk calls dialparties.agi dialparties.agi checks the asterisk database (show database [from cli]) for data matching items like Call Wating (CW) Call Forward (CF) etc. If one is present (in a defiend order) then rather than dialing, dialparties invokes that option. If none of the options are set, dialparties returns control back to a near regular dial string, and Dial takes over and places the call as the A party was expecting. Using defined etensions (by default in AMP they are the regular American ones), the B party (callee) can activate these features. What basically happens here is a database put command is used to put the value in the asterisk database and then play a recorded anouncement to the user before hanging the call up. for CF its a little more complicated as you might have to specify the B number and the C number, but essentially it puts the data in the database and confirms it Now the only thing that is missing is a web / gui provsioning system - so that admins can take the features off again, else its a databse del command at the terminal --- the best way to see this in action is to set some things like CW (*73 i think) and then do a show database at the CLI - you will also get back other things like the SIP registery David On 4/26/05, Christian Wengel [EMAIL PROTECTED] wrote: Hi! I looking for an example how to use the dialparties.agi from Asterisk Management Portal 1.10.007a. I tried to understand it by reading the extensions.conf of AMP, but without success. Is anybody out there, who can give me a more easy example or an explanation. Thanks, Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco SIP Firmware Price Increase
I've heard a few times that the firmware for Cisco Phones to use them with SIP is going to increase $150. Is this true? If you follow the cisco license agreements, yes. The cisco list price of the 7960 (as an example) includes their non-sip software for something like $650. (Street price is substantially less.) If you want that brand new phone with a sip license (and software), then there is an add-on for the sip license (something like $100 US, now making that a $750 phone). If you buy a used/reconditioned phone from any non-cisco-sponsored reseller, you're supposed to pay for a new license for whatever software is installed in the used phone (even if its the original software shipped from cisco with the phone). That cost is very high, and oriented to discourage reselling any cisco equipment. (Many authorized resellers get around it as they install whatever software you want and don't bother reporting it to cisco, or charging for it, etc.) In other words, whatever software is installed on any cisco box is _not_ transferable to the next buyer. Rather interesting from the standpoint that one _can't_ remove the software from a cisco 7960, so there is no way to ever resell a used cisco phone legally. (No way to comply with the license terms.) Cisco _does_ keep track of serial numbers by customer, therefore if you try to purchase a license or maintenance from _any_ source, that source has to validate the serial number against the cisco records. If you were not the original purchaser of that serial number, they are not supposed to sell you the license or maintenance agreement. (Just another way to control the used equipment market.) Good product though. ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream BT101 Firmware
Use this - http://gs-firmware.gratissip.dk/ Works fine, instructions are clear here. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Adam Goryachev Sent: Wednesday, April 27, 2005 10:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Grandstream BT101 Firmware I've been trying to upgrade a grandstream BT-101 phone, whatever I do, it doesn't seem to want to upgrade. I've pointed it at the grandstream TFTP IP listed on the wiki, I've also tried pointing it at my own TFTP server (after putting all the firmware images downloaded from grandstream website into /tftpboot). Is there any specific trick I should know (eg, upgrade in some sequance, or hack my tftp server in some way, special files/etc..) Also, any comments on what the overall best firmware is for these handsets currently would be interesting... Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Redirect two channels to each other?
I've been scratching my head trying to think of a way to do this, but without success yet. I'm using the Manager API. If I have two channels linked to each other (i.e. direct connection), or even if they are independent channels, I can transfer them both to the same extension by using Action: Redirect and using Channel: for one and ExtraChannel: for the other. This is most useful for putting them both in the same Meetme conference. What I want to do is find a way to take two unrelated existing channels (which for the sake of argument might be sitting in MusicOnHold, separate conferences, the same conference or whatever), and link them together into a direct call rather than having them talk via their own Meetme conference. Does anyone have any ideas if this can be done? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialing out from remote.
AMP does exactly this, why not look at their dial plan for instructions. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Daniel Dziubanski Sent: Wednesday, April 27, 2005 8:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Dialing out from remote. I'm attempting the following set up. During Hours - Receptionist Takes the call (no problem works great) After hours I would like to add a item to the receptionist to transfer the call to my cell, any direction would be a great help. I have 4 PSTN incoming lines as backup and Voicepulse. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connection Timeout problem with SIP phones from Gnet
Hey guys,
I'm fairly new to Asterisk. Our objective is to have a VoIP PBX
connected to our PSTN lines. So, right now, I have a box running OpenNA
Linux, with a 2.4.29 kernel. Asterisk 1.07 and the latest Zaptel drivers
also.
I have 2 Gnet SIP phones connected on the same switch as the Asterisk
box. So far, our phones authenticate with *, because when I do sip show
users, I see our 2 phones there.
The problem I have is this, when I try to dial the other extension, in
this case 502, from 501, after a few seconds, I get a busy signal. If I
check on the phone's logs, it says connection timeout.
Here's my dialplan, keep in mind, all the outgoing and incoming stuff is
irrelevant, since there's no PSTN line connected to it. Only the VoIP
matters for now.
extensions.conf:
[globals]
JIEF=SIP/501
TEST=SIP/502
[incoming]
exten = s,1,Answer()
exten = s,2,Playback(goodbye)
exten = s,3,Hangup()
[internal]
exten = _5XX,1,Dial(SIP/${EXTEN})
include = outgoing
[outgoing]
ignorepat = 9
exten = _9NXXNXX,1,Dial(${LOCALTRUNK}/${EXTEN:1})
exten = _9NXXNXX,2,Playback(invalid)
exten = _9NXXNXX,3,Hangup
[prompts]
exten = *1,1,Answer()
exten = *1,2,Record(test:gsm)
exten = *1,3,Playback(test)
exten = *1,4,Hangup()
And here's sip.conf:
[general]
port=5060
bindaddr=172.16.1.200
srvlookup=yes
dtfmmode=inband
allow=all
[501]
type=friend
host=172.16.1.201
canreinvite=yes
context=internal
username=501
secret=1234
allow=all
dtfmmode=inband
[502]
type=friend
host=172.16.1.202
canreinvite=yes
context=internal
username=502
secret=1234
allow=all
dtfmmode=inband
Cheers,
--
Jean-Francois Theroux
Systems administrator
PrivalODC
450.761.9973
http://www.privalodc.com
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Re: [Asterisk-Users] Determinating SIP Phone status
Elmar Haneke wrote: Hi, how can I determine the status (busy, offline, ringing, duration of current call) of an SIP phone? Remember that the SIP phone is a kingdom of it's own. Right now, Asterisk does not really now anything about what is happening out there in the SIP woods. We know about our own interactions with the phone, and you will see the status of our calls to the phone with SHOW CHANNEL lskdfjs. If you set limits in sip.conf, you can check what's in use with SIP SHOW INUSE. In the future, if someone is willing to sponsor that development, we could subscribe to the status of the phone or accept PUBLISH notifications so that we actually know the status of the phone. /Olle ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Re] Re: [Re] Re: [Asterisk-Users] Alternatives to SpanDSP??
Cyril VELTER wrote: I just installed it and will keep you informed if a new crash occur, but even with pre15, crash where not very frequent and usually come in series (~ one serie of 3/4 crashes every two weeks, so we might have to wait some time...). I'm pretty happy with the receiving side of spandsp (I don't use the sending side yet), processing about 60 incomming fax per days from a lot of differents sender. The success rate is quite good, but there is ~2 or 3 fax per day which are truncated or with missing pages. I'm wondering if implementing ECM should improve this and if you plan to do it someday ? Intermixed with the T.38 work I am doing, is work to flesh out the T.30 implementation to be complete. Of course, that will include ECM. Someone is working on making HylaFAX play nicely with spandsp, so HylaFAX does queuing and spandsp does the FAX transfers. If that all works out we should have a very nice FAX platform. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Using Asterisk to dial a number and then wait to dial the extension
Tony Mountifield wrote: Andrew Elchuk [EMAIL PROTECTED] wrote: I did some searching and haven't found a solution to my problem. But right now we are performing a transition from an old system to a new system using asterisk and only a few people are on the new system and testing it out. Anyways, I was wondering if it is possible to dial a phone number with asterisk, and then after that callee picks up to dial an extension? In one forum message I found that you could use 'w' in the dial string to act as a half second wait. I tried doing: exten = 109,1,Dial(ZAP/g1/6525798ww109) This would dial the other phone system, but would not wait 3 seconds til the other system answered and then dial the extension. I also tried using: exten = 109,1,Dial(ZAP/g1/6525798|D(109)) But this did the same thing as the above. Is there another way to dial a number then on the same channel send 3 more digits after the other party answers? Thanks. The D() option is the correct way to do it, but only works if your Zap interface can tell when the remote party answers. Typically, digital lines (ISDN, T1, E1) can tell, but analogue lines can't. Cheers Tony Is there a way to determine if the zap interface is able to know when the other party picks up? It is connected to analog lines by the way. Also, is there a way that might not be as correct but would none the less still work? Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Re] Re: [Re] Re: [Asterisk-Users] Alternatives to SpanDSP??
Hi Steve, Good work. process_baud is a fairly big routine, and your backtrace doesn't give the actual line number at which things fall over. However, studying the code I see that I do not protect against the possibility of a divide by zero during the initial coarse carrier estimation of any of the fast modems. I just created 0.0.2pre16, which should eliminate this. Can you try it, and tell me what happens? I just installed it and will keep you informed if a new crash occur, but even with pre15, crash where not very frequent and usually come in series (~ one serie of 3/4 crashes every two weeks, so we might have to wait some time...). I'm pretty happy with the receiving side of spandsp (I don't use the sending side yet), processing about 60 incomming fax per days from a lot of differents sender. The success rate is quite good, but there is ~2 or 3 fax per day which are truncated or with missing pages. I'm wondering if implementing ECM should improve this and if you plan to do it someday ? Thanks for your work, Cyril ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alternatives to SpanDSP??
Cross posting on purpose to transition the thread to -dev The issue in this thread is the frame transfer rate for the TDM analog card almost always exceeds the 1.000 seconds expected by the design. The frame transfer rate seldem impacts voice (the missed frames aren't noticed), but seriously impact code such as spandsp. From: Andrew Kohlsmith [EMAIL PROTECTED] On April 27, 2005 09:04 am, Rich Adamson wrote: I would sort of disagree with the spiking thingie (now). If you modify the zttest app to provide timing output in terms of seconds and microseconds, you don't see the spiking impacting those measurements. Rather, you see 8,192 bytes arriving in something greater then 1.000 seconds on a very consistent basis. Do you have a copy of this patch? I'd like to work on this problem with you (in my ample spare time, ha!). No I don't. I just inserted printf's in the 80+ line app to inspect the actual timing values (as opposed to viewing that mostly meaningless percentage number). The design of the card (and asterisk) is 100% oriented around receiving 8,192 bytes from the card every 1. seconds exactly. Any significant variation from 1.000 seconds will result in a missed frame (1024 bytes) sooner or later. *nod* What I've not been able to figure out is why the delay. I'm 95% sure it has more to do with asterisk code (including drivers) then it does with other system interrupt handlers, interrupt latency, etc. Those _other_ things certainly can impact it, but there is definitely something within asterisk that is directly related to the TDM card and its drivers. (Its almost consistent enough to look closer at the clocking on the TDM itself. That assumes a clock on the TDM card is responsible for raising the interrupt to the O/S via the pci bus.) Well the clock on the TDM400P is the same as what is used in the T100P, X100P (or is it X101P?) and TE110P. It's just a cheap crystal oscillator within the TJ320 so at least in theory the same problem should exist with those cards if it were an oscillator issue. That crystal oscillator is supposedly a standalone component that drives whatever other chips (on the card) the designer wants to use if for. Presumably, it is driving the 3050 (I didn't check). But, through some mechanism, the 3050 is serially sending pcm data bytes to the TJ320, and it appears _it_ buffers up that data and raises the pci interrupt to the O/S. So, any component associated with that process is including in my definition of clocking the interrupts (not just the crystal). Even cheap oscillators are more accurate than this though. :-) I'm curious though if the CPU spiking in the wctdm driver has something to do with it (causing the time to stretch), especially since this isn't seen on the other cards, only within that driver, and it's only that card that seems to have it. If one includes a couple of printf's to watch the seconds and microseconds used in the zttest calculation, then execute 'zttest -v', the reported times will consistently be something like 1.021234 seconds. Even though vmstat shows the spiking, it does not show up in the time reported for the zttest to receive 8,192 bytes of data. That would suggest the spiking isn't the root cause for the TDM card's missed frames. Since the vmstat spiking occurs roughly every ten seconds, one would expect it to have an impact on at least some of the zttest output. But, I've not seen that happen as yet. Opinion: the TDM analog card is subject to a number of system level issues, but underlying those issues seems to be an asterisk-code problem (including drivers) that does not support receiving the expected 8,192 bytes from the TDM card in 1. seconds. (According to Steve Underwood, that was not a problem about six to nine months ago, but it is now.) (I'll reply to your original post about the zttest stuff in -dev and we can continue this there.) I'll modify the zttest.c app and post the mod's on the -dev list, and maybe we can narrow down the root cause for the TDM issues. Direct eamil for those that want is fine ([EMAIL PROTECTED]). I'll be out of the office for the remainder of today, but will continue with this later today or tomorrow morning. Rich ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Redirect two channels to each other?
As ny 10 year old step-daugher says I don't get it.. Can't you just do a redirect if you specify the channels, * doesn't care if they are bridged together or not. You may end up with zombie channels if the other leg does not drop, but you could do a soft hangup and take care of that.. Or am I missing something -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Wednesday, April 27, 2005 10:41 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Redirect two channels to each other? I've been scratching my head trying to think of a way to do this, but without success yet. I'm using the Manager API. If I have two channels linked to each other (i.e. direct connection), or even if they are independent channels, I can transfer them both to the same extension by using Action: Redirect and using Channel: for one and ExtraChannel: for the other. This is most useful for putting them both in the same Meetme conference. What I want to do is find a way to take two unrelated existing channels (which for the sake of argument might be sitting in MusicOnHold, separate conferences, the same conference or whatever), and link them together into a direct call rather than having them talk via their own Meetme conference. Does anyone have any ideas if this can be done? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Redirect two channels to each other?
On Wednesday 27 April 2005 10:40 am, Tony Mountifield wrote: I've been scratching my head trying to think of a way to do this, but without success yet. I'm using the Manager API. If I have two channels linked to each other (i.e. direct connection), or even if they are independent channels, I can transfer them both to the same extension by using Action: Redirect and using Channel: for one and ExtraChannel: for the other. This is most useful for putting them both in the same Meetme conference. What I want to do is find a way to take two unrelated existing channels (which for the sake of argument might be sitting in MusicOnHold, separate conferences, the same conference or whatever), and link them together into a direct call rather than having them talk via their own Meetme conference. I have no ideas, other than Meetme. It sounds like it would involve some direct modification of the * code. Its similar to the pickup code - perhaps start there. Let me know if you find anything - id be intersted in a solution for it as well, i just dont have the time to find a solution. -josiah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco SIP Firmware Price Increase
Not that I know of I am a Cisco partner and the Category 1 contract is still at least half that or less. He was talking about the SIP-license... Not the SmartNET. If you have a SmartNET, you CAN download the SIP load but to use it, you need the license. I think that's the point; to use sip please pay an additional $150US. Downloading the image is supposedly illegal unless you have a license. Now, what is the true list price of a new 7960 with sip? (Be careful to read the license terms before answering that question.) see Global Pricelist section Cisco IP Telephony Phone User Licenses: SW-SMH-UL-7912 SIP license for single 7912 IP phone D $80 SW-SMH-UL-7912= Spare SIP license for single 7912 IP phone S $80 SW-SMH-UL-7905 SIP or H.323 license for single 7905 IP phoneD $80 SW-SMH-UL-7905= Spare SIP or H.323 license for single 7905 IP phone S $80 SW-SM-UL-7960 SIP and MGCP license for single 7960 IP phoneD $150 SW-SM-UL-7940 SIP and MGCP license for single 7940 IP phoneD $150 So, as you see, the license - at least for 7940/7960 - already costs $150... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Determinating Phone status
ChanIsAvail Show application Chanisaval -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Elmar Haneke Sent: Wednesday, April 27, 2005 10:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Determinating Phone status Hi, how can I determine the status (busy, offline, ringing, duration of current call) of an SIP phone? Elmar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phone Recommendation.
On Tue, 26 Apr 2005, Dana Olson wrote: You mean like the problem I described earlier on this list? http://lists.digium.com/pipermail/asterisk-users/2005-April/103153.html I am not sure why I didn't think of disabling call waiting, but that seemed to work with a Grandstream BudgeTone phone... I'm doing more testing now. Sounds exactly like the same problem. Of course, the $65 grandstreams allow you to disable call waiting.. The stupid $130+ Polycom's don't. :( -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- sean a. newton [EMAIL PROTECTED] louisville, ky, usa http://wewt.net Another day, another convertible and another hotel full of cops.-- Hunter S. Thompson -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Determinating SIP Phone status
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Wednesday, April 27, 2005 11:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Determinating SIP Phone status Elmar Haneke wrote: Hi, how can I determine the status (busy, offline, ringing, duration of current call) of an SIP phone? Remember that the SIP phone is a kingdom of it's own. Right now, Asterisk does not really now anything about what is happening out there in the SIP woods. We know about our own interactions with the phone, and you will see the status of our calls to the phone with SHOW CHANNEL lskdfjs. If you set limits in sip.conf, you can check what's in use with SIP SHOW INUSE. In the future, if someone is willing to sponsor that development, we could subscribe to the status of the phone or accept PUBLISH notifications so that we actually know the status of the phone. /Olle Olle, If one can get the 'PUBLISHED' info from the SIP device would that give us a better solution than having to use the hint thingamabob to see if the person is on the phone ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Using Asterisk to dial a number and then wait to dial the extension
Andrew Elchuk wrote: Tony Mountifield wrote: Andrew Elchuk [EMAIL PROTECTED] wrote: I did some searching and haven't found a solution to my problem. But right now we are performing a transition from an old system to a new system using asterisk and only a few people are on the new system and testing it out. Anyways, I was wondering if it is possible to dial a phone number with asterisk, and then after that callee picks up to dial an extension? In one forum message I found that you could use 'w' in the dial string to act as a half second wait. I tried doing: exten = 109,1,Dial(ZAP/g1/6525798ww109) This would dial the other phone system, but would not wait 3 seconds til the other system answered and then dial the extension. I also tried using: exten = 109,1,Dial(ZAP/g1/6525798|D(109)) But this did the same thing as the above. Is there another way to dial a number then on the same channel send 3 more digits after the other party answers? Thanks. The D() option is the correct way to do it, but only works if your Zap interface can tell when the remote party answers. Typically, digital lines (ISDN, T1, E1) can tell, but analogue lines can't. Cheers Tony Is there a way to determine if the zap interface is able to know when the other party picks up? It is connected to analog lines by the way. Also, is there a way that might not be as correct but would none the less still work? Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Nevermind I got 'er working. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Redirect two channels to each other?
Maybe this would best be explained in a diagram: 1). person A --- music on hold and person B --- music on hold 2). *some manager API action* 3). person A --- person B This is what I think he's asking about, how do you take two parties on different conversations and put them together without using a meetme conference? MATT--- -Original Message- From: Alexander Lopez [mailto:[EMAIL PROTECTED] Sent: Wednesday, April 27, 2005 11:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Redirect two channels to each other? As ny 10 year old step-daugher says I don't get it.. Can't you just do a redirect if you specify the channels, * doesn't care if they are bridged together or not. You may end up with zombie channels if the other leg does not drop, but you could do a soft hangup and take care of that.. Or am I missing something -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Wednesday, April 27, 2005 10:41 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Redirect two channels to each other? I've been scratching my head trying to think of a way to do this, but without success yet. I'm using the Manager API. If I have two channels linked to each other (i.e. direct connection), or even if they are independent channels, I can transfer them both to the same extension by using Action: Redirect and using Channel: for one and ExtraChannel: for the other. This is most useful for putting them both in the same Meetme conference. What I want to do is find a way to take two unrelated existing channels (which for the sake of argument might be sitting in MusicOnHold, separate conferences, the same conference or whatever), and link them together into a direct call rather than having them talk via their own Meetme conference. Does anyone have any ideas if this can be done? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IP Softphone Recommendations
Also try the snom soft phone: http://www.snom.com/snom360softphone.html. Sorry, Windows only:-( But at least its free! Enjoy, CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Salas M Sent: Wednesday, April 27, 2005 12:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IP Softphone Recommendations Ing CIP Alejandro Celi Mariátegui wrote: El mar, 26-04-2005 a las 09:42, Guillermo Salas M escribió: I´m using X-lite on windows and linux, looks pretty well. Do you have the link of the X-Lite Linux version? Not found in the xlite website. Saludos desde Ecuador. g Go to http://support.xten.com and register for an account. Later, send an email to [EMAIL PROTECTED] requesting being a beta tester. For testing purposes, you can download the latest version from: http://xten.com/apps/xprolinuxbeta/xlite-linux-24.bz2 To installl: run this command on the file donwloaded: bunzip2 xlite-linux-24.bz2 The result is a file xlite-linux-24, which is the executable, you simply run it from the command line. You may need to do a: chmod +x xlite-linux-24 first to make it executable. Regards from PERU... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote Phones - No Audio In Either Direction
From: [EMAIL PROTECTED] on behalf of Sean Kennedy [EMAIL PROTECTED] Posted At: 26 April 2005 21:25 Conversation: [Asterisk-Users] Remote Phones - No Audio In Either Direction Posted To: Asterisk-Users Subject: Re: [Asterisk-Users] Remote Phones - No Audio In Either Direction Paul Tyreman wrote: Hi, After months of testing Asterisk, I am finally ready to roll it out, replacing my previous VOIP server (brekeke's ondo SIP Server), which was very restrictive. However, I am experiencing some problems with phones which are on a different network to the server (connecting via the internet). I have managed to get the phone to register with the Asterisk server, and I can make a call and hear it ringing, but once connected no audio can be heard in either direction. I have opened the following ports: 5004, 5060, 5061 and 1 - 10010 on my router, but am still having no joy. When I used ondo, I had to add my WAN IP address to the configuration files, so I was wondering if I have to do that in some .conf file in Asterisk ? Hope someone can help ? Thanks, Paul. I had this problem on a vpn ( highly recommended, given how easy it is to implement openvpn now ). I changed the IP address in the SIP file to my server ( 192.168.1.1, remember, on a vpn ), everything just worked. Good luck Sean Thanks Sean, Can I add two lines to bindaddr in sip.conf, so its like this; bindaddr = 10.x.x.x; Local IP address bindaddr = 81.x.x.x; WAN IP address So that both internal and external phones see the server ? Paul. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] QuadBRI card on Suse 9.2 Unable to load qozap.ko
Hi, I successfully installed zaptel,libpri,asterisk and qozap in a Suse 9.2. I removed the old modules loaded as default by Suse. Now I'm triying to load qozap.ko but I receive this error: insmod: error inserting 'qozap.ko': -1 Unknown symbol in module and in dmesg: qozap: unsupported module, tainting kernel. qozap: disagrees about version of symbol zt_receive qozap: Unknown symbol zt_receive qozap: disagrees about version of symbol zt_ec_chunk qozap: Unknown symbol zt_ec_chunk qozap: disagrees about version of symbol zt_transmit qozap: Unknown symbol zt_transmit qozap: disagrees about version of symbol zt_unregister qozap: Unknown symbol zt_unregister qozap: disagrees about version of symbol zt_register qozap: Unknown symbol zt_register I found in google thi errors but with no success. Someone can help me ? Bye ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Confused on G723 and G729
I see that G723 and G729 require a license to be used, or can be used (in the case of G723) in pass-through mode only. My question is.. if my voip terminator supports G723 and G729 only, do I still need a license? Or is that considered pass-through? If so, do I need to do anything special to get it to work? I'm also a litle confused about why G723 can do pass-through but can't do voicemail access? What's the difference, or the logic behind this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Redirect two channels to each other?
In article [EMAIL PROTECTED], mattf [EMAIL PROTECTED] wrote: Maybe this would best be explained in a diagram: 1). person A --- music on hold and person B --- music on hold 2). *some manager API action* 3). person A --- person B This is what I think he's asking about, how do you take two parties on different conversations and put them together without using a meetme conference? Thanks Matt, that is exactly what I am asking. I assume you haven't found a way either, otherwise you would have mentioned it! :-) Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call a peer over the asterisk manager with a php script
Guy Boehm wrote: fputs($socket, Channel: 6159bfb47b9\r\n\r\n); Response: Error Message: Invalid channel the Channel: var needs to be in the form of type/dev/numbertocall like Channel: IAX2/user:[EMAIL PROTECTED]/14085551212 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
