Re: [asterisk-users] Pickup re-invite
I have 800 kbps in both directions reliably at the endpoint location. When I was testing, there weren't any computers in the office, or any other phones. The server has a 10 Mb ethernet connection in a datacenter, and I usually don't see more than 8 channels at once, so I don't think it's bandwidth. The endpoints I have been testing on have been rock solid in all other modes of operation, except Pickup. Asterisk is trying to do an external RTP bridge, as evidenced below. How do I make it not do that. I have already specified canreinvite=no for all peers. nat=yes for all the peers except the upstream carriers. It's also set as the global default. Retransmitting #6 (NAT) to (Phone Public IP):1126: INVITE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Via: SIP/2.0/UDP (Server IP):5060;branch=z9hG4bK5e02a020;rport From: *88 sip:[EMAIL PROTECTED] Domain;tag=as64bce3f7 To: T S St. Pierre sip:[EMAIL PROTECTED] Domain;tag=d8b4a9e50086b57 Contact: sip:[EMAIL PROTECTED] IP Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Communicate Freely 1.4 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge)- Why!!! Content-Type: application/sdp Content-Length: 266 On Tuesday 11 December 2007 00:58, dave cantera wrote: tim, sounds like a problem I had with bandwidth... too many devices communicating on the same network connection to the internet... have you tcpdump'd or used a bandwidth tool to see what the usage is? nat=yes or nat=no? should be yes.. did you change the router between upgrades? just some random thoughts.. daveC Tim St. Pierre wrote: Hello Folks. I'm wondering if anyone has any helpful hints. I recently upgraded to 1.4.11, and I'm having problems with pickup, both directed, and the pickup feature. My server is on the public internet, and all phones are behind a NAT router, somewhere else on the public internet. When a ringing phone is picked up by another phone, you have audio for a few seconds, then the call is dropped. The console shows No response to our critical packet A SIP debug of the conversation between the phone and the server shows a re-invite request right when the call drops. The phone is of course using the internal IP address as it's contact, and it looks to me like the server is trying to use it. I have canreinvite=no for both the general sip.conf, as well as per-peer. I am using the whole range of Aastra Enterprise IP phones. Interestingly enough, some phones show their true IP address and port in the Asterisk registration database. I believe this is where the phones have successfully communicated with a uPNP router, and discovered their public address. These phones can successfully pickup the call. If I pipe the pickup call through the Local channel, it works. Why is asterisk still trying to re-invite even though I have explicitly told it not to in the config? It worked fine in 1.2 Any suggestions, or requests for more information? Thanks for any help. -Tim -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pickup re-invite
Hello Folks. I'm wondering if anyone has any helpful hints. I recently upgraded to 1.4.11, and I'm having problems with pickup, both directed, and the pickup feature. My server is on the public internet, and all phones are behind a NAT router, somewhere else on the public internet. When a ringing phone is picked up by another phone, you have audio for a few seconds, then the call is dropped. The console shows No response to our critical packet A SIP debug of the conversation between the phone and the server shows a re-invite request right when the call drops. The phone is of course using the internal IP address as it's contact, and it looks to me like the server is trying to use it. I have canreinvite=no for both the general sip.conf, as well as per-peer. I am using the whole range of Aastra Enterprise IP phones. Interestingly enough, some phones show their true IP address and port in the Asterisk registration database. I believe this is where the phones have successfully communicated with a uPNP router, and discovered their public address. These phones can successfully pickup the call. If I pipe the pickup call through the Local channel, it works. Why is asterisk still trying to re-invite even though I have explicitly told it not to in the config? It worked fine in 1.2 Any suggestions, or requests for more information? Thanks for any help. -Tim -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voip providers and sip origination and termination?
You are seeing the difference between a resale product and a wholesale product. Origination and termination and telecom terms used to describe which way the call is going. Time costs - no matter what. Even if the provider pays a flat rate for their PRIs, the capacity multiplied by the number of minutes in a month is the maximum time they have to sell. Retail packages can provide an unlimited number of minutes by estimating the usage of their customers, and charging a rate that covers this. There is generally a user agreement in place to enforce their estimates. Origination and Termination are wholesale products where you have unlimited (within practical limits) access under their user agreement, because you pay a rate that is proportional to the cost of them providing the service. In terms of back end, an origination and termination service will often allow you to set callerID number, and will allow a peering arrangement that is much more conducive to having several incoming DID numbers, and multiple channels from the same asterisk machine. -Tim On September 10, 2006 15:43, Christopher Corn wrote: can someone please explain the differnces to me??? I have an asterisk system im setting up for a small office (4 or 5 phones) and as im looking for a voip provider, i find that voip providers generally have unlimited plans, and those that offer sip origination and termination get charged for the minute, for their outgoing and incoming calls. is there a difference in the backend architecture here? if so, what? or is this is just a difference in marketing terms and setup? for example, http://www.broadvoice.com offers an unlimited plan in the US for calls, though they never use the term sip origination and termination. they say their systems also supports asterisk. yet http://www.bandwidth.com/content/enterprise?page=voice_services_origination _terminationcampaignId=7013JBJ calls it sip origination and termination any info is appreciated! thanks! -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpZIGsdzJlIU.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] music onhold choppy music problems
There is a different mpg123 that is included with asterisk. It seems to work a lot better than the other version that gets installed from a port or package. I'm not sure why, but try removing your existing one, and run make mpg123 install from the unpacked directory. On September 10, 2006 17:02, Matt wrote: anyone has any success in using music onhold. even if we have ztdummy installed we still got choppy music. buy our old asterisk 1.0.9 is working, why is that? thanks for any help in advance. Best Regards matt -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgp1Lpf4ZkjVQ.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can someone recommend a voip provider that...
vitel isn't bad. They have a nice asterisk-friendly interface, and their rates are good. Quality has been fine. -Tim On September 10, 2006 17:51, Christopher Corn wrote: ok maybe thats asking for too much. how about a voip provider that provides 729 codec support ? :) Christopher Corn [EMAIL PROTECTED] wrote:offers unlimited calls, in and out in the US asterisk support no setup fee and support 729 codec? and of course is reliable and clear thanks alot. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpBr1kxL9AEm.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Max Size of Conf Files
I have an extension.conf composite (including all the included files) that is over 2000 lines. I do all my rating in the dialplan and it seems to work just fine. I produced these from spreadsheets containing cost vs. number information for overseas calls, so it has to pattern match every call against an 1800 line context. -Tim On September 10, 2006 19:08, Steve Totaro wrote: I have almost 1,000 800 numbers that are routed any number of ways. Currently I call on fastagi which checks a database and returns the extension or route that DID is supposed to take. The DIDs are all over the place as far as sequence, so pattern matching is out of the question. My question is, is there a max file size for a conf file? Will defining the routing for each of the 1,000 DIDs in extensions.conf effect performance or eat up huge amounts of RAM? I assume these numbers get inserted into the BerkleyDB at startup and reload, can it handle it? Any other pros or cons to still having a database, but instead of fastagi asking for an exten on the fly, the database writes the conf file only when routing is changed or toll free numbers are added? Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpl70WkFlf9c.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using residential voip for business?
They do this because business customers tend to use more minutes than residential customers, and an unlimited plan is always an ESTIMATION of usage. It costs them for every minute you use, so they try to sell residential customers a block of time, and call it unlimited, which it really isn't (read the fine print). I have found that unlimited plans VERY rarely cost less than paying per minute in any situation. Do the math. -Tim On September 10, 2006 19:42, Christopher Corn wrote: I spoke to a voip provider today who mentioned that though they offer an unlimited plan, if we use it for a business and it is over-utilized, it will be canceled. is this true for all residential voip plans? i have a small office of about 4 or 5 phones. i tend to chose residential plans because they have the unlimited offer for outgoing/incoming. thx -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpvrORX41lrW.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting system time via Asterisk
You could probably build one easily. Create a context in extensions.conf that answers the call, asks them the time, saves the digits as a variable, then passes these variables as arguments to a shell script using the System() application. Create a shell script that takes it's arguments and uses them to set the clock. On September 10, 2006 19:51, Gary Eck wrote: Looking at a Asterisk server which will not be attached to the Internet - and the user will be pretty computer illiterate. Has anyone seen a script or some mechanism to set the server time by using an extension, and entering the date/time via the keypad? Thanks! -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgp59wbqWm8Q0.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Context
That is the default behavior. If you don't include the contexts into each other, they can't call each other. On September 11, 2006 03:35, Khaled Chehab wrote: Dear I have two contexts how could I isolate context A from context B ,in other words I want to ban context A from calling context B Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpIXF6QsrpyF.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP parameter to prevent a call from being added in missed calls logs
This isn't really possible at the asterisk level. The phones log missed calls as calls that ring, but are not answered. It's not possible to have a call ring, and not be answered, but still ring at the phone. On September 11, 2006 05:07, Olivier wrote: Hi, If you set Asterisk to ring several extensions for an incoming call, it appears that the call will be added in every phone's missed calls logs though the call was picked by one extension. In the long run, this prevent users from using missed calls features as these logs would filled with many calls which have been responded by another team mate. Is there any parameter in SIP protocols which would ask individual phones not to add a given call to its missed calls logs ? If negative, how would set Asterisk so that only missed calls are only recorded once ? Regards -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpZ8tznYtrb2.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk
Make a context called DID or something like that, and set your peer entry in sip.conf to have your provider's calls go tho this context. The incoming SIP invites will be directed to the DID [EMAIL PROTECTED] server. Use Goto to direct the calls where you want them to end up. ie. [DID] exten = 6477226929,1,Goto(phones|5101|1) exten = 6477226930,1,Goto(ea-mainmenu|s|1) On September 11, 2006 07:30, Richard Klingler wrote: hello If I want to use asterisk to hookup to a SIP account I just use the register line in sip.conf with the extension number at the end... But how about if I want to use a SIP trunk from a provider which gives me 10 DID numbers with the same account? thanx in advance rick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgp8w4fk2qtMY.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Forwarding in SIP.conf
If you don't set the callerID in the channel, it will get passed on as-is. Don't change it, and it will stay the same. -TIm On September 9, 2006 12:27, [EMAIL PROTECTED] wrote: I have a follow up question. How do I pass on the caller ID of the call I'm forwarding to the other party? I can pass on the channels caller ID but prefer to pass on the forwarding party's number instead. -- Original message -- From: [EMAIL PROTECTED] Thanks all. It works fine now. -- Original message -- From: Tim St. Pierre [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpt5tN7gJk5i.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scope of contexts
Nope. A context ends when a new one starts. The only way for a call to
continue is to have a maching extension, and the next higher priority. If
you want a call to continue in another context, you need to use the Goto()
application.
On September 9, 2006 19:04, Rene wrote:
Hi all,
I am trying to understand contexts a bit better. The problem I have is
when you know when a context is finished. Is this when a new context
starts?
Example:
[context1]
exten = _9170X,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
[context2]
exten = 6394,1,Dial(Local/6275/n)
When your call starts at context1, will it automatically go to context2
when context1 is finished?
Hope someone can shed some light on this
Thanks,
Rene
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sip://[EMAIL PROTECTED]
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Re: [asterisk-users] Whcih phones are better for mass deployment
I am a really big fan of Aastra phones. It's a splinter company of Northern Telecom, so their quality is very good. Provisioning is done via a text file on either a tftp or an ftp server. There is a global file and a per phone file. When you have a good set of config files built, you can include an option for the phones to check the files every day at a predetermined time and reboot if there are any changes. You can also send an SIP NOTIFY to cause the phones to update their config if you change something and need it applied immediately. There is no config utility needed, as the files are human readable. There is an encryption utility if you are concerned about security. When you deploy a new phone, you need only set the TFTP server address. After that, the phone can get all it's settings from the server. I have about 40 of them deployed at client sites that I usually don't have access to. I can change everything from here. Sound quality is great, most of them support PoE and have a passthru ethernet port. The displays are backlit, there is a full duplex speaker phone and headset jack on all models. There is also a built in directory function that loads from a .csv file on the server. BLF support is good on the 9133i and the 480i. I can't say enough good things about these phones. Manufacturer support is also very good. Free firmware downloads from the website and good documentation. -Tim On September 9, 2006 18:51, Zeeshan Zakaria wrote: I am having hard time with grandstream phones for a 30 phone setup. When a change in configuration is required, I have to change their configurations manually for almost all of them. Their configuration utility is not very straight forward to use. For my next installation, I would prefer some other phones with better configuration and remote accress utility. My question to those of you with more experience, what IP phones are better for mass deployment and easy management of updates and configurations? Or what other solution is better for mass deployment of phones? -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpUZT8t5RNms.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Whcih phones are better for mass deployment
I'll speak on the Aastra, since that is what I know, although most of this applies to Polycom as well. There is no windows software needed at all. Personally, I haven't been a Microsoft customer in more than half a decade. Their operating systems are not appropriate for telecom applications. You can do all your configuration with a text editor. This is good for several reasons: 1) You can administer the phone config directly on the server over an ssh connection. 2) A shell script can create phone config files. I have a shell script that appends to extensions.conf, sip.conf, voicemail.conf, and creates a phone config file. You can automate things very easily this way. 3) Since there is only one setting to put into the phone to use the remote config, a customer can be talked through a factory reset and reset the server address over the phone if they really screw things up. You can have an inventory of phones with your config server address already set. All you need is the phone MAC address, and you can build a config file. This means that you could send phones to customers without them having decided what to do with them yet. 4) These phones are reliable and well constructed. They will require less maintenance, and will last longer. 5) They have features that are appropriate to a business environment. 6) They will usually find their way around a NAT firewall, so they are essential plug-and-play at the customer site. Let me know if you have any more questions. -Tim On September 9, 2006 23:01, Zeeshan Zakaria wrote: Can you explain a little bit what make them better for mass deployment. Do they have Windows based software to communicate with all the installed phones and upgrade them and also to remotely monitor them. Is there a separate cost for these software tools or are they free? -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpKBfHPOMykJ.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip peer question
It's all in the Asterisk database, which is a Berkeley DB format as far as I know. If it's just for migration, you could probably just move the database. If you want to do this while the system is up, maybe using an external database for this information would work better. -Tim On September 8, 2006 05:36, Dijkstra, Roelof wrote: Hello, We currenty have an asterisk cluster running, with a quad PRI and a quad BRI. This all works pretty well. What i was wondering: If i do a show sip peers I see all the ip addresses of the phones that registered, also, when restarting the server. Is there any way of copying this information to another server? Regards, Roelof Dijkstra Network Engineer EMEA Compuware Europe BV The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpkEje1c9N6M.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Forwarding in SIP.conf
Do you have gafachi-o in your sip.conf? Since it's not a valid host name, you need to have an entry in sip.conf to tell asterisk how to make a call to gafachi-o. That's why it is telling you No such host. On September 8, 2006 12:57, [EMAIL PROTECTED] wrote: It sounds like a good idea, I tried it and get this error Sep 8 09:52:17 WARNING[27193]: chan_sip.c:1968 create_addr: No such host: gafachi-o Sep 8 09:52:17 NOTICE[27193]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) In Extensions.conf I have exten = 4305,1,Dial(SIP/[EMAIL PROTECTED]) ; permit transfer In Sip.conf I have [4305] type=friend user=4305 secret= ;context=from-sip callerid= host=dynamic nat=yes canreinvite=no dtmfmode=rfc2833 ;incominglimit=1 ;[EMAIL PROTECTED] ;disallow=all ;allow=ulaw ;allow=alaw ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! -- Original message -- From: William Piper [EMAIL PROTECTED] whatever the did is needs to be put in the extensions.conf told to dial your cellphone. Example: exten = _011123445566,1,Dial,SIP/[EMAIL PROTECTED] assuming that your using a SIP carrier, replace 1234567890 with your cellphone 1.2.3.4 with the carrier's IP or carriers context name in sip.conf. bp On 9/8/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I'm using it for virtual numbers. I have international virtual number from a DID provider and want to forward it to my cell phone. In Sip.conf I have the channel [4305] type=friend user=4305 secret= ;context=from-sip callerid= host=dynamic nat=yes canreinvite=no dtmfmode=rfc2833 ;incominglimit=1 ;[EMAIL PROTECTED] ;disallow=all ;allow=ulaw ;allow=alaw ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! and in extensions.conf I have exten = 4305,1,Dial(SIP/4305,120,rt) ; permit transfer This had worked in the past when I forwarded it through the Linksys ATA but now have run out of ATA's. -- Original message -- From: Tim St. Pierre [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpJihN8D8XLj.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Forwarding in SIP.conf
Check your Dial() string to make sure that you haven't mistyped and put gafachi-o instead of gafachi-out. Specifiying the full host name will also work. As a hint, you can refresh these changes with out restarting your server (and therefore without disrupting any calls in progress) extensions reload will refresh the extensions file reload will reload all your configs sip reload will reload only sip configs (and re-register everything) Very handy when working on an active machine. On September 8, 2006 14:19, [EMAIL PROTECTED] wrote: Tim, this is the way I have Gafachi set up in sip.conf and works well with channels that have an ATA attached to it but not the virtual one. I have changed the host in extensions.conf to the .sip.gafachi.com. But I have calls on the server and cannot restart it yet. I'll keep you posted and thanks for the feedback. [gafachi-out] type=peer secret=xx username=x fromuser=x fromdomain=xxx host=.sip.gafachi.com ;usereqphone=yes; This provider requires ;user=phone on URI ;nat=yes rtptimeout=60 dtmfmode=rfc2833 -- Original message -- From: Tim St. Pierre [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpSQeinqKcDG.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
With SIP, asterisk processes the digits it receives in the invite from the Polycom. There is no communication of dialplan information in SIP. The polycom should send the digits as soon as you press dial. You can program the polycom with a dialplan that will tell it when to send the digits, but that only works if you dial off-hook. I like on hook dialling, since it sends what i tell it, when I tell it. This should never happen when you press dial - it should try right away. My 301 does this, maybe they changed something in the newer firmware? -Tim On September 8, 2006 14:33, Mike wrote: I've been running into an issue with my Polycom 501 and Asterisk. I realized, after much mucking around, that when I dial a number (and press the send key) that is invalid , but could still match an Asterisk pattern (example: I dial 567, which is not a valid extension, but my diaplan accepts _567 as a pattern) instead of sending the call as is and ultimately failing, the phone is intelligent enough to sit and wait for extra digits in case I meant to dial 567111. Now thats a problem for me. How can I make Asterisk (or the 501) treat the attempted extension 567 as a valid try and let Asterisk handle the error ?(instead of the phone trying to do what it think is best and handling the error on it's own). Is there an Asterisk setting for that? Failing that, is there a Polycom setting to disable this intelligent error handling? Mike -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpHvcCeRVwOA.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] distinguishing users by their domain
Isn't there a way to specify a context based on the incoming domain in sip.conf? On September 8, 2006 14:03, Ricardo Carvalho wrote: So... does anybody know how can I do this? Maybe using a way to distinguish users not by their username, but by other fields of SIP INVITE messages? Regards, Ricardo. Ricardo Carvalho wrote: In extensions.conf I want to implement a dial plan that distinguishes the users that wish to dial a PSTN number by their own domain, so that [EMAIL PROTECTED] goes out to PSTN by a different DID than [EMAIL PROTECTED] I tried the following line, but that doesn't distinguish between domains, and then if [EMAIL PROTECTED] or [EMAIL PROTECTED] dials some PSTN number, both calls goes out using same DID (did1): exten = _[0-9]./john,1,Dial(SIP/[EMAIL PROTECTED],120) I tried then using the following lines: exten = _[0-9]./[EMAIL PROTECTED],1,Dial(SIP/[EMAIL PROTECTED],120) exten = _[0-9]./[EMAIL PROTECTED],1,Dial(SIP/[EMAIL PROTECTED],120) But those syntax doesn't work. How can I do it? Any clues? Thanks, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgp8ng4FvO5vJ.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No dialtone, just directly busy
Could you send us some CLI output? Look for something like this Invalid extension s in context whatever your dial context is It could be that lifting the handset without dialing is opening a channel to the s extension, since there are no digits being dialed. There is a workaround for this, but it means creating a dialplan that produces dialtone and waits for digits. -Tim On September 8, 2006 14:44, Henrik Woffinden wrote: Hi, I'm using Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s. I've got 3 ISDN phones attached. When I want to dial out I can do it in 2 ways.. 1) Type in number with handle still on.. Lift handle and we dial the number 2) Lift handle and then press the number Both methods should work, but only the first does. With the second I expected a dialtone but it goes immedately to busy signal. No dialtone first. Why is that? -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpGxKd71JHts.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
Now that is really odd. Try sip debug peer (peername of the polycom) This will let you see the sip packets go by when you do this, so you can see the responses it is, or isn't getting. I'll have to look up the SIP response codes, but I do know that there is one for not found which should correspond with an invalid extension. Because the call is not actually set up yet, asterisk will return a not found message rather than answer the call, only to direct it to an i extension. This is only used for calls already in progress. I don't know if there is a sip response for need more digits or something like that. Turning on the sip debug will tell you EXACTLY what the polycom is saying to asterisk, and vice versa. Note: I like to hit scroll lock after I hit call, before I hangup so that it doesn't fill my screen up with all the cancel messages - that will put you just below the important parts of the data. On September 8, 2006 15:21, Mike wrote: Thanks Tim. I've been trying to find out what's happening. Basically, somehow, it seems that my Polycom 501 knows what extensions are valid and which aren't in my dialplan. Obviously, the 501 doesn't really know that, but Asterisk seems to return it this info (sort of :valid, invalid or could be valid, need more digits to know) when I press send. I know it sounds mad, and I would love nothing more than being told I am an idiot because or x and y. Why do I feel that this info is passed from Asterisk to the 501? Well, take the following (very simple) dialplan [context_a] Exten = 1234,1,Noop(foo) Exten = _9,1,Noop(bar) Exten = i,1,Noop(invalid) What happens when I dial out is the following: 1) 1234: Noop(foo) ; good 2) 4: A congestion tone is heard from the phone (but Asterisk's CLI doesn't show anything...no sent into invalid extension '4' in context 'context_a', but no invalid handler 3) 934 : It's invalid, but it could match the pattern is I added some digits. I expect an invalid extension message, but what actually happens is the phone tries the send something (I can see an icon moving on the phone) but the phone stays quiet (no stuttering tone or whatever). It waits, I can input more digits on the phone. Let's just take 1) and 2). Why is Asterisk not going into the i extension like it should? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim St. Pierre Sent: September 8, 2006 2:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? With SIP, asterisk processes the digits it receives in the invite from the Polycom. There is no communication of dialplan information in SIP. The polycom should send the digits as soon as you press dial. You can program the polycom with a dialplan that will tell it when to send the digits, but that only works if you dial off-hook. I like on hook dialling, since it sends what i tell it, when I tell it. This should never happen when you press dial - it should try right away. My 301 does this, maybe they changed something in the newer firmware? -Tim On September 8, 2006 14:33, Mike wrote: I've been running into an issue with my Polycom 501 and Asterisk. I realized, after much mucking around, that when I dial a number (and press the send key) that is invalid , but could still match an Asterisk pattern (example: I dial 567, which is not a valid extension, but my diaplan accepts _567 as a pattern) instead of sending the call as is and ultimately failing, the phone is intelligent enough to sit and wait for extra digits in case I meant to dial 567111. Now thats a problem for me. How can I make Asterisk (or the 501) treat the attempted extension 567 as a valid try and let Asterisk handle the error ?(instead of the phone trying to do what it think is best and handling the error on it's own). Is there an Asterisk setting for that? Failing that, is there a Polycom setting to disable this intelligent error handling? Mike -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgp3K2UaXtL4T.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto Dialer question
You need to make a cron job that runs a script that creates a spool file. If you dump a file into /var/spool/asterisk/outgoing/ it will create a call based on the information within the file. Do a search for asterisk spool outgoing for all the formats and examples. I have used it and it works rather well. -Tim On September 8, 2006 22:39, Hall, Eric M. wrote: Hello group I have a customer that has asked me to build an auto dialer that will call customer a few day before an appt and remind them of the time and date of the appt. Does anyone have any good links for apps that could do this type of auto calling? They also request that information be pulled from a database and be able to pull reports on who was called and if they call was picked up. Thanks for any help the group could give me! Eric -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgppIGrMT1ean.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No dialtone, just directly busy
Try this exten = s,1,Disa(no-password) It's a dirty hack, but it might work. It will dump the phone straight into the disa application, which will play dialtone and allow you to dial into the current context. -Tim On September 8, 2006 16:12, Henrik Woffinden wrote: That's exactly what happens: When I pick up the handle, this is what I get: -- Extension 's' in context 'from-inside' from '11' does not exist. Rejecting call on channel 0/2, span 2 Do you know what to do in the dialplan? Best regards, Henrik Woffinden Tim St. Pierre wrote: Could you send us some CLI output? Look for something like this Invalid extension s in context whatever your dial context is It could be that lifting the handset without dialing is opening a channel to the s extension, since there are no digits being dialed. There is a workaround for this, but it means creating a dialplan that produces dialtone and waits for digits. -Tim On September 8, 2006 14:44, Henrik Woffinden wrote: Hi, I'm using Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s. I've got 3 ISDN phones attached. When I want to dial out I can do it in 2 ways.. 1) Type in number with handle still on.. Lift handle and we dial the number 2) Lift handle and then press the number Both methods should work, but only the first does. With the second I expected a dialtone but it goes immedately to busy signal. No dialtone first. Why is that? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpDGQsB6ZG9U.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Forwarding in SIP.conf
Call forwarding doesn't go in sip.conf, it has to go in the dialplan.
You set up your outbound provider in sip.conf
In your dialplan, you use the Dial application like this:
exten = _NXXNXX,1,Dial(SIP/outgoingprovider/${EXTEN})
This will dial out to a PSTN number based on the extension passed to it.
What is it you want to do? Call forwarding on not-registered or no answer?
That needs a database and a macro. What is your goal?
-Tim
On September 7, 2006 17:14, [EMAIL PROTECTED] wrote:
I looked through the forums but could not find exactly what I needed. I
need help setting up call forwarding in sip.conf, where the call forwards
to PSTN number without a sip phone but just the channels in sip.conf
without any hardware or softphone. Any help will be greatly appreciated.
--
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IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
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Re: [asterisk-users] cmd SET time value
I had to do the same thing for Goto(). Perhaps there is a limitation when
using the pipe character in variables. That's a bit of a nuisance.
On September 8, 2006 01:33, Benjamin Jacob wrote:
Ok, Had to work around this one. An innefficient implementation, but
here goes :
exten =
s,n(getFwdTime),Set(fwdTimeHrsMins=${DB(CFWDTimeHrsMins/${ARG1})}) ;
exten = s,n(getFwdTime),Set(${IF(${LEN(${fwdTimeHrsMins})}
0?fwdTimeDays=${DB(CFWDTimeDays/${ARG1})}|fwdTimeDay=${DB(CFWDTimeDay/${ARG
1})}
|fwdTimeMonths=${DB(CFWDTimeMonths/${ARG1})})})
exten =
s-dialFwdTime,1,GotoIfTime(${fwdTimeHrsMins}|${fwdTimeDays}|${fwdTimeDay}|$
{fwdTimeMonths}?s-dialFwd,1)
==
Basicaly, am storing individual entries of time, and putting them back
together in GotoIftime.
If anyone's got a better solution, lemme know.
cheerz.
Ben.
Benjamin Jacob wrote:
Nope Tim,
had tried that already, duznt work.
Here's the cli output
===
Executing Set(SIP/4000-097afc90, fwdTime=*|mon-tue|*|*) in new stack
Sep 7 12:22:20 WARNING[24964]: pbx.c:6062 pbx_builtin_setvar:
Ignoring entry 'mon-tue' with no = (and not last 'options' entry)
Sep 7 12:22:20 WARNING[24964]: pbx.c:6062 pbx_builtin_setvar:
Ignoring entry '*' with no = (and not last 'options' entry)
==
with my macro line being
exten = s,n(getFwdTime),Set(fwdTime='${DB(CFWDTime/${ARG1})}')
Ben.
Tim St. Pierre wrote:
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Date:
Thu, 7 Sep 2006 00:28:20 -0500
Single quotes - ' - work when I set other variables that contain
special characters. Give that a try,
-Tim
On September 6, 2006 23:18, Benjamin Jacob wrote:
Hello ppl,
Ive a couple of macros defined to call fwd based on time to a
number/voicemail.
Very elementary.
=
11. [macro-dialexten]
12. exten = s,1,Dial(SIP/${ARG1}) ;
1. [macro-stdpbx1exten]
2. exten = s,1,Set(fwdedNum=${DB(CFWD/${ARG1})})
3. exten = s,n,GotoIf(${fwdedNum}?getFwdTime:dialExten)
4. exten = s,n(getFwdTime),Set(fwdTime=${DB(CFWDTime/${ARG1})}) ;
5. exten = s,n,GotoIf(${fwdedNum} !=
VoiceMail?s-dialFwdTime,1:s-vmFwdTime,1) ; goto VoiceMail or dial
Fwded num
6. exten = s,n(dialExten),Macro(dialexten,${ARG1}) ; dial Called exten
7. exten =
s-vmFwdTime,1,GotoIfTime(${fwdTime}?s-vmFwdTime,vmFwd:s,dialExten)
;if
fwdTime not set or time matches,
;send to VM, else dialExten
8. exten = s-vmFwdTime,n(vmFwd),VoiceMail(${ARG1})
9. exten =
s-dialFwdTime,1,GotoIfTime(${fwdTime}?s-dialFwdTime,dialFwd:s,dialExt
en)
;if fwdTime not set or time matches,
; call fwdedNum, else dialExten
10. exten = s-dialFwdTime,n(dialFwd),Macro(dialexten,${fwdedNum})
===
I save the fwdedNum in DB, and also the fwding time.
Now, when i retrieve the time value from db and set it, using cmd SET,
it takes only the initial part of the time value string.
e.g. if time to be checked is *|mon-tue|*|*, the time set is * ONLY!!
The cmd Set's syntax uses the | (pipe) notation to separate variables.
Thats why this behaviour.
Any work around this guys??
Thanks in advance
Ben.
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--
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sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]
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Re: [asterisk-users] using SIP to connect remote other VoIP server
Could you be more specific? Do you want to set up linking between two asterisk servers? Is this to a service provider? A single SIP registration and peer entry will handle multiple channels, and can also handle different numbers at the destinations. Try to get away from thinking of things in terms of lines PRI and VoIP use channels and routing instead. A SIP registration and peer statement is used to tell a servers where to find each other. You could have multiple calls going to different extensions using only one entry. It's all about how you set up your routing. What is it that you want to do? -Tim On September 6, 2006 21:27, tengulre wrote: How to using SIP to connect remote other VoIP server? is it only running one line voice if I registered a one SIP account? anybody can give me some sample configuration files? thanks a lot! -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpwY6afK4LQD.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cmd SET time value
Single quotes - ' - work when I set other variables that contain special
characters. Give that a try,
-Tim
On September 6, 2006 23:18, Benjamin Jacob wrote:
Hello ppl,
Ive a couple of macros defined to call fwd based on time to a
number/voicemail.
Very elementary.
=
11. [macro-dialexten]
12. exten = s,1,Dial(SIP/${ARG1}) ;
1. [macro-stdpbx1exten]
2. exten = s,1,Set(fwdedNum=${DB(CFWD/${ARG1})})
3. exten = s,n,GotoIf(${fwdedNum}?getFwdTime:dialExten)
4. exten = s,n(getFwdTime),Set(fwdTime=${DB(CFWDTime/${ARG1})}) ;
5. exten = s,n,GotoIf(${fwdedNum} !=
VoiceMail?s-dialFwdTime,1:s-vmFwdTime,1) ; goto VoiceMail or dial
Fwded num
6. exten = s,n(dialExten),Macro(dialexten,${ARG1}) ; dial Called exten
7. exten =
s-vmFwdTime,1,GotoIfTime(${fwdTime}?s-vmFwdTime,vmFwd:s,dialExten) ;if
fwdTime not set or time matches,
;send to VM, else dialExten
8. exten = s-vmFwdTime,n(vmFwd),VoiceMail(${ARG1})
9. exten =
s-dialFwdTime,1,GotoIfTime(${fwdTime}?s-dialFwdTime,dialFwd:s,dialExten)
;if fwdTime not set or time matches,
; call fwdedNum, else dialExten
10. exten = s-dialFwdTime,n(dialFwd),Macro(dialexten,${fwdedNum})
===
I save the fwdedNum in DB, and also the fwding time.
Now, when i retrieve the time value from db and set it, using cmd SET,
it takes only the initial part of the time value string.
e.g. if time to be checked is *|mon-tue|*|*, the time set is * ONLY!!
The cmd Set's syntax uses the | (pipe) notation to separate variables.
Thats why this behaviour.
Any work around this guys??
Thanks in advance
Ben.
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--
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IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]
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Re: [asterisk-users] Asterisk Linksys PAP2 ATA
Glad to help. Happy dialling. On September 2, 2006 23:05, Nick Ellson wrote: Hi Tim, The dial plan trick worked great. Added |40[01]x| to my plan and 4000-4019 connect instantly from the PAP2 :) Added it to my X-Lite as well, and worked there too. Thanks! -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpa0WjPdmz6f.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware ? Analog DID trunks (ILT)
I have never tried this, but what about an analog FXS card, set to use featd or em_wink signalling? The FXS card will supply battery (digium hardware actually supplies the appropriate voltages). You would just have to use the appropriate signalling type to provide the winks. -Tim On September 2, 2006 09:45, Jerry Jones wrote: Do not know of a card that does. But think a digium T1 to a channel bank (ie Adit600) would. On Sep 1, 2006, at 2:06 PM, Tim Sharp wrote: I am looking at CTPX's VP2000 product. I haven't tried it yet. Please let me know if you find a solution that works. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] Behalf Of Jonn R Taylor Sent: Friday, September 01, 2006 12:15 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Hardware ? Analog DID trunks (ILT) Is there a card that supports analog DID trunks, alosi known as ILT trunks or Incoming Loop Trunk. They work by providing talk battery to the CO, incoming calls happen by pulling loop sending a wink accepting the DID dtmf digits for the station being called. Jonn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpXaGTpPVvL6.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfer 3/4 digits
Are you using # to transfer? If so, it's not sending it as a new call, it's just sending asterisk digits using whatever DTMF mode. Asterisk parses these based on a first match in the dialplan. Make sure that the longer extension numbers are loaded first in the dialplan. -Tim On September 2, 2006 20:12, Ronald Wiplinger wrote: Kevin Smith wrote: Dialing a number and transferring a number are two different things. And no offense, you are not really providing a lot of details along with your problem. So you can dial the numbers but not transfer from one to the other. I was not thinking that it would be too much difference. Therefore I also do not know what more info could help to distinguish the problem. I hardly can post my entire configuration. What does the CLI say when you try the transfer? That would provide a lot of information that could clue you in to what is going on. You hit another problem with that. I hardly see here anything anymore. The messages fly by so fast, Especially annoying messages: chan_sip.c:10888 handle_request_register: Registration from 'sip:192.168.250.20' failed for '192.168.250.244' - Username/auth name mismatch -- Got SIP response 486 Busy Here back from 192.168.250.244 -- Got SIP response 400 Bad Request back from xx.xx.xx.126 NOTICE[5936]: chan_sip.c:9600 handle_response_register: Failed to authenticate on REGISTER to '[EMAIL PROTECTED]' (Tries 3) . It would be nice to filter the CLI for such investigation for a moment. What type of phones are you using? Some phones have the ability to pattern match and wait for a certain number of seconds before sending the number to asterisk. For example. On our Polycom phones a user has 3 seconds (between digits) to enter in 10 digits. This could be where most of your problem is. That is a very good point and I will contact the manufacturer of these no-name phones. My guess the problem lies with the Phones, not Asterisk form the information you provided. I disagree with that! Why Asterisk treats dialing and transfer different. That makes not really sense, does it? bye Ronald Kevin Ronald Wiplinger wrote: David Gagnon wrote: Ronald, You seem to be a little bit angry about VoIP. If so, I could give you my old Nortel system. Does this would make you happy? David David, I am not angry about VoIP, but please send my your old Nortel system ! I just do not understand why I can DIAL 601 and 6014, but not use blind transfer. Is the question too difficult? I am sure there is somewhere a switch to say, wait two seconds (as for dialing) before you assume it is a complete number. It is also strange that snom phone can do it correct, because it uses the ok key. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ronald Wiplinger Envoyé : 2 septembre 2006 04:20 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Blind transfer 3/4 digits Anthony Rodgers wrote: With respect, the problem is with your numbering plan.. This answer is therefore totally nonsense !!! (With all respect!!!) Both answers have actually not lead to any step further, but to more messages. I use to refer to such answers as NON-ANSWERS. Please only reply if and really only if you know a solution for the problem! Thanks for your understanding. bye Ronald - again, I am not angry at all. WHERE do you see a problem in the numbering plan? I see the problem in ASTERISK, because it does not wait for the last digit!!! Where can I set that it waits for it? The beauty on voip IS that you can have different length and overlapping, bye Ronald CP On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote: I found a problem in blind transfer: I have an extension number 601 and I have an extension 6014 If I get a call on 615 (snom) and transfer to 6014 it works, since snom requires me to hit ok If I get a call on 601 and transfer to 6014, than 601 will get the busy signal and I hang up as usually with transfer. Howerver the caller get the announcements: I could not get that, What could be the problem ? bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpy80N6wIb40.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to send correct Caller ID on PRI
Somewhere in your outbound routing section of the dialplan, you need to have this line: ,n,Set(CALLERID(number)=whateverthenumbershouldbe) Personally, I like to set a variable in sip.conf, perhaps PSTNCALLERID, that I use in the above line. That way I can set PSTN caller ID numbers on a per extension basis. On September 2, 2006 18:58, Zeeshan Zakaria wrote: Hi, While dialing calls form my client's office, where I was working, the caller ID goes as the extension number of the phone from where caller is calling. I tried to playaround with config files, also changed info in their Grandstream GX-2000 phones, but to no avail. What am I missing here and how to take control of the outgoing caller IDs? Thanks -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpV0oYwkLnKI.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Roundrobin not working on PRI
You probably have to set all your PRI channels as part of a trunk group. Additional calls to the same number should show the same number. Make sure that when they hit your dialplan, there is somwhere for a second call to go (ie. a queue, voicemail, another extension, etc.) -Tim On September 2, 2006 19:07, Zeeshan Zakaria wrote: Hi everybody, My client had just installed a PRI in his office for his phone line, with 30 DIDs. Main phone number ends in 1900 and DIDs last 4 digits are from 3570 to 3599. Now when caller calls number ending in 1900, call comes in with DID 1900, and asterisk answers it. Second caller calls, call comes in again as DID 1900, asterisk rejects it because the line is busy. So the caller gets message from the PSTN side, that line is busy. My understanding was that caller will call number ending with 1900 and call will come in on Asterisk with DID 3570. Another caller will call and call will come in as DID 3571. And this was supposed to be done by the PRI service provider. Am I doing something wrong here, do I have to configure something in Asterisk, or do I have to call the service provider. They say that everything is ok on their side and our system needs to be configured properly. -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpikms68XpsE.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Linksys PAP2 ATA
There is a dialplan setting in the advanced config. If you modify this to recognize your three or four digit extension pattern, it will dial instantly after you dial an extension. -Tim On September 2, 2006 19:14, Bob Chiodini wrote: Nick, I know some adults that can have an entire conversation in the same amount of time. Does pressing the # key speed up dialing? If so look for a timer in the PAP config or tell the kids to press #. IIRC the spa3k had something similar, but never did much in-house dialing. $86 is a pretty good price. I paid more than that for the spa3000 6 months ago. Bob... Nick Ellson wrote: Hey Bob, I think the SPA31-2 is the new guy on the block. Only $10 more too mail order. $86 was the best I saw. So I have the PAP2 with two cheapy $4 wall phones mounted in the kids room, they are calling each other and my laptop.. Only issue so far is that to call one PAP2 from the other there is a 10 sec delay before the ringback/ring occurs.. and a 3 5 year old can have an entire conversation before the phone even rings. ;) Calling from my X-Lite soft phone to the PAP2 is nearly instant. But it does have my wife actually jazzed about having two more phones where she works in the house so she can join the fun.. Score! A free pass to buy more toys! Another PAP2 and a SPA3102 for me Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpQvEnWrmFMX.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID has extra digits to strip
You could create a function that uses GotoIf() to detect the extra digits. The line it points to could strip the extra digits. What version of asterisk are you using? (the functions are different pre-1.2.1) On September 2, 2006 18:37, Bart Fisher wrote: About 70% of the time, my Local DID provider sends me ANI II digits (see http://www.nanpa.com/number_resource_info/ani_ii_assignments.html) where there will be an extra 2 digits added to the Caller ID - For example 62714222 where '62' = Cell Phone for example.. The problem is, I have not found a way to remove these digits before it's used by Asterisk in CDR and Voice Mail with any Asterisk script or command. What can I do to strip these digits from Caller ID before answering the call so CDR and Voice Mail Caller ID announcement show correct number? TIA Bart ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpVBb3xmIRpI.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Linksys PAP2 ATA
You have to set in in the PAP2. When using SIP, it has to send an invite with the number it wants to be connected to. The Sipura has to know a complete number to send - it can't send it in pieces. You need to make the dialplan in the Sipura match what you have programmed in Asterisk. Ie. My extensions are 51XX, and 52XX, so in the Sipura dialplan, I added 5[12]XX - this will match any of my extensions, and complete the call. This can be a problem if you use direct 10 digit dialing, and dial to an area code beginning with 51 or 52. You could get around this (if it's a likely issue) by prefixing a 9 to the 10 digit patterns, or inserting a . (I think) to make it wait for another digit. -Tim On September 2, 2006 20:43, Nick Ellson wrote: Hey Bob, Just tested the PAP2, yes a # sends right away. I am looking for why, still new at the dial plan stuff.. this is the default.. Should I be looking for a way to have the PAP2 NOT deal with dialing and let Asterisk handle it? (*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.) -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpukDjjdyjsf.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail as email and attachment
Try putting a comma between the e-mail address and |attach=yes On September 1, 2006 00:51, Benjamin Jacob wrote: Tim and guys, The sendmail daemon was indeed down. So i turned it on, but messages not going thru(maybe some sendmail config, will investigate on that) I do see entries in /var/log/maillog, and a line which says, Mail accepted for delivery. That aside, the line that I had mentioned(in my voicemail.conf) , 5600 = 5600, Benjamin Jacob, [EMAIL PROTECTED]|attach=yes In maillog, I see the To address to be [EMAIL PROTECTED]|attach=yes . Is it some bug in asterisk mailing code, or am i doing something wrong? Ben. Tim St. Pierre wrote: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Date: Thu, 31 Aug 2006 22:18:48 -0500 It looks like your configuration is fine on the asterisk side. I would look at your MTA configuration, as well as whether or not your ISP is blocking SMTP ports on the way out. A good way to test is to use the mail command line utility. Just type mail [EMAIL PROTECTED] Type some text, then enter twice, followed by control-d If you get the e-mail, you asterisk config needs to be fixed. If you don't get the e-mail, you need to figure out what is wrong with e-mail, not asterisk. -Tim On August 31, 2006 06:50, Benjamin Jacob wrote: Hello All, Am relatively new to Asterisk, but kinda slogging my ass off on it. My first couple of qs to begin with : 1) I tried the voicemail on no-answer thing. and my line in the voicemail.conf, duz have an email address and also attach=yes, 5600 = 5600, Benjamin Jacob, [EMAIL PROTECTED]|attach=yes I still havent really received a mail or the attachment. Don't I have to specify the mail server IP etc??I searched high and low for this. 2) For configuration changes, which is the best option to take up, use Asterisk Realtime, or Asterisk Manager APIs. Thanks in advance. Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpUaHSYwETTa.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Operator Console(s)/Shared Call Appearances
Here's something else that might be helpful.
;Splits the call so that the executive doesn't see his or her own name
needlessly)
[executive-extensions]
_X.,1,Dial,1,(SIP/${EXTEN}Local/[EMAIL PROTECTED]|20t)
_X.,2,Voicemail(${EXTEN}|su)
[receptioncue]
;This tells the receptionists who the call is for, and rings them all,
regardless of which executive the call is going to.
_X.,1,Set(CALLERID(name)=Exec-${EXTEN}${CALLERID(name)})
_X.,n,Dial(SIP/reception1SIP/reception2SIP/reception3)
This will ring all the phones, but will prefix the extension of the executive
being called to the name display on the receptionists phones. Additionally,
you could use set(SIP_INFO='info=alert-group') to make an Aastra phone only
ring once (but continue to be answerable). This will scale infinitely
without any config change - just send any calls to be handled this way to the
executive extensions context. If you want to use a name instead of the
extension, just create a line for each executive that replaces Exec-${EXTEN}
with their name.
-Tim
The above will ring the call on the executive phone, as well
On September 1, 2006 17:30, Mr. Jones wrote:
Hi Folks,
I'm back on this subject again. What's the best way to have both
phones ring simultaneously across say 3 operators + the executive?
TIA
On 7/24/06, Jerry Jones [EMAIL PROTECTED] wrote:
Asterisk does not yet support bridged calls
You can easily have a button labeled exec 1 ring on her phone at the
same time it rings the execs phone, and have one light if he is on
the phone
Also FOP works great
On Jul 23, 2006, at 3:42 PM, Mr. Jones wrote:
Thanks Sebastian -
You're right - I have limited experience in this area :)
I think the idea below is workable, except we actually want it to work
in the other direction - sort of.
Essentially we want the receptionist to screen the calls when she's
available. The executive should have option to answer the phone if its
after hours, or they know the receptionist isn't available (or perhaps
they recognize the caller ID and just want to take the call).
Can you think of how this might work? I suppose the executive could be
a member of his own queue?
What do you think about this idea;
1. Call comes in at one of the executive numbers.
2. Executive phone starts ringing for a predetermined time.
3. The callerid is changed to also reflect the name/number of called
executive, so that the receptionist knows for who the call was.
4. The call is dropped into a queue for the receptionist (queue
because
multiple calls to the receptionist at the same time are possible).
This setup isn't all that hard, and doesn't require more than 4 sip
accounts / phones and one queue, with one agent. Furthermore, if your
company starts to grow, and more receptionists that have to answer
the
phone are needed, it's quite easy, all you have to do is add a sip
account, one agent and add that agent to the existing queue. (About 2
minutes...)
--
Sebastian Berm
iPronto Communications
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sip://[EMAIL PROTECTED]
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Re: [asterisk-users] Can QUEUE member be assigned from a GlobalVar set in EXTENSIONS.CONF?
Here's an idea.
Set up static queues that go to named extensions using the Local channel, at a
specified context.
In this context, specify your mappings using the global variables
Ie.
; Queue
member = Local/[EMAIL PROTECTED]
;Extensions.conf
[globals]
TECHSUPPORT=SIP/5101
SALES=SIP/5102
[support-people]
exten = techsupport,1,Dial(${TECHSUPPORT}|20t)
exten = techsupport,2,Voicemail(${TECHSUPPORT}|su)
exten = sales,1,Dial(${SALES}|20t)
exten = sales,2,Voicemail(${SALES}|su)
This sets all the variables within extensions.conf, and uses the local channel
to translate the placeholders.
I haven't tried it, but it should work in theory.
On September 1, 2006 17:52, Gary G. Hendershot wrote:
what I would like to do is set some global variables in EXTENSIONS.CONF
then read the contents of these to use as members of a QUEUE in
QUEUES.CONF ...
example: (from [global] section of extensions.conf)
TECH-SUPPORT=SIP/5000; set value of TECH-SUPORT to be SIP/5000
example: (from [tech-support] queues.conf]
member = ${TECH-SUPPORT}
my intent here is have a central location where I can easily assign/change
members of the queue.
Should this work ??? I have tried it and it dont seem too but then I have
done something stupid in my syntax. I have confrmed that the
${TECH-SUPPORT} variable is assigned properly, is available and works as
expected within EXTENSIONS.CONF, but it seems it cant be used in
QUEUE.CONF.
The agents scenario will not work for my application as it is a small 5
person shop where everyone wears 5 hats. There really are no dedicated
agents and my users would object to having to logon/logoff and so on.
I can of course hard wire the QUEUES.CONF with the values, have tried
this and it works fine! I was just looking for lazy way to manage it. Any
ideas on this would be appreciated.
Regards
G.Hendershot
--
Tim St. Pierre
IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]
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Re: [asterisk-users] Asterisk Linksys PAP2 ATA
They work fairly well, just make sure it's a PAP2-NA, not one that is locked to Vonage. They are pretty easy to configure. -Tim On September 1, 2006 22:12, Nick Ellson wrote: I was loonking for an easy off the shelf ATA to get two analog phones up on Asterisk. I am not yet ready to by a full 4 port digium card until My wife can see this work with FWD and a real phone :) I see that Fry's sells the Linksys PAP2, which appears to be a SIP adaptor? I have found no posts on it being able to log into Asterisk. Any one tried this? Nick -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpNJaU6uXLbh.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What does 'trunk' mean in outgoing and incoming?
Traditionally, a trunk was a group of channels interconnecting switches. On a PBX, they are the incoming and outgoing lines to the rest of the world. A trunk is any channel that can carry a call to selectable destinations, as opposed to a subscriber line that only goes to one place. In the IP world, it's more of an organizational concept. I have separate config files - sip-trunks.conf and sip-phones.conf sip-trunks is where I put config to help my machine talk to other machines. sip-phones is were the configuration for phones go. It really doesn't matter, since it's all the same file. It's just saying it that way because that's what that type of connection would be doing in a traditional telephony way of thinking. On September 1, 2006 20:42, Larry Alkoff wrote: I'm configuring a Sipura SPA-3000 to go with my existing and working Asterisk 1.2.5 setup. The Sipura configuration files give an extension context [201] in sip.conf with the instruction This goes into the Incoming settings for your Trunk. It also gives a extension context of [pstn-spa3k] in sip.conf with the instruction This section goes into the Outgoing Settings for your Trunk. What does 'Incoming' and 'Outgoing' settings for your trunk mean? Where do trunks live and what are they meant to do? In my setup I have in sip.conf a [telasip-gw] context that references a context=telasip-in in extensions.conf. In extensions I have a [telasip-in] and [telasip-out] context. Which if any of these are 'trunks'? The Future of Telephony doesn't say much about trunks. Larry -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgp9Ia1oKzQcf.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] * during voicemail greeting to access mailbox
I have it working, it's a rather handy feature. You need to make the a extensions, as you have learned, pointing to the VoiceMailMain application. On my system, I use macros for the extension handling (ie. dialing an extension runs a macro that rings the phone, sends to voicemail based on the status of the extension, etc.) My I put this extension in the actual macro. Failing that, it might work if it's in the context that your extensions are in. It has to be in the same context that called the voicemail busy or no-answer application. Best of luck, let me know if it doesn't work, and we'll follow up from there. -Tim On August 30, 2006 23:39, Marty Mastera wrote: I'm trying to allow access to an individuals mailbox by having them dial their own DID, wait for their voicemail greeting and pressing * (to be followed by a password prompt). For some reason I thought that this functionality was built-in to Voicemail but must not be since it doesn't work...(I do see * being pressed in debug logs but no action follows). I did find the following in the wiki: Also. during the prompt if the caller presses: '*' - the call jumps to extension 'a' in the current voicemail context. This needs an example '#' - the greeting and/or instructions are stopped and recording starts immediately. When using the zero '0' and star '*' it's important to note that the context you placed the application voicemail in is irrelvant, it's the context for the voicemail box that we're looking for in the dialplan for the jump to the 'a' or 'o' extention. I tested this using a macro to call voicemail and pressing '0' and/or '*' first jumped back to the macro that called voicemail, to look for 'o' and/or 'a' in that macro, if it failed it jumpted to the context defined in voicemail.conf. I'm running CVS version 1.07 I have added an 'a' extension to the voicemail context but still nothing. Does anyone have this working? btw I'm using 1.2.1 Marty -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpZek6tUX9Md.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail as email and attachment
It looks like your configuration is fine on the asterisk side. I would look at your MTA configuration, as well as whether or not your ISP is blocking SMTP ports on the way out. A good way to test is to use the mail command line utility. Just type mail [EMAIL PROTECTED] Type some text, then enter twice, followed by control-d If you get the e-mail, you asterisk config needs to be fixed. If you don't get the e-mail, you need to figure out what is wrong with e-mail, not asterisk. -Tim On August 31, 2006 06:50, Benjamin Jacob wrote: Hello All, Am relatively new to Asterisk, but kinda slogging my ass off on it. My first couple of qs to begin with : 1) I tried the voicemail on no-answer thing. and my line in the voicemail.conf, duz have an email address and also attach=yes, 5600 = 5600, Benjamin Jacob, [EMAIL PROTECTED]|attach=yes I still havent really received a mail or the attachment. Don't I have to specify the mail server IP etc??I searched high and low for this. 2) For configuration changes, which is the best option to take up, use Asterisk Realtime, or Asterisk Manager APIs. Thanks in advance. Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpsLbKxazdut.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with recording
Try creating an extension with a lower priority that answers the channel first. If you don't, the application will run, but the call will timeout as no answer, since it was never actually answered. It sounds weird, but this is how you get messages like please check the number and try your call again without getting billed for the call. - Asterisk doesn't indicate answer until you tell it to, or until it bridges a call. -Tim On August 31, 2006 06:55, Giedrius Augys wrote: Hi, I am trying to record a speech with this command: exten = 205,3,Record(speech:wav). But it records aproximately about 10 seconds and asterisk hangs up. Does somebody know how to solve this problem, I also tried with max duration, but it didn't help.. -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpvHPpCvCVQ1.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 and Asterisk
The SPA3000 may be looking for proper -48v battery voltage to detect the line. A lot of gateways, PBXs, cheaper ATAs, etc. only put out about 24v or so. Most electronic phones can't tell the difference, but it's not up to specifications. Maybe an SPA3000 can. You can always check it with a voltmeter. -Tim On August 31, 2006 12:26, Mark Willis wrote: Francisco Seratti wrote: Hi pals, im trying to save some money in cellphones calls, so i bought a GSM gateway and a Sipura SPA3000 gateway. The GSM gw is currently working, and now im trying to configure the SPA, but every call i send, i get a 503 service unavailable. It does that if no line is plugged in. Mark -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpV03uwHYHO4.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
