[asterisk-users] PRI Crapping Out Regularly
We have a server with a TE120 on a partial PRI trunk that several times a day declares the PRI trunk down and stops handling calls until the asterisk is stopped, the zaptel/te120 modules reloaded, and asterisk started. Just before things go down, the log shows the following error: ERROR[9424] chan_zap.c: Write to 28 failed: Unknown error 500 at which point a show pri spans reports PRI span 1/0: Provisioned, Down, Active and a pri show span 1 reports: Primary D-channel: 24 Status: Provisioned, Down, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 (a) What is causing this? (b) How can it be fixed? (c) Why does Asterisk not recover automatically to what appears to be an intermittent problem? -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ --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] Looking for PSTN provider with unlimited inbound/outbound plan
Tell me when to stop laughing. Multiple channels and unlimited minutes ? No sane person will give that to you. On Dec 30, 2007 2:16 AM, Steve Finkelstein [EMAIL PROTECTED] wrote: Hi all, I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. Thanks for any potential leads. Happy holidays! - sf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan
Justin Case wrote: Tell me when to stop laughing. Multiple channels and unlimited minutes ? No sane person will give that to you. Yap I agree... but but for about $900 per month one could get T1 (24 channels) unlimited in/out as far I seen last time our providers rates. Senad On Dec 30, 2007 2:16 AM, Steve Finkelstein [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. Thanks for any potential leads. Happy holidays! - sf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Looking for PSTN provider with unlimited inbound/outbound plan
Senad, Mind if I ask who that provider is? Thanks. Sent from my iPhone On Dec 31, 2007, at 8:10 AM, Senad Jordanovic [EMAIL PROTECTED] wrote: Justin Case wrote: Tell me when to stop laughing. Multiple channels and unlimited minutes ? No sane person will give that to you. Yap I agree... but but for about $900 per month one could get T1 (24 channels) unlimited in/out as far I seen last time our providers rates. Senad On Dec 30, 2007 2:16 AM, Steve Finkelstein [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. Thanks for any potential leads. Happy holidays! - sf ___ --Bandwidth and Colocation Provided by http://www.api- digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Require IP Phones in Pakistan
Hello All, We need IP Phones in Lahore, Pakistan. Preferred brands are Atcom, Polycom and Grandstream. However any other good brand is also acceptable. Our client is interested in cheaper phones. Can anyone provide in Pakistan ? Regards, -- Kashif Naeem Director Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Gmail: [EMAIL PROTECTED] Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ --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] IVR help, please
Doug Lytle wrote: Jay Moore wrote: Hi list. I'm new to IVRs and trying to set up one that toggles an auto-forward flag on or off for specific accounts. Why don't you post what you've currently written and we'll go from there? Doug Actually, after switching to AEL, I think I finally got it working properly. Thank you for your response, however. Jay ___ --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] Polycom Digit Map
I need the digit map to call China. Example number: 011-86-10-6887- 011-International (obvious) 86 is country code (China) 10 is city code (Beijing) Last 8 digits are the number. I tried using 011xxx.T but it always asks me to enter more digits. Tried some variations as well, but no joy. -Michael ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Digit Map
On Dec 31, 2007, at 11:36 AM, Michael Munger wrote: I need the digit map to call China. Example number: 011-86-10-6887- 011-International (obvious) 86 is country code (China) 10 is city code (Beijing) Last 8 digits are the number. I tried using 011xxx.T but it always asks me to enter more digits. Tried some variations as well, but no joy. Yours should work if you wait long enough for t to timeout. How about 01186xx? Plus, IARC, when dialing offhook, pressing # should terminate dialing and send what it has at that point. ___ --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] How to use AddQueueMember with IAX2 peers?
Hi all, I've been working on this for days and can't find a solution. I need to use AddQueueMember for my agent logins to my Queues -- but a number of my agents are outside the main server, which is connected to my asterisk network over IAX2. I can't just do a AddQueueMember(queuename) because it puts in a complicated member calleridnum like: IAX2/peername:65723/23 Which won't exist when it comes time to transfer a call to that member. Help! I have tried using the chan_local formatted strings instead like Local/[EMAIL PROTECTED] -- but you lose all sorts of functionality if you do it that way -- -- Chris ___ --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] Digium Asterisk Appliance voicemail logs
Here is some information I received from my account rep at Digium regarding this information: -- Digium -- That's news to me as well as the rest of the sales team. We were told that users cannot change the 1gb flash card. I just spoke with one of the Sales Engineers and he stated that it is apparently possible to change out the flash card on the AA50. However, it is not supported because the read/write speeds could be different on a new flash card, and thus not work. -- Also here is what I was told when I asked my digium rep if we could nfs or samba mount additional storage space to the appliance (since it does run on linux). -- Digium -- It is not possible to do so. However, the 1gb Flash card on the appliance will store up to 3000 minutes worth of voicemail. -- Sincerely, Gregory Malsack President Classic Services Select Digium Reseller -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Saturday, December 29, 2007 7:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium Asterisk Appliance voicemail logs Barry D. Hassler wrote: Does anyone know how much space the appliance has for voicemail and/or logs? Doesn't have an embedded disk from what I can see, and only a 1G flash card? Correct. Nearly all of the 1GB CompactFlash card is available for voicemail, logs, CDRs, etc, and of course larger CompactFlash cards can easily be used. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.17.13/1204 - Release Date: 12/31/2007 12:20 PM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.17.13/1204 - Release Date: 12/31/2007 12:20 PM ___ --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] Directories Used by Asterisk
It is when you type 'make install' that these directories get created. 'make linux26' IS obsolete as another poster mentioned. broadband Voice wrote: I successfully obtained the Asterisk code and extracted them into /usr/src. When I make and install asterisk, zaptel, libpri etc. Are they supposed to move automatically into their respective directories? I cannot find: /etc/asterisk/ /usr/lib/asterisk/modules/ /var/lib/asterisk Do I have to manually create them or this is failed install? Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Asterisk Appliance voicemail logs
On Mon, 2007-12-31 at 12:02 -0600, Gregory Malsack wrote: Here is some information I received from my account rep at Digium regarding this information: -- Digium -- That's news to me as well as the rest of the sales team. We were told that users cannot change the 1gb flash card. I just spoke with one of the Sales Engineers and he stated that it is apparently possible to change out the flash card on the AA50. However, it is not supported because the read/write speeds could be different on a new flash card, and thus not work. -- Also here is what I was told when I asked my digium rep if we could nfs or samba mount additional storage space to the appliance (since it does run on linux). -- Digium -- It is not possible to do so. However, the 1gb Flash card on the appliance will store up to 3000 minutes worth of voicemail. -- And how long will that flash card last with the log and vmail churn? Flash devices have a limited number of writes you can do to a single cell before it wears out and cannot be written to any more. So does one have to throw out the whole appliance when one wears out the flash card? b. signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Digit Map
Jerry Jones wrote: Yours should work if you wait long enough for t to timeout. I think your digit map needs a T on the end of it if you want to allow timeouts for that match. ___ --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] app_echo.c
I would GUESS that if this line is removed, asterisk is settling on slin codec for the channel and does not try to negotiate anything better? Hence it will work without it. Mojo Bhrugu Mehta wrote: hi, all I have test echo application for just fun. I can'nt understand why this is used below in .c file, format = ast_best_codec(chan-nativeformats); ast_set_write_format(chan, format); ast_set_read_format(chan, format); without this this application work fine. then why this is used. any suggestion?? Bhrugu mehta ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Asterisk Appliance voicemail logs
Brian J. Murrell wrote: And how long will that flash card last with the log and vmail churn? Flash devices have a limited number of writes you can do to a single cell before it wears out and cannot be written to any more. So does one have to throw out the whole appliance when one wears out the flash card? Well it IS an appliance, after all. And that IS the American way! Throw it out when done with it. We do it with appliances, wives and our pets! John Novack -- Dog is my co-pilot ___ --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] Realtime sip.conf
I don't understand the USERS vs PEER vs FRIENDS. I just use Peer for everything. Has to do with can I only contact you or can you contact me too? ... Peer does it all. RealTime does have an issue. If you don't turn on caching, then it holds no state information. So if you think you're going to encouter firewall issues and need NAT=yes, then realtime will run in a static mode where you'll need to reload each time you change anything (like a password). I think the proper command is something like SIP PRUNE. Finally, putting something like sip.conf into realtime wasn't a move I wanted to make. I simply generate a SIP.conf file myself via my own program and run a SIP RELOAD (or simply reboot) each time I make a big change. Changes don't happen often so no biggie, where as I did want to make live changes to other SIP users without reloading (like a person using our web interface to change their own password). On 12/29/07, hugolivude [EMAIL PROTECTED] wrote: Hi - I'm looking into realtime and I'm having a bit of a problem with the SIP part. My review of the posts seems to indicate that I should use realtime static for the [general] part of my sip.conf including the registration commands: register=did:secret@domain/did context and use realtime realtime (funny name!) for peers and friends: [myprovider] type=peer auth=md5 username=... fromuser=... fromdomain=... secret=... host=... port=5060 nat=yes canreinvite=yes qualify=no disallow=all allow=ulaw dtmfmode=rfc2833 insecure=port,invite context=incoming-sip Is this correct? What's throwing me off is this statment found @ http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static: NOTE: You can only store a static config OR a RealTime config. You cannot, for example, store sip.conf and use sipfriends via RealTime. If I am correct, it would suggest that I'll have to do a reload when I add a DiD, but a reload won't be necessary if a new SIP client is added. Do I have it right? Also, what's the difference between a peer and a user? I used to think that a user was an agent authorized to call in to my * box, a peer was an agent I could reach and a freind was both. What's throwing me off now is the statement found @ http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peerview_comment_id=14966 : With newer versions of Asterisk the concept of SIP 'users' will be phased out. I can't understand this especially in the context of extconfig.conf that uses both a sipuser and sippeer entry. Could someone clarify for me? Thanks, H ___ --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 -- /Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Digit Map
That was one of the many iterations I tried already. It seems to respond in that it recognizes that I am dialing 01186106887, but then it only connects me to a dial tone and says Enter More Digits. There has to be something simple I am over looking here. I understand regular expressions, etc... I do have a tendancy to make a problem more complex than it really is though! This is my current digit map: 2XX|[2-9]11|0T|011|[0-1][2-9]x|[2-9]x|[2-9]x xxT Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, December 31, 2007 12:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom Digit Map On Dec 31, 2007, at 11:36 AM, Michael Munger wrote: I need the digit map to call China. Example number: 011-86-10-6887- 011-International (obvious) 86 is country code (China) 10 is city code (Beijing) Last 8 digits are the number. I tried using 011xxx.T but it always asks me to enter more digits. Tried some variations as well, but no joy. Yours should work if you wait long enough for t to timeout. How about 01186xx? Plus, IARC, when dialing offhook, pressing # should terminate dialing and send what it has at that point. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Digit Map
Michael Munger wrote: only connects me to a dial tone and says Enter More Digits. It actually says this? I would say then it's not the phone, but your phone system's programming. The Polycoms don't verbally say anything, at least not the ones I deal with. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Digit Map
Doug Lytle wrote: Michael Munger wrote: only connects me to a dial tone and says Enter More Digits. It actually says this? I would say then it's not the phone, but your phone system's programming. The Polycoms don't verbally say anything, at least not the ones I deal with. Doug No it doesn't SAY it -- the polycoms put on the screen Enter more digits. I think it's when what you've dialed doesn't match an entry in your digit map, or possibly when asterisk says that extension does not match anything So try: 011XXT in your digit map, meaning 011 plus at least six digits, consider it good because you can't know how long the string will be in advance. You want to allow for the smallest possible, which I suspect would be a three digit country code, like in Tonga (676) -- and you want to allow for the longest possible, to account for stuff like in Tajikistan: 992 37962 is BEFORE the local number, so you'd want 011+ at least 9 Xs following it 011XX -- Tricky! ** Moj ___ --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] Digium Asterisk Appliance voicemail logs
Gregory Malsack wrote: -- Digium -- That's news to me as well as the rest of the sales team. We were told that users cannot change the 1gb flash card. I just spoke with one of the Sales Engineers and he stated that it is apparently possible to change out the flash card on the AA50. However, it is not supported because the read/write speeds could be different on a new flash card, and thus not work. That is correct; we would not recommend using just *any* CF card, as the write speed of the card needs to be pretty high to be able support multiple voicemail messages being written simultaneously. With that said, though, it is possible to use a higher capacity CF card, but my previous response that said it was 'easy' was a bit of an overstatement :-) It can be done, and our support department does know how to get you the files you would need to populate the replacement card. -- Digium -- It is not possible to do so. However, the 1gb Flash card on the appliance will store up to 3000 minutes worth of voicemail. This is correct as well; the Linux kernel on the AA50 does not have NFS support nor SMB support, and there are no userspace tools present to handle NFS or SMB mounting of filesystems. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --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] Digium Asterisk Appliance voicemail logs
Brian J. Murrell wrote: And how long will that flash card last with the log and vmail churn? Flash devices have a limited number of writes you can do to a single cell before it wears out and cannot be written to any more. All modern flash cards (not flash chips, which are lower level) have built-in wear leveling. There is still an upper limit to what the card can handle, but keeping in mind the target market for this device (a small office with less than 50 users) it's not likely that the voicemail volume is going to be so extreme as to wear out the CF card. We do not ship the AA50 with Asterisk logs enabled to the CF card as far as I know, primarily for this reason. So does one have to throw out the whole appliance when one wears out the flash card? No. The files to repopulate the CF card are available to users who have active support subscriptions and they can replace the card. Users can also, of course, make a backup copy of the card on a new card when they receive the unit and have a ready-to-install replacement should any problems occur. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Digit Map
Mojo with Horan Company, LLC wrote: So try: 011XXT in your digit map, meaning 011 plus at least six digits, consider it good Err duh, that's ten X's not six :) To account for the Tajikistan example plus a little bit of local number. Really, it's dead simple to just do it like 011XT, which means 011 plus ANYTHING else plus a timeout :) Moj ___ --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] Digium Asterisk Appliance voicemail logs
Kevin P. Fleming wrote: the Linux kernel on the AA50 does not have NFS support nor SMB support, and there are no userspace tools present to handle NFS or SMB mounting of filesystems. FUSE? But it's probably not on the appliance. Regards, Philipp Kempgen ___ --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] Digium Asterisk Appliance voicemail logs
On Mon, 2007-12-31 at 14:16 -0600, Kevin P. Fleming wrote: No. The files to repopulate the CF card are available to users who have active support subscriptions and they can replace the card. Users can also, of course, make a backup copy of the card on a new card when they receive the unit and have a ready-to-install replacement should any problems occur. That's all fair enough then. I was just concerned with the message that was being sent along with the replacing the CF card is unsupported message. b. signature.asc Description: This is a digitally signed message part ___ --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] Problem with Polycom Soundpoint IP 320 Hardphone
Hey all, I've setup my asterisk install on a CentOS5 server, I've got a few IAX2 and SIP softphone clients connected on the same subnet and at least 1 external IAX2 softphone. However I'm having some difficulty getting the Polycom hardphone to function correctly. Watching the logs and debug trace it: - Registers correctly - Is able to make calls to other peers However it is not able to answer calls made to it. That is, the handset actually rings, but I've no way to answer it. The answer soft key, picking up the phone, etc. all have no effect. And I'm at a loss as to what setting should be altered to fix it. Any ideas? Possibly a tangent, but also affecting this handset, is that trying to dial out over an external SIP trunk fails on the first attempt. But calling an internal peer and then trying a second time makes it mysteriously work. Any help greatly appreciated, -- Glenn ___ --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] Problem with Polycom Soundpoint IP 320 Hardphone
On Mon, 2007-12-31 at 21:13 +, Glenn Gillen wrote: I'm having some difficulty getting the Polycom hardphone to function correctly. Watching the logs and debug trace it: - Registers correctly - Is able to make calls to other peers However it is not able to answer calls made to it. The first thing I'd do would be to capture a SIP trace of the call, using either sip set debug from the Asterisk CLI, or a packet sniffer such as tcpdump or Wireshark. I'd also turn up the core verbosity at the Asterisk CLI and look for clues that might be present there while the call is being made. --- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Digit Map
At 14:27 12/31/2007, Mojo with Horan Company, LLC wrote: Mojo with Horan Company, LLC wrote: So try: 011XXT in your digit map, meaning 011 plus at least six digits, consider it good Err duh, that's ten X's not six :) To account for the Tajikistan example plus a little bit of local number. Really, it's dead simple to just do it like 011XT, which means 011 plus ANYTHING else plus a timeout :) Moj I think you might need a dot . in there to accept any length: dialplan.1.digitmap=*xxx|*|[2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT|xxxT dialplan.1.digitmap.timeOut=3 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1
On Sunday 30 December 2007 14:40:40 Mindaugas Kezys wrote: Thank you! Will it come to 1.4.16.3 or 1.4.17? Yes, it will. -- Tilghman ___ --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] One Way Delay in Audio Over Analog
I have been trying to track down the cause/fix for a problem and I am out of ideas... I am hoping one of you can point me in the right direction. The symptom is that when a calls is placed from an internal extension through an analog line to a number on the pstn the caller can hear the callee but the callee can not hear the caller for as long as ten seconds. The problem appears to happen fairly consistently on the same pstn numbers. However, I have not seen a common characteristic in those numbers. For example, one of them is a direct number to a cell phone and another is to a Verizon fiber-optic phone/data service. The problem does not seem to be related to the type of SIP phone being used by the caller - for example, we have tried both X-Lite and Polycom phones without a change in behavior. The problem does not appear to occur if the callee then calls into our system (at least the one time I was able to have this happen). Turning on or off echo cancellation and/or call progress does not seem to change the behavior. I will appreciate any ideas you have. I am certainly stumped. Thanks and Happy New Year! -Brian ___ --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] Problem with Polycom Soundpoint IP 320 Hardphone
glenn, check your handset cord... it might be plugged into the wrong port in the back of the phone. perhaps the headset jack... daveC Glenn Gillen wrote: Hey all, I've setup my asterisk install on a CentOS5 server, I've got a few IAX2 and SIP softphone clients connected on the same subnet and at least 1 external IAX2 softphone. However I'm having some difficulty getting the Polycom hardphone to function correctly. Watching the logs and debug trace it: - Registers correctly - Is able to make calls to other peers However it is not able to answer calls made to it. That is, the handset actually rings, but I've no way to answer it. The answer soft key, picking up the phone, etc. all have no effect. And I'm at a loss as to what setting should be altered to fix it. Any ideas? Possibly a tangent, but also affecting this handset, is that trying to dial out over an external SIP trunk fails on the first attempt. But calling an internal peer and then trying a second time makes it mysteriously work. Any help greatly appreciated, -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Digit Map
Doug wrote: At 14:27 12/31/2007, Mojo with Horan Company, LLC wrote: Mojo with Horan Company, LLC wrote: So try: 011XXT in your digit map, meaning 011 plus at least six digits, consider it good Err duh, that's ten X's not six :) To account for the Tajikistan example plus a little bit of local number. Really, it's dead simple to just do it like 011XT, which means 011 plus ANYTHING else plus a timeout :) Moj I think you might need a dot . in there to accept any length: dialplan.1.digitmap=*xxx|*|[2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT|xxxT dialplan.1.digitmap.timeOut=3 Oooh, too true. Thanks for remembering! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
On Dec 28, 2007 8:28 PM, Al lists [EMAIL PROTECTED] wrote: what method is preferred: haylafax and Iaxmodem or spnadsp for faxing. What are you trying to do and do you have a T1 or ISDN line? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
at this time is terminating a SIP trunk, each DID will get its own fax box. I guess at this time i'm looking to find a tutorial for installing iaxmodem and hylafax as it seems to be the answer. On Dec 31, 2007 9:11 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote: On Dec 28, 2007 8:28 PM, Al lists [EMAIL PROTECTED] wrote: what method is preferred: haylafax and Iaxmodem or spnadsp for faxing. What are you trying to do and do you have a T1 or ISDN line? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
Unless your provider provides a T.38 gateway, fax over SIP is pretty much guaranteed to be unusable. Often you can get away with it over a LAN using G711a or G711u, but any of the lower bandwidth codecs /won't/ be able to properly handle fax calls. Whilst I haven't used it myself, I believe IAXmodem and Hylafax are used for sending and receiving faxes from a local PSTN termination point such as T1 or ISDN. The IAXmodem web site explains the pitfalls of faxing over the internet. See http://iaxmodem.sourceforge.net/faq.php for more info. Last time I heard IAXModem didn't support T.38 because the IAX2 protocol didn't support T.38 - whether that's still the case or not, I don't know. Al lists wrote: at this time is terminating a SIP trunk, each DID will get its own fax box. I guess at this time i'm looking to find a tutorial for installing iaxmodem and hylafax as it seems to be the answer. On Dec 31, 2007 9:11 PM, Andrew Joakimsen [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Dec 28, 2007 8:28 PM, Al lists [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: what method is preferred: haylafax and Iaxmodem or spnadsp for faxing. What are you trying to do and do you have a T1 or ISDN line? ___ --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 http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
If by fax box you mean an ATA with a fax machine attached them Asterisk 1.4 with T38 passthrough should work if the SIP provider has T.38 capabilites. If by fax box you mean a 'faxmail inbox' then no Asterisk cannot help you terminate that from SIP. Get a Cisco gateway, make sure your provider uses T.38 and connect that to your Asterisk via T1 or E1. On Jan 1, 2008 12:50 AM, Al lists [EMAIL PROTECTED] wrote: at this time is terminating a SIP trunk, each DID will get its own fax box. I guess at this time i'm looking to find a tutorial for installing iaxmodem and hylafax as it seems to be the answer. On Dec 31, 2007 9:11 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote: On Dec 28, 2007 8:28 PM, Al lists [EMAIL PROTECTED] wrote: what method is preferred: haylafax and Iaxmodem or spnadsp for faxing. What are you trying to do and do you have a T1 or ISDN line? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
Rob Hillis wrote: Last time I heard IAXModem didn't support T.38 because the IAX2 protocol didn't support T.38 - whether that's still the case or not, I don't know. There are actually two reasons. One is that T.38 over IAX is not defined. The other is the current T.38 termination support in spandsp is only for the full FAX machine it contains. T.38 termination to the class 1 FAX modem (T.31) interface for HylaFAX is a work in progress. When that is done, I hope we will have a sipmodem to replace iaxmodem, offering bother audio and T.38 to HylaFAX functionality. Steve ___ --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] One Way Delay in Audio Over Analog
Brian Alexander wrote: I have been trying to track down the cause/fix for a problem and I am out of ideas... I am hoping one of you can point me in the right direction. The symptom is that when a calls is placed from an internal extension through an analog line to a number on the pstn the caller can hear the callee but the callee can not hear the caller for as long as ten seconds. The problem appears to happen fairly consistently on the same pstn numbers. However, I have not seen a common characteristic in those numbers. For example, one of them is a direct number to a cell phone and another is to a Verizon fiber-optic phone/data service. The problem does not seem to be related to the type of SIP phone being used by the caller - for example, we have tried both X-Lite and Polycom phones without a change in behavior. The problem does not appear to occur if the callee then calls into our system (at least the one time I was able to have this happen). Turning on or off echo cancellation and/or call progress does not seem to change the behavior. I will appreciate any ideas you have. I am certainly stumped. Thanks and Happy New Year! -Brian Brian, What about some facts ? Hardware ? Software versions ? /Mats ___ --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
