Re: [asterisk-users] fax / t38 gateway
I think Brendan is asking about endpoints (how to connect fax machines to pure VoIP). Short answer: - you could connect standalone T.38-enabled analog gateways to 1.4, - with 1.6, you can also use an analog board inside a server and connect fax machines to this board. ___ -- 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] command - set sip_codec- does not work with asterisk-1.4.21
hello:
i want to test the g729 with asterisk. my scenario is sipp(ulaw)-asterisk1
with g729-asterisk2 with g729.
I want to test g729 module with asterisk-1.4.21, when i make calls from
asterisk 1 to asterisk 2, the asterisk 1 always send ulaw to asterisk 2. my sip
in asterisk 1 is with codec g729 and enforce that use g729, the sip in asterisk
2 also work with G729 only, but asterisk 2 reports the condec compatibility
problem. both of asterisks can show g729 are there.
===
exten = 2005,1,Answer
exten = 2005,2,Set(${SIP_CODEC}=g729) // does not work
exten = 2005,3,DIAL(SIP/[EMAIL PROTECTED],30,r)
exten = 2005,4,Hangup
===use sipp t call asterisk 1 then forward to asterisk 2 with sip 1000.
--sip.conf
[1000]
username=1000
allowtransfer=yes
type=friend
secret=1000
qualify=yes
canreinvite=yes
host=dynamic
insecure=very
fromuser=1000
;dtmfmode=rfc2833
disallow=all
allow=g729
;allow=ulaw
;allow=alaw
context=internal
when the calls coming from asterisk 1, i can not see any g729 inforamtion, the
Noop(${SIP_CODEC}) show nothing.
some people reported the bug for sip_codec, i followed that, but i still can
not. sovle. anyone knows how to set sip_codec in dialplan or have any right
format for set sip_codec in dialplan?
thanks!
zhu
-
雅虎邮箱，您的终生邮箱！___
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Re: [asterisk-users] Sonicwall potentially causing long ping times toSIP phones
Craig Van Ham wrote: I had weird issues when using a Sonicwall, gave up. Same here, avoid them! I use the SnapGear SG560 now. - Mike ___ -- 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 video
On Thu, 23 Oct 2008, Nhadie wrote: hi, hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. What's the other end? Grandstreams won't take H263p, so force eyebeam/x-lite to only use H263, and just put this in as the codec in sip.conf. I presume you've read this too: http://www.voip-info.org/wiki-Asterisk+video i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start troubleshooting this? I'm using Video with asterisk version 1.2 and it's working very well so-far. Using Grandsteam phones - done one test with an ATL phone and made a few calls to someone using x-lite. Tring to get Ekiga to work, but I can't get the codecs for my wifes Acer One notebook. Biggest problem is codec selection. Make sure both ends support the same thing, and that asterisk can route it. Gordon ___ -- 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 video
Gordon Henderson wrote: On Thu, 23 Oct 2008, Nhadie wrote: hi, hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. What's the other end? Grandstreams won't take H263p, so force eyebeam/x-lite to only use H263, and just put this in as the codec in sip.conf. hi sir both sides using eyebeam, i also hardset codec to use basic H236. then i allowed the codec, allow=h236. i'm using Asterisk 1.4.21.2., could that be the reason? coz document says video on 1.4 is on infancy. I presume you've read this too: http://www.voip-info.org/wiki-Asterisk+video i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start troubleshooting this? I'm using Video with asterisk version 1.2 and it's working very well so-far. Using Grandsteam phones - done one test with an ATL phone and made a few calls to someone using x-lite. Tring to get Ekiga to work, but I can't get the codecs for my wifes Acer One notebook. Biggest problem is codec selection. Make sure both ends support the same thing, and that asterisk can route it. Gordon ___ -- 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] problems with some incoming/outgoing calls
Hi, I've been very puzzled lately. I installed a phone system for a friend a few weeks ago, and they're having a problem that I can't get rid of, actually 2 problems. Before I go into the problems, let me tell you about the setup. It's a pretty small setup with only 4 handsets, all Polycom 320s, the server is a Dell SC440 with Intel E2180 CPU (dual core, 2GHz) and 512MB Ram. Internet Connection is an ADSL2, with a not so reliable ISP in australia. For incoming calls, I had a Digium TDM410P with 4xFXO modules and HWEC. Because of these problems, i replaced the Digium card with a Sangoma A200D, but it didn't make any difference to the problems. All phones are hooked up to a Netgear PoE switch. Almost forgot to mention that this is not my first Asterisk setup, and in fact it is my 4th, and I used various SIP handsets before, and also different cards (Analog and Digital), so I'm not a total noob. Let's get to the problems... 1) Some incoming calls cannot be picked up Sometimes, incoming calls, coming through the analog card, cannot be picked up. All handsets are set to ring at the same time on incoming calls. and most of the time, calls can be answered on any of the handsets, but maybe 3 or 4 times a day, all handsets will be ringing, and you go to one handset to answer the call, you pick the handset, and it doesn't answer the call, it keeps ringing, then you go to another handset, and still can't pick up, sometimes, you can even try all 4 handsets, and no luck. but, at other times, you can't answer on the first handset, but you can on another, and it is totally random. but people are pretty pissed off for running around to answer a call. and what puzzles me is that you can sit around watching logs for hours, and it won't happen, other times, it happens 3 times in a row. any ideas? 2) Delay on outgoing calls via SIP People have been saying that when they call people, there's a delay for the call to be answered. For example, caller dials a number, callee answers the ringing phone, but caller is still listening to a ringing tone, and after a few seconds (up to 15 seconds) it sounds like the callee has just answered the call, when in fact, he had already answered a few seconds before. Problem with this is that some callees will hangup before the caller starts talking. These calls are going via pennytel, in australia, which seems to be a pretty good VOIP provider around here, and I've been using it on other setups and never had these issues. Well, sorry for the long first post, but I would really appreciate any suggestions you have. Cheers, Fernando ___ -- 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] Astribank loop current adjustment
For a door opener on an Astribank FXS port we need a loop current of 24.5mA . It does not function with the Astribank now, the dialtone becomes quiet immediately after pressing the button on that device. I've seen a limit of 23mA in the zaptel source. Is it possible to change the loop current of the Astribank somehow? Udo ___ -- 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 video
On Thu, 23 Oct 2008, Nhadie wrote: Gordon Henderson wrote: On Thu, 23 Oct 2008, Nhadie wrote: hi, hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. What's the other end? Grandstreams won't take H263p, so force eyebeam/x-lite to only use H263, and just put this in as the codec in sip.conf. hi sir both sides using eyebeam, i also hardset codec to use basic H236. then i allowed the codec, allow=h236. i'm using Asterisk 1.4.21.2., could that be the reason? coz document says video on 1.4 is on infancy. I don't know about 1.4 - Not tried it yet. But that's one more reason to not upgrade to 1.4 yet. How could they have broken video when it was working in 1.2, or am I missing something obvious? Maybe I'll jump directly to 1.6 when it's released stable. Gordon ___ -- 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 video
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Sent: Thursday, October 23, 2008 6:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk video Gordon Henderson wrote: On Thu, 23 Oct 2008, Nhadie wrote: hi, hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeB eam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. What's the other end? Grandstreams won't take H263p, so force eyebeam/x-lite to only use H263, and just put this in as the codec in sip.conf. hi sir both sides using eyebeam, i also hardset codec to use basic H236. then i allowed the codec, allow=h236. i'm using Asterisk 1.4.21.2., could that be the reason? coz document says video on 1.4 is on infancy. Make sure you have h263 and not h236 ... I have the same configuration as yours and it work ok If you still have problems post your relevant sip configuration I presume you've read this too: http://www.voip-info.org/wiki-Asterisk+video i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start troubleshooting this? I'm using Video with asterisk version 1.2 and it's working very well so-far. Using Grandsteam phones - done one test with an ATL phone and made a few calls to someone using x-lite. Tring to get Ekiga to work, but I can't get the codecs for my wifes Acer One notebook. Biggest problem is codec selection. Make sure both ends support the same thing, and that asterisk can route it. Gordon ___ -- 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 video
Hi sir, sorry typo on the e-mail, this is what i have: sip.conf [100666] type=friend secret=666 allow=h263 allow=all dtmfmode=rfc2833 canreinvite=no host=dynamic context=testvideo nat=yes [100777] type=friend secret=777 allow=h263 allow=all dtmfmode=rfc2833 canreinvite=no host=dynamic context=testvideo nat=yes extensions.conf [testvideo] exten = 666,1,Dial(SIP/100666|30|t) exten = 666,n,Hangup exten = 777,1,Dial(SIP/100777|30|t) exten = 777,n,Hangup Robert Augustyn wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Sent: Thursday, October 23, 2008 6:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk video Gordon Henderson wrote: On Thu, 23 Oct 2008, Nhadie wrote: hi, hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeB eam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. What's the other end? Grandstreams won't take H263p, so force eyebeam/x-lite to only use H263, and just put this in as the codec in sip.conf. hi sir both sides using eyebeam, i also hardset codec to use basic H236. then i allowed the codec, allow=h236. i'm using Asterisk 1.4.21.2., could that be the reason? coz document says video on 1.4 is on infancy. Make sure you have h263 and not h236 ... I have the same configuration as yours and it work ok If you still have problems post your relevant sip configuration I presume you've read this too: http://www.voip-info.org/wiki-Asterisk+video i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start troubleshooting this? I'm using Video with asterisk version 1.2 and it's working very well so-far. Using Grandsteam phones - done one test with an ATL phone and made a few calls to someone using x-lite. Tring to get Ekiga to work, but I can't get the codecs for my wifes Acer One notebook. Biggest problem is codec selection. Make sure both ends support the same thing, and that asterisk can route it. Gordon ___ -- 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] switching from 1.6.0-beta9 to 1.6.0.1 problems
Hello everyone! I've just switched from Asterisk 1.6.0-beta9 to 1.6.0.1 and my mISDN is not working. Here's what happens, if I try to call the line: bach P[ 1] -- !! lib: No free channel! P[ 1] -- we have already send Release_complete I haven't changed the configuration fles. Should I change something there? If you need more info, just tell me and I'll provide it, if I can. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] problems with some incoming/outgoing calls
Hi, which version of asterisk are you running? Perhaps if you post your extensions.conf and others related files you could get more accurate help. If you answer a ringing phone and you can't answer the call, there you could have a network or sip config problem, that means that the SIP packet is not returning to the pbx. Regards. Lucas On Thu, 23 Oct 2008 10:30:29 -0200, Fernando Serto [EMAIL PROTECTED] wrote: Hi, I've been very puzzled lately. I installed a phone system for a friend a few weeks ago, and they're having a problem that I can't get rid of, actually 2 problems. Before I go into the problems, let me tell you about the setup. It's a pretty small setup with only 4 handsets, all Polycom 320s, the server is a Dell SC440 with Intel E2180 CPU (dual core, 2GHz) and 512MB Ram. Internet Connection is an ADSL2, with a not so reliable ISP in australia. For incoming calls, I had a Digium TDM410P with 4xFXO modules and HWEC. Because of these problems, i replaced the Digium card with a Sangoma A200D, but it didn't make any difference to the problems. All phones are hooked up to a Netgear PoE switch. Almost forgot to mention that this is not my first Asterisk setup, and in fact it is my 4th, and I used various SIP handsets before, and also different cards (Analog and Digital), so I'm not a total noob. Let's get to the problems... 1) Some incoming calls cannot be picked up Sometimes, incoming calls, coming through the analog card, cannot be picked up. All handsets are set to ring at the same time on incoming calls. and most of the time, calls can be answered on any of the handsets, but maybe 3 or 4 times a day, all handsets will be ringing, and you go to one handset to answer the call, you pick the handset, and it doesn't answer the call, it keeps ringing, then you go to another handset, and still can't pick up, sometimes, you can even try all 4 handsets, and no luck. but, at other times, you can't answer on the first handset, but you can on another, and it is totally random. but people are pretty pissed off for running around to answer a call. and what puzzles me is that you can sit around watching logs for hours, and it won't happen, other times, it happens 3 times in a row. any ideas? 2) Delay on outgoing calls via SIP People have been saying that when they call people, there's a delay for the call to be answered. For example, caller dials a number, callee answers the ringing phone, but caller is still listening to a ringing tone, and after a few seconds (up to 15 seconds) it sounds like the callee has just answered the call, when in fact, he had already answered a few seconds before. Problem with this is that some callees will hangup before the caller starts talking. These calls are going via pennytel, in australia, which seems to be a pretty good VOIP provider around here, and I've been using it on other setups and never had these issues. Well, sorry for the long first post, but I would really appreciate any suggestions you have. Cheers, Fernando ___ -- 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] users.conf and sip call-limit
Does the call-limit directive work on those SIP items defined in users.conf as it relates to presence and queues? Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: [EMAIL PROTECTED] This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- 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] Atxfer Command
Hi, We are testing new Asterisk 1.6.0.1 because we would like to use the Attended Transfer feature and we are trying to use the new action Atxfer developed for AMI. As far as we know, it is suposed to be in this release as it can be read in Digium's changelog /New command: Atxfer. See doc/manager_1_1.txt for more details or manager show command Atxfer from the CLI/ But, when we try to see information in the CLI console, this command doesn't exist, neither AtxferAction works, we got a message saying that this command is unknown Are we missing something? Thanks in advance David ___ -- 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] Atxfer Command
David Monteagudo Sanz wrote: Hi, We are testing new Asterisk 1.6.0.1 because we would like to use the Attended Transfer feature and we are trying to use the new action Atxfer developed for AMI. As far as we know, it is suposed to be in this release as it can be read in Digium's changelog /New command: Atxfer. See doc/manager_1_1.txt for more details or manager show command Atxfer from the CLI/ But, when we try to see information in the CLI console, this command doesn't exist, neither AtxferAction works, we got a message saying that this command is unknown Are we missing something? Thanks in advance David The AMI atxfer command is not in 1.6.0 and will be in no releases of 1.6.0. It is in 1.6.1, for which a beta is currently available. Mark Michelson ___ -- 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] Devstate and Voicemail
Have an interesting problem, Using asterisk 1.6.0.1 Phone A receives voicemail, dials into VoiceMailMain, Phone B's BLF for A lights up. Phone A deletes the voicemail but still in VoiceMailMain, Phone B's BLF for A goes off. Phone A hang's up, Phone B's BLF for A goes on. From this point forward the Phone B's BLF for A seems to always show the opposite of what it should. I've looked at 'core show hints' and it is in fact reporting INUSE when it's not, and NOT_INUSE when it is. Is this a bug or just some configuration option that I am missing? Is there a way to manually change the devicestate for a channel? Marc Hudson ___ -- 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] Hylafax asterisk iaxmodem problem
Hello all, I have an asterisk box running in a customer with Hylafax, iaxmodem, asterisk 1.2.18. The service can receive faxes, from a lot of fax machines, but there are a couple of them that asterisk Hylafax cannot complete. This calls arrive the asterisk box, asterisk detect that this calls are fax, asterisk answer the call, and then Hangup the call. But hylafax do not receive nothing. When I run zap show channel 1, on the asterisk CLI. The outpuit shows, File Descriptor: 20 Span: 2 Extension: Dialing: no Context: from-zaptel Caller ID: Calling TON: 0 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Offhook Why some faxes do not get received? What could be wrong? Any clue wil be welcomed. Thanks in advanced. VoipCrazy. ___ -- 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] Channels are increasing without limit - Please Help!
Suddenly my system crash whem I see core show channels are increasing until reaches its limit at asterisk.conf It seems channels (Local, Zap, SIP) are not being closed. The problem persists and I don't know what to do Please help me! ___ -- 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] adding a second extension
And this phone are connected in a local LAN?? Because I see Asterisk receiving a Bad request from 68.156.63.118 If those phones are not in your local LAN, try with a soft phone first. Could be Zoiper or Xlite. Besides, use SIP SET DEBUG, for SIP debugging and try to see why is 101 sending a 400 Bad request back to Asterisk. On Wed, Oct 22, 2008 at 9:10 PM, Stephen Reese [EMAIL PROTECTED] wrote: On Wed, Oct 22, 2008 at 8:15 PM, Juan Rodríguez [EMAIL PROTECTED] wrote: What kind of phone are you trying to connect to 101??? and from where? Both phones are Cisco, 101 is a 7960 and 102 is a 7912. 101 can contact 102 by dialing 101 but not the other way around, I just get a busy tone. -- Juan E. Rodríguez Cel. 829-886-5565 Work: 809-724-9227 ___ -- 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] Sonicwall potentially causing long ping times to SIP phones
Sorry for asking the obvious question, but are there other elements of the slow path besides the Sonicwall? I mean, what is in front of the Sonicwall? Also, might the Sonicwall be positioned as some kind of choke point in the topology, thus leading to genuine sporadic congestion? James Lamanna wrote: Date: Wed, 22 Oct 2008 11:35:12 -0700 From: James Lamanna [EMAIL PROTECTED] Subject: [asterisk-users] Sonicwall potentially causing long ping times toSIP phones Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on sip show peer shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify? Thanks. -- James smime.p7s Description: S/MIME Cryptographic Signature ___ -- 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] Hylafax asterisk iaxmodem problem
voip crazy wrote: This calls arrive the asterisk box, asterisk detect that this calls are fax, asterisk answer the call, and then Hangup the call. But hylafax do not receive nothing. What does the CLI say about this call? Thanks, Lee. ___ -- 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] Channels are increasing without limit - Please Help!
My version is 1.4.21.1 On Thu, Oct 23, 2008 at 11:38 AM, Daniel - Asterisk [EMAIL PROTECTED]wrote: Suddenly my system crash whem I see core show channels are increasing until reaches its limit at asterisk.conf It seems channels (Local, Zap, SIP) are not being closed. The problem persists and I don't know what to do Please help me! ___ -- 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] Sonicwall potentially causing long ping times to SIP phones
We've had LOTS of problems with Sonicwalls doing bad things to SIP and RTP connections. I've seen the delay thing, as well as the Sonicwall throwing away entries from the ARP table because of inactivity. I've also seen sporadic, intermittent problems with transfer from one phone to another. I have no doubt that a new, properly configured Sonicwall can be made to function properly in a VoIP environment, but we are not Sonicwall experts, nor are many of the purported experts. In every case where we've had problems with VoIP behind a Sonicwall, the problems ALL disappear when we put the phones on a LAN segment that does not pass through the Sonicwall. So, now that's our going in position. If it works, great, but if it doesn't, our solution is to take the Sonicwall out of the picture. My $.02 . Bruce Komito WPTI Telecom (775) 236-5815 On Thu, 23 Oct 2008, Bill Michaelson wrote: Sorry for asking the obvious question, but are there other elements of the slow path besides the Sonicwall? I mean, what is in front of the Sonicwall? Also, might the Sonicwall be positioned as some kind of choke point in the topology, thus leading to genuine sporadic congestion? James Lamanna wrote: Date: Wed, 22 Oct 2008 11:35:12 -0700 From: James Lamanna [EMAIL PROTECTED] Subject: [asterisk-users] Sonicwall potentially causing long ping times toSIP phones Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on sip show peer shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify? Thanks. -- James ___ -- 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] Hylafax asterisk iaxmodem problem
Lee Howard wrote: voip crazy wrote: This calls arrive the asterisk box, asterisk detect that this calls are fax, asterisk answer the call, and then Hangup the call. But hylafax do not receive nothing. What does the CLI say about this call? With high verbosity, I might add. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Sonicwall potentially causing long ping times to SIP phones
Bill Michaelson wrote: Sorry for asking the obvious question, but are there other elements of the slow path besides the Sonicwall? I mean, what is in front of the Sonicwall? Also, might the Sonicwall be positioned as some kind of choke point in the topology, thus leading to genuine sporadic congestion? The device in front of the SonicWall is a Cisco Router. Ping times to the ethernet interface of the router are good (~10ms). Also, having a user behind the SonicWall ping the PBX results in an average 20-30ms ping time. So it seems as though the lag is specific to SIP signaling (specifically the OPTIONS requests that asterisk qualify sends out). Unfortunately I can't really ask the client to dump their SonicWall (which we do not manage). On the SonicWall, I know it is configured for Consistent NAT and SIP Transformations are disabled. -- James On Wed, Oct 22, 2008 at 11:35 AM, James Lamanna [EMAIL PROTECTED] wrote: Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on sip show peer shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify? Thanks. -- James ___ -- 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] Channels are increasing without limit - Please Help!
I'm restarting my system without solution and I've extended my call limit to 10 calls (asterisk.conf) to avoid call rectriction. But, why now? It was working well from July until this morning. Thanks in advance for every help you can give. Daniel On Thu, Oct 23, 2008 at 12:04 PM, Daniel - Asterisk [EMAIL PROTECTED]wrote: My version is 1.4.21.1 On Thu, Oct 23, 2008 at 11:38 AM, Daniel - Asterisk [EMAIL PROTECTED]wrote: Suddenly my system crash whem I see core show channels are increasing until reaches its limit at asterisk.conf It seems channels (Local, Zap, SIP) are not being closed. The problem persists and I don't know what to do Please help me! ___ -- 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] Sonicwall potentially causing long ping times to SIP phones
Can you get another public IP? If so put another router in. Use vlans to seperate the traffic. Sent from my iPhone On 23-Oct-08, at 11:28 AM, James Lamanna [EMAIL PROTECTED] wrote: Bill Michaelson wrote: Sorry for asking the obvious question, but are there other elements of the slow path besides the Sonicwall? I mean, what is in front of the Sonicwall? Also, might the Sonicwall be positioned as some kind of choke point in the topology, thus leading to genuine sporadic congestion? The device in front of the SonicWall is a Cisco Router. Ping times to the ethernet interface of the router are good (~10ms). Also, having a user behind the SonicWall ping the PBX results in an average 20-30ms ping time. So it seems as though the lag is specific to SIP signaling (specifically the OPTIONS requests that asterisk qualify sends out). Unfortunately I can't really ask the client to dump their SonicWall (which we do not manage). On the SonicWall, I know it is configured for Consistent NAT and SIP Transformations are disabled. -- James On Wed, Oct 22, 2008 at 11:35 AM, James Lamanna [EMAIL PROTECTED] wrote: Hi, I'm having an issue where some phones behind a sonicwall are auto- congesting. The status on sip show peer shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify? Thanks. -- James ___ -- 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] adding a second extension
On Thu, Oct 23, 2008 at 12:39 PM, Juan Rodríguez [EMAIL PROTECTED] wrote: And this phone are connected in a local LAN?? Because I see Asterisk receiving a Bad request from 68.156.63.118 If those phones are not in your local LAN, try with a soft phone first. Could be Zoiper or Xlite. Besides, use SIP SET DEBUG, for SIP debugging and try to see why is 101 sending a 400 Bad request back to Asterisk. Both of these phones are on my local lan but the Asterisk server is at a colo facility on the internet outside of the local lan. The local lan does use NAT/PAT. I see an error Warning: 399 Bad Request - 'Malformed/Missing FROM: field'. Is this a problem? Thanks --- ns1*CLI --- SIP read from 68.156.63.118:1082 --- INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14 From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:1083;user=phone;transport=udp User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Expires: 300 Content-Length: 274 Content-Type: application/sdp v=0 o=102 157742 157742 IN IP4 172.16.2.18 s=Cisco 7912 SIP Call c=IN IP4 68.156.63.118 t=0 0 m=audio 16384 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 - --- (14 headers 12 lines) --- Sending to 68.156.63.118 : 1083 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] --- Reliably Transmitting (NAT) to 68.156.63.118:1082 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14;received=68.156.63.118 From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914 To: sip:[EMAIL PROTECTED];user=phone;tag=as355e0f84 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=ns1.neocipher.net, nonce=7c2e1ba9 Content-Length: 0 Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) Found user '102' --- SIP read from 64.2.142.116:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060 From: sip:[EMAIL PROTECTED];tag=as401a34d4 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 3064 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 - --- (10 headers 0 lines) --- --- SIP read from 64.2.142.116:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060 From: sip:[EMAIL PROTECTED];tag=as401a34d4 To: sip:[EMAIL PROTECTED];tag=as7a2f92a1 Call-ID: [EMAIL PROTECTED] CSeq: 3064 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=575628ec Content-Length: 0 - --- (10 headers 0 lines) --- Responding to challenge, registration to domain/host name inbound18.vitelity.net REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 64.2.142.116:5060: REGISTER sip:inbound18.vitelity.net SIP/2.0 Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK6245e988;rport From: sip:[EMAIL PROTECTED];tag=as751cb0af To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 3065 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username=rsreese, realm=asterisk, algorithm=MD5, uri=sip:inbound18.vitelity.net, nonce=575628ec, response=b765dbdebba8af18b19707efe651d65d Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- --- SIP read from 68.156.63.118:1082 --- ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14 From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914 To: sip:[EMAIL PROTECTED];user=phone;tag=as355e0f84 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Content-Length: 0 - --- (9 headers 0 lines) --- ns1*CLI --- SIP read from 68.156.63.118:1082 --- INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bKd82d2d0850902de2 From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:1083;user=phone;transport=udp User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Proxy-Authorization: Digest
Re: [asterisk-users] Sonicwall potentially causing long ping times to SIP phones
On 10/23/08, Bruce Komito [EMAIL PROTECTED] wrote: We've had LOTS of problems with Sonicwalls doing bad things to SIP and RTP connections. I've seen the delay thing, as well as the Sonicwall throwing away entries from the ARP table because of inactivity. I've also seen sporadic, intermittent problems with transfer from one phone to another. I have no doubt that a new, properly configured Sonicwall can be made to function properly in a VoIP environment, but we are not Sonicwall experts, nor are many of the purported experts. In every case where we've had problems with VoIP behind a Sonicwall, the problems ALL disappear when we put the phones on a LAN segment that does not pass through the Sonicwall. So, now that's our going in position. If it works, great, but if it doesn't, our solution is to take the Sonicwall out of the picture. My $.02 . Bruce Komito WPTI Telecom (775) 236-5815 I wouldn't single out SonicWalls when it comes to breaking SIP traffic. Most of the anything but simple PAT devices I've seen that implement any SIP specific fixups usually end up breaking something along the line. Unless the product is from a company where SIP is their core competency (like Ingate, or /maybe/ Cisco) it's best to stay away and/or disable the SIP specific fixups wherever possible. I'm looking forward to the day when SIP-TLS is the norm and these devices have no idea what kind of traffic is flowing through them! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- 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] fax / t38 gateway
Indeed I am going for pure voip and trying to figure out how to implement t.38, as you suggest. On Oct 23, 2008, at 2:08 AM, Olivier wrote: I think Brendan is asking about endpoints (how to connect fax machines to pure VoIP). Short answer: - you could connect standalone T.38-enabled analog gateways to 1.4, Like what? I'm not familiar with this tech, I googled around a bit but didn't come up with much. I think I just don't know the lingo yet. : ( Could you point out one of these? - with 1.6, you can also use an analog board inside a server and connect fax machines to this board. So basically what you're saying is that to do this (convert the analog to t.38) myself I would still need to have analog coming into my asterisk server (which makes sense, but doesn't help me avoid paying for normal phone lines)... Sounds to me like in this situation t.38 would be purely for getting faxes around on my own asterisk(s) if that became necessary. Which leads me to my other question again, is there some sort of internet service that will do the analog to t.38 conversion for me and then pass the t.38 on to my asterisk server? ___ -- 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 Andrew you mentioned something about sip providers that support t.38? When you say support, do you mean that they have passthrough turned on, or they will actually do an analog t.30 to t.38 conversion for you? That may be what I'm after... If you, or anyone else, know of a provider that does this could you point me in the right direction? Thank you all for your thoughts. Brendan Martens ___ -- 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] Sonicwall potentially causing long ping times to SIP phones
You're absolutely right. I only mention Sonicwall, because those are the ones we see most often and there is a perception out there that, because Sonicwall is the (disputed) leading firewall, it should work. Bruce Komito WPTI Telecom (775) 236-5815 On Thu, 23 Oct 2008, Kristian Kielhofner wrote: On 10/23/08, Bruce Komito [EMAIL PROTECTED] wrote: We've had LOTS of problems with Sonicwalls doing bad things to SIP and RTP connections. I've seen the delay thing, as well as the Sonicwall throwing away entries from the ARP table because of inactivity. I've also seen sporadic, intermittent problems with transfer from one phone to another. I have no doubt that a new, properly configured Sonicwall can be made to function properly in a VoIP environment, but we are not Sonicwall experts, nor are many of the purported experts. In every case where we've had problems with VoIP behind a Sonicwall, the problems ALL disappear when we put the phones on a LAN segment that does not pass through the Sonicwall. So, now that's our going in position. If it works, great, but if it doesn't, our solution is to take the Sonicwall out of the picture. My $.02 . Bruce Komito WPTI Telecom (775) 236-5815 I wouldn't single out SonicWalls when it comes to breaking SIP traffic. Most of the anything but simple PAT devices I've seen that implement any SIP specific fixups usually end up breaking something along the line. Unless the product is from a company where SIP is their core competency (like Ingate, or /maybe/ Cisco) it's best to stay away and/or disable the SIP specific fixups wherever possible. I'm looking forward to the day when SIP-TLS is the norm and these devices have no idea what kind of traffic is flowing through them! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- 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] Devstate and Voicemail
On Thu, 2008-10-23 at 09:36 -0700, Marc Hudson wrote: I've looked at 'core show hints' and it is in fact reporting INUSE when it's not, and NOT_INUSE when it is. That definitely sounds like a bug to me. Could you please report this on the bug tracker, so that the developers can take a look and try to reproduce and solve the problem? -- Jared Smith Training 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] Devstate and Voicemail
Jared Smith schrieb: On Thu, 2008-10-23 at 09:36 -0700, Marc Hudson wrote: I've looked at 'core show hints' and it is in fact reporting INUSE when it's not, and NOT_INUSE when it is. That definitely sounds like a bug to me. Could you please report this on the bug tracker, so that the developers can take a look and try to reproduce and solve the problem? Sounds a bit like http://bugs.digium.com/view.php?id=13668 or http://bugs.digium.com/view.php?id=13238 Maybe they're all related to each other. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] is there a reference guide to pri debug span messages?
Maybe I just haven't thought of the right google search terms -- but is there a website/guide out there that will help me understand the output from pri debug span? ___ -- 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] is there a reference guide to pri debug span messages?
You would be much better off trying this question on the asterisk-users list. Much more traffic and geared towards Asterisk in general :) -Brandon John Cheng wrote: Maybe I just haven't thought of the right google search terms -- but is there a website/guide out there that will help me understand the output from pri debug span? ___ -- 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] Returning to Voicemail after returning call
Hello all, I've got dialout= and callback= set in my voicemail.conf so that I can have users return calls to folks who have left messages. They really like this feature. But when the callback is over, a normal hangup occurs instead of the caller being put back into voicemail at the next message. Is it possible that the users be returned into the voicemail system where they left off? thanks Mark ___ -- 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] is there a reference guide to pri debug span messages?
On Oct 23, 2008, at 3:10 PM, John Cheng wrote: Maybe I just haven't thought of the right google search terms -- but is there a website/guide out there that will help me understand the output from pri debug span? ___ perhaps this might be helpful? Q.931 Spec___ -- 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] Inbound DID + voice ports needed for vote monitoring project
Hey folks, I am involved with a group that is going to use Twitter, SMS, iPhone, and Asterisk to field-monitor the upcoming US elections. The group is pretty large scale and you can find out more here: http://votereport.pbwiki.com We need some help with SIP telephony infrastructure. Specifically, we need approximately 200 inbound SIP ports, driven by just one US DID. We have a beefy asterisk box located in NYC and can take delivery of this traffic via the public internet comfortably. Is there a carrier on the list who can provide this kind of capacity between now and November 4 pro-bono, for the good of the US democratic process? Please contact me off-list if this sounds like something you can do. You would receive press and publicity as a partner in return. Thanks, Dave ___ -- 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] Inbound DID + voice ports needed for vote monitoring project
Hello Dave, We can offer you. What area DID you are looking for. Jai Buy SIP DID, www.didforsale.com On Thu, Oct 23, 2008 at 2:20 PM, David Troy [EMAIL PROTECTED] wrote: Hey folks, I am involved with a group that is going to use Twitter, SMS, iPhone, and Asterisk to field-monitor the upcoming US elections. The group is pretty large scale and you can find out more here: http://votereport.pbwiki.com We need some help with SIP telephony infrastructure. Specifically, we need approximately 200 inbound SIP ports, driven by just one US DID. We have a beefy asterisk box located in NYC and can take delivery of this traffic via the public internet comfortably. Is there a carrier on the list who can provide this kind of capacity between now and November 4 pro-bono, for the good of the US democratic process? Please contact me off-list if this sounds like something you can do. You would receive press and publicity as a partner in return. Thanks, Dave ___ -- 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] adding a second extension
I am able to now call the second extension when setup like this so I
believe I'll leave it alone for a while. Basically added the extension
102 to the main incoming line:
exten = 101,1,Dial(SIP/101SIP/102SIP/[EMAIL PROTECTED],30)
exten = 101,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?lbl_default_1:)
exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
exten = 101,n(lbl_default_0),Hangup()
exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30)
exten = 101,n,Goto(lbl_default_0)
exten = 102,1,Dial(SIP/102,20)
exten = 102,n,Hangup
exten = 102,n,Voicemail([EMAIL PROTECTED])
Both extensions can call each other and both extensions ring when the
main line is called... Strange but whatever.
On Thu, Oct 23, 2008 at 1:47 PM, Stephen Reese [EMAIL PROTECTED] wrote:
On Thu, Oct 23, 2008 at 12:39 PM, Juan Rodríguez [EMAIL PROTECTED] wrote:
And this phone are connected in a local LAN??
Because I see Asterisk receiving a Bad request from 68.156.63.118
If those phones are not in your local LAN, try with a soft phone first.
Could be Zoiper or Xlite.
Besides, use SIP SET DEBUG, for SIP debugging and try to see why is 101
sending a 400 Bad request back to Asterisk.
Both of these phones are on my local lan but the Asterisk server is at
a colo facility on the internet outside of the local lan. The local
lan does use NAT/PAT. I see an error Warning: 399 Bad Request -
'Malformed/Missing FROM: field'. Is this a problem?
Thanks
---
ns1*CLI
--- SIP read from 68.156.63.118:1082 ---
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14
From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914
To: sip:[EMAIL PROTECTED];user=phone
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Max-Forwards: 70
Contact: sip:[EMAIL PROTECTED]:1083;user=phone;transport=udp
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
UPDATE
Supported: replaces, 100rel
Expires: 300
Content-Length: 274
Content-Type: application/sdp
v=0
o=102 157742 157742 IN IP4 172.16.2.18
s=Cisco 7912 SIP Call
c=IN IP4 68.156.63.118
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (14 headers 12 lines) ---
Sending to 68.156.63.118 : 1083 (no NAT)
Using INVITE request as basis request - [EMAIL PROTECTED]
--- Reliably Transmitting (NAT) to 68.156.63.118:1082 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14;received=68.156.63.118
From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914
To: sip:[EMAIL PROTECTED];user=phone;tag=as355e0f84
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=ns1.neocipher.net,
nonce=7c2e1ba9
Content-Length: 0
Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000
ms (Method: INVITE)
Found user '102'
--- SIP read from 64.2.142.116:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060
From: sip:[EMAIL PROTECTED];tag=as401a34d4
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 3064 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
-
--- (10 headers 0 lines) ---
--- SIP read from 64.2.142.116:5060 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060
From: sip:[EMAIL PROTECTED];tag=as401a34d4
To: sip:[EMAIL PROTECTED];tag=as7a2f92a1
Call-ID: [EMAIL PROTECTED]
CSeq: 3064 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=575628ec
Content-Length: 0
-
--- (10 headers 0 lines) ---
Responding to challenge, registration to domain/host name
inbound18.vitelity.net
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 64.2.142.116:5060:
REGISTER sip:inbound18.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK6245e988;rport
From: sip:[EMAIL PROTECTED];tag=as751cb0af
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 3065 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=rsreese, realm=asterisk,
algorithm=MD5, uri=sip:inbound18.vitelity.net, nonce=575628ec,
response=b765dbdebba8af18b19707efe651d65d
Expires: 120
Contact: sip:[EMAIL PROTECTED]
Event: registration
Content-Length: 0
---
--- SIP read from 68.156.63.118:1082 ---
ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP
Re: [asterisk-users] Inbound DID + voice ports needed for vote monitoring project
Dear Sir, Please let us know which specific location you need and we can offer that service for you Regards On Fri, Oct 24, 2008 at 12:20 AM, David Troy [EMAIL PROTECTED] wrote: Hey folks, I am involved with a group that is going to use Twitter, SMS, iPhone, and Asterisk to field-monitor the upcoming US elections. The group is pretty large scale and you can find out more here: http://votereport.pbwiki.com We need some help with SIP telephony infrastructure. Specifically, we need approximately 200 inbound SIP ports, driven by just one US DID. We have a beefy asterisk box located in NYC and can take delivery of this traffic via the public internet comfortably. Is there a carrier on the list who can provide this kind of capacity between now and November 4 pro-bono, for the good of the US democratic process? Please contact me off-list if this sounds like something you can do. You would receive press and publicity as a partner in return. Thanks, Dave ___ -- 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] Emerging dilema? DID forwarding meets SMS
We have a number of DID's that do the standard VoIP tricks: ringing multiple locations, findme-followme etc. What is happening more and more is that customers call those DID numbers, and draw the reasonable conclusion that they are calling mobile numbers because they literally can HEAR that the called party is on a mobile. Consequently many of those customers draw the conclusion that they can safely send SMS's to those DID numbers. Naturally the SMS messages disappear into the ether. It occurrs to me that relaying SMS messages following dialplan logic may become an increasingly common objective. I say the SMS messages 'naturally' disappear but maybe I'm just ignorant to this topic because it has not been important to us in the past. Currently we routinely SEND SMS's from Asterisk triggered by other dialplan events. So far we've never needed to RELAY from one DID to another. Are terrestrial carriers even presented with SMS messages? Is anyone using Asterisk to relay SMS messages? -Karl ___ -- 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] changing from default codec
hello, I am using sip, my default codec is set to gsm in sip.conf Using call files, is there a way to send out a call using ulaw while other channels are using gsm ? tia. ___ -- 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] Asterisk and Cisco Call Manager Express (CME)
I was thinking about complicating my Voip setup by adding CME. I found this example here: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration and here: http://www.pasewaldt.com/cme/cme_index.htm Would anyone like to comment on their experiences using CME with Asterisk... I would like one of my Cisco phones to remain SIP connected directly to my Asterisk system. The second phone I would like to revert back from SIP and connect it to CME and then CME to Asterisk. Is this reasonable or is it a huge pain in the rear? ___ -- 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] problems with some incoming/outgoing calls
Hi, Sorry I forgot to mention versions and post files. Asterisk version: pbx:/etc/asterisk# asterisk -rx core show version Asterisk 1.4.22 built by root @ coope-pbx on a i686 running Linux on 2008-10-22 09:36:35 UTC I'm running zaptel 1.4.12.1 and wanpipe 3.3.14. Also tried zaptel 1.4.11 and 1.4.12, and wanpipe 3.2.7.1, and the problem happens on all versions. extensions.conf: http://www.pastebin.ca/1235312 sip.conf: http://www.pastebin.ca/1235317 zapata.conf: http://www.pastebin.ca/1235318 zaptel.conf: http://www.pastebin.ca/1235322 Let me know if I should be posting any other conf files. On Fri, Oct 24, 2008 at 1:27 AM, Lucas Alvarez [EMAIL PROTECTED] wrote: Hi, which version of asterisk are you running? Perhaps if you post your extensions.conf and others related files you could get more accurate help. If you answer a ringing phone and you can't answer the call, there you could have a network or sip config problem, that means that the SIP packet is not returning to the pbx. Regards. Lucas On Thu, 23 Oct 2008 10:30:29 -0200, Fernando Serto [EMAIL PROTECTED] wrote: Hi, I've been very puzzled lately. I installed a phone system for a friend a few weeks ago, and they're having a problem that I can't get rid of, actually 2 problems. Before I go into the problems, let me tell you about the setup. It's a pretty small setup with only 4 handsets, all Polycom 320s, the server is a Dell SC440 with Intel E2180 CPU (dual core, 2GHz) and 512MB Ram. Internet Connection is an ADSL2, with a not so reliable ISP in australia. For incoming calls, I had a Digium TDM410P with 4xFXO modules and HWEC. Because of these problems, i replaced the Digium card with a Sangoma A200D, but it didn't make any difference to the problems. All phones are hooked up to a Netgear PoE switch. Almost forgot to mention that this is not my first Asterisk setup, and in fact it is my 4th, and I used various SIP handsets before, and also different cards (Analog and Digital), so I'm not a total noob. Let's get to the problems... 1) Some incoming calls cannot be picked up Sometimes, incoming calls, coming through the analog card, cannot be picked up. All handsets are set to ring at the same time on incoming calls. and most of the time, calls can be answered on any of the handsets, but maybe 3 or 4 times a day, all handsets will be ringing, and you go to one handset to answer the call, you pick the handset, and it doesn't answer the call, it keeps ringing, then you go to another handset, and still can't pick up, sometimes, you can even try all 4 handsets, and no luck. but, at other times, you can't answer on the first handset, but you can on another, and it is totally random. but people are pretty pissed off for running around to answer a call. and what puzzles me is that you can sit around watching logs for hours, and it won't happen, other times, it happens 3 times in a row. any ideas? 2) Delay on outgoing calls via SIP People have been saying that when they call people, there's a delay for the call to be answered. For example, caller dials a number, callee answers the ringing phone, but caller is still listening to a ringing tone, and after a few seconds (up to 15 seconds) it sounds like the callee has just answered the call, when in fact, he had already answered a few seconds before. Problem with this is that some callees will hangup before the caller starts talking. These calls are going via pennytel, in australia, which seems to be a pretty good VOIP provider around here, and I've been using it on other setups and never had these issues. Well, sorry for the long first post, but I would really appreciate any suggestions you have. Cheers, Fernando ___ -- 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 and Cisco Call Manager Express (CME)
Dare I ask why you want to do this? Dave On Oct 23, 2008, at 10:00 PM, Stephen Reese wrote: I was thinking about complicating my Voip setup by adding CME. I found this example here: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration and here: http://www.pasewaldt.com/cme/cme_index.htm Would anyone like to comment on their experiences using CME with Asterisk... I would like one of my Cisco phones to remain SIP connected directly to my Asterisk system. The second phone I would like to revert back from SIP and connect it to CME and then CME to Asterisk. Is this reasonable or is it a huge pain in the rear? ___ -- 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 and Cisco Call Manager Express (CME)
On Thu, Oct 23, 2008 at 10:57 PM, David Gibbons [EMAIL PROTECTED] wrote: Dare I ask why you want to do this? Dave I know it seems counter intuitive but I've several examples of it being done and for me it would be for the experience of working with CME. A lot of companies utilize Cisco hardware, I figure why not check it out. I enjoy using Asterisk for my SIP server but there are a number of different configurations out there including using Asterisk as a Voicemail server and Cisco Call Manger as the device to interface with the phone rather then having to flash them and all of that even though I've done it twice and it's not a bad process. Mainly just curious... ___ -- 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] How to add contexts in asterisk realtime?
On Wed, 2008-10-22 at 14:56 -0500, Terry Wilson wrote: hi for any context ,you must to open /etc/asterisk/extensions.conf and insert this line : exten =Realtime/[EMAIL PROTECTED] and (reload) or (restart now) your asterisk You don't have to restart asterisk, just a 'dialplan reload' will suffice. So really there is no impact to a running system. You've obviously never tried doing that on a system with 50,000+ extensions and having to reload every time a new customer signs up via an online web interface... Well, if you have 50K extensions, you'll find the trunk/1.6.x versions a bit easier to bear in this respect; I've redone the reload process so that it takes longer, but the magic is that it locks the dialplan and swaps in the new dialplan in about 4-10 microseconds. So, no matter the size of the dialplan, literally no interruption to running code takes place... But you'll find that you can only do so many restarts per unit time... That said, I'd still advise using a db if large numbers of non-pattern numbers are what's in the extensions... I've not done benchmarks on speed, but it could be, that if you use the fast pattern matcher, that the dialplan lookups could be faster than db lookups. If anybody's done any comparisons, let me know... murf ___ -- 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 -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Devstate and Voicemail
Philipp Kempgen wrote: Jared Smith schrieb: On Thu, 2008-10-23 at 09:36 -0700, Marc Hudson wrote: I've looked at 'core show hints' and it is in fact reporting INUSE when it's not, and NOT_INUSE when it is. That definitely sounds like a bug to me. Could you please report this on the bug tracker, so that the developers can take a look and try to reproduce and solve the problem? Sounds a bit like http://bugs.digium.com/view.php?id=13668 or http://bugs.digium.com/view.php?id=13238 Maybe they're all related to each other. Philipp Kempgen Yeah, looks like http://bugs.digium.com/view.php?id=13668, getting -1/0/0 in 'sip show inuse'. Odd that VoiceMailMain of all things happened to trigger it in this case. Thanks, Marc Hudson ___ -- 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
