[asterisk-users] error retrieving a video voicemail in asterisk 11
Using asterisk 11.16.0 I am unable to retrieve any voicemail with a video attachment while using any video phone. This does work in my 1.8.23.1 installation. The file is skipped with the ast_streamfile error (and moved to OLD), and the prompts following that message display the ast_best_codec error. [Apr 7 16:05:50] WARNING[17497][C-6fdd]: file.c:1017 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/2036/INBOX/msg (format (ulaw|h264)): No such file or directory [Apr 7 16:05:50] WARNING[17497][C-6fdd]: app_voicemail.c:8609 play_message: Playback of message /var/spool/asterisk/voicemail/default/2036/INBOX/msg failed [Apr 7 16:05:50] -- SIP/2036-00ee Playing 'vm-advopts.gsm' (language 'en') [Apr 7 16:05:50] WARNING[17497][C-6fdd]: channel.c:940 ast_best_codec: Don't know any of (h264) formats The file does exist in h264 format -rw-r--r-- 1 root root 298102 Apr 7 16:05 msg.h264 -rw-r--r-- 1 root root301 Apr 7 16:05 msg.txt -rw-r--r-- 1 root root 124524 Apr 7 16:05 msg.wav Passthrough h264 video does work. I do have h264 and ulaw codecs on the peer and videosupport=yes in sip.conf. I also tried enabling h264 in the general section of sip.conf and gsm in voicemail.conf with the same results. If I disable the h264 codec for the peer, I can listen to the audio portion of the message: [Apr 7 16:41:05] -- SIP/2036-00f3 Playing '/var/spool/asterisk/voicemail/default/2036/Old/msg.slin' (language 'en') Any guesses what I might be doing wrong? Did something related change in asterisk 11? -- Stephen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agent Penalty
Can anyone shed any light on an issue with agent penalties?
I have 2 queues set up with agents working both queues, but where agent
1 should have a penalty for queue 2 and agent 2 should have a penalty
for queue 1. When a call is sent to either queue, it rings agents with
and without penalties at the same time.
I set up a second system and cannot replicate the issue on the test
system. I have completely stopped asterisk and restarted it thinking
that perhaps a change did not take effect with a reload.
Below are the configurations, debug and some cli output.
This is running 1.0.2, but I didn't see any fixed bugs related to this
on mantis.
queues.conf
[support]
music = 2
strategy = ringall
context = support
timeout = 15
announce-holdtime = yes
announce-frequency = 60
member = Agent/235,0
member = Agent/223,1
[commercial]
music = 2
strategy = ringall
context = default
timeout = 10
announce-holdtime = yes
announce-frequency = 60
member = Agent/223,0
member = Agent/235,1
agents.conf
[agents]
ackcall=no
wrapuptime=5000
agent = 223,,Agent 1
agent = 235,,Agent 2
extensions.conf
[agents]
exten = 223,1,SetVar(BASEEXTEN=${EXTEN})
exten = 223,2,Dial(SIP/${EXTEN},24,rt)
exten = 223,3,Hangup
exten = 235,1,SetVar(BASEEXTEN=${EXTEN})
exten = 235,2,Dial(SIP/${EXTEN},24,rt)
exten = 235,3,Hangup
[features]
exten = 904,1,AgentCallbackLogin(${CALLERIDNUM}|[EMAIL PROTECTED])
exten = 904,2,Hangup
exten = 905,1,Dial(Local/[EMAIL PROTECTED]/n,,D(#))
exten = seagentlogout,1,AgentCallbackLogin(${CALLERIDNUM})
exten = seagentlogout,2,Hangup
[support]
exten = 1,1,DigitTimeout(2)
exten = 1,2,Background(hpbx/20)
exten = 1,3,Random(50:69)
exten = 1,4,SetCIDName(Support Queue - ${CALLERIDNAME})
exten = 1,5,Queue(support|t)
exten =
1,69,Monitor(wav,queue-1-${UNIQUEID}-${EPOCH}-${CALLERIDNUM},bm)
exten = 1,70,Goto(4)
-- Executing DigitTimeout(SIP/10.0.226.26-b622b728, 2) in new
stack
-- Set Digit Timeout to 2
-- Executing Random(SIP/10.0.226.26-b622b728, 50:69) in new
stack
-- Executing SetCIDName(SIP/10.0.226.26-b622b728, Support Queue -
Cell Phone) in new stack
-- Executing Queue(SIP/10.0.226.26-b622b728, support|t) in new
stack
-- Started music on hold, class '2', on SIP/10.0.226.26-b622b728
-- Stopped music on hold on SIP/10.0.226.26-b622b728
-- Playing 'queue-youarenext' (language 'en')
-- Told SIP/10.0.226.26-b622b728 in support their queue position
(which was 1)
-- Playing 'queue-thankyou' (language 'en')
-- Started music on hold, class '2', on SIP/10.0.226.26-b622b728
-- outgoing agentcall, to agent '235', on 'Local/[EMAIL PROTECTED],1'
-- Called Agent/235
-- outgoing agentcall, to agent '223', on 'Local/[EMAIL PROTECTED],1'
-- Called Agent/223
-- Executing SetVar(Local/[EMAIL PROTECTED],2, BASEEXTEN=235) in
new stack
-- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/235|24|rt) in new
stack
-- Called 235
-- Agent/235 is ringing
-- Executing SetVar(Local/[EMAIL PROTECTED],2, BASEEXTEN=223) in
new stack
-- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/223|24|rt) in new
stack
-- Called 223
-- Agent/223 is ringing
-- SIP/235-8429 is ringing
-- SIP/223-cc03 is ringing
show queues
commercial has 0 calls (max unlimited) in 'ringall' strategy (33s
holdtime), C:7, A:0, SL:0.0% within 0s
Members:
Agent/223 has taken 1 calls (last was 6392 secs ago)
Agent/235 with penalty 1 has taken no calls yet
No Callers
support has 0 calls (max unlimited) in 'ringall' strategy (21s
holdtime), C:14, A:1, SL:0.0% within 0s
Members:
Agent/235 has taken 3 calls (last was 71 secs ago)
Agent/223 with penalty 1 has taken 2 calls (last was 6318 secs
ago)
No Callers
show agents
223 (Agent 1) available at '[EMAIL PROTECTED]' (musiconhold is
'default')
235 (Agent 2) available at '[EMAIL PROTECTED]' (musiconhold is
'default')
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RE: [Asterisk-Users] return a value from dial macro
I would really appreciate any insight here. I have seen a number of
posts in the past regarding implementation of a voicemail detection
scheme using silence detection as well as the machine detect, but
without MACRO_RESULT, there doesn't appear to be any way to actually
implement this.
Thanks
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Dolloff
Sent: Tuesday, April 26, 2005 8:43 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] return a value from dial macro
Does anyone know of a way to pass a value back to the dial plan after
calling a macro from the dial app in the 1.0 release? I think this
should be pretty simple, but I can't quite figure out how.
The example would work except that the modified value of found is not
usable when Dial ends. I think that the MACRO_RESULT would do this,
but
it doesn't appear to have made it into 1.0
I want to stop going through the priorities after completion of a
successful dial, but only if MachineDetect returns 0. If it returns 1
I
want to hangup on the called party and goto the next priority
exten = 223,3,SetVar(__found=0)
exten = 223,4,Dial(SIP/[EMAIL PROTECTED],48,rtgM(md))
exten = 223,5,GotoIf($[${found} = 1]?7)
exten = 223,6,Voicemail(u${EXTEN})
exten = 223,7,Hangup
[macro-md]
exten = s,1,MachineDetect(700,2,2200)
exten = s,2,GotoIf($[${MACHINE} = 1]?3:5)
exten = s,3,SoftHangup(${CHANNEL})
exten = s,4,Goto(6)
exten = s,5,SetVar(found=1)
exten = s,6,NoOp
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[Asterisk-Users] return a value from dial macro
Does anyone know of a way to pass a value back to the dial plan after
calling a macro from the dial app in the 1.0 release? I think this
should be pretty simple, but I can't quite figure out how.
The example would work except that the modified value of found is not
usable when Dial ends. I think that the MACRO_RESULT would do this, but
it doesn't appear to have made it into 1.0
I want to stop going through the priorities after completion of a
successful dial, but only if MachineDetect returns 0. If it returns 1 I
want to hangup on the called party and goto the next priority
exten = 223,3,SetVar(__found=0)
exten = 223,4,Dial(SIP/[EMAIL PROTECTED],48,rtgM(md))
exten = 223,5,GotoIf($[${found} = 1]?7)
exten = 223,6,Voicemail(u${EXTEN})
exten = 223,7,Hangup
[macro-md]
exten = s,1,MachineDetect(700,2,2200)
exten = s,2,GotoIf($[${MACHINE} = 1]?3:5)
exten = s,3,SoftHangup(${CHANNEL})
exten = s,4,Goto(6)
exten = s,5,SetVar(found=1)
exten = s,6,NoOp
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RE: [Asterisk-Users] Echo Cancellation
We use a product from oriontelecom.com. The interface is rough, but we have not had a single problem since putting this in. Stephen Dolloff DLS Internet Services 847-854-4799 x256 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Cook Sent: Wednesday, February 09, 2005 5:11 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Echo Cancellation Can anyone provide a good manufacturer of echo cancellation equipment for a PRI? -- Richard Cook [EMAIL PROTECTED] T: 705-497-9320 x2010 image001.gif___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linksys PAP2-NA
Does the PAP-NA2 work with the Sipura firmware and tftp provisioning options? Stephen -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric Merkel Sent: Wednesday, September 22, 2004 9:07 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Linksys PAP2-NA I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It installed pretty easily and has worked great so I went to order some more of these units today. When I logged into Tech Data this morning, the PAP2-NA was now marked as discontinued and no longer available and only the PAP2 version was available which is the Vonage branded version. :( I saw someone on the list say that they heard from Cisco that these units were not due out until Dec. Did Cisco/Linksys pull these units off the shelves? -- Eric Merkel MetaLINK Technologies, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATA186 v3.1 SIP - Attended transfer: NO JOY
I have a similar issue with Sipura using compact headers, but not with regular headers. I am working on reproducing with the latest CVS. Maybe you are using compact SIP headers on your ATA186? http://bugs.digium.com/bug_view_page.php?bug_id=0001843 Stephen -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Florian Overkamp Sent: Wednesday, June 16, 2004 12:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ATA186 v3.1 SIP - Attended transfer: NO JOY Hi, I'm still hassling with the consultative/attended transfer stuff. Someone please help me identify this A lot has already been said about the ATA186. Some report it works fine, others say it doesn't. Lets get clarity on this. My scenario is reasonably simple (I think) Phone A: SIP/video1 Phone B: SIP/werkkamer Phone C: IAX2/provider Phone A calls phone B, they chat: *CLI show channels Channel (ContextExtensionPri ) State Appl. Data SIP/werkkamer-91f5 (from-werkkamer 1 ) Up Bridged Call SIP/video1-e2a0 SIP/video1-e2a0 (pbx1202 1 ) Up Dial SIP/swissSIP/snomSIP/werkkamerIAX2/florianSIP/video1 2 active channel(s) Phone B hits flash and gets a dialtone. Dials a number and connects to phone C: *CLI show channels Channel (ContextExtensionPri ) State Appl. Data IAX2[172.28.8.8:4569]/7 ( s1 ) Up Bridged Call SIP/werkkamer-2507 SIP/werkkamer-2507 (pbx4307076 2 ) Up Dial IAX2/provider/4307076 SIP/werkkamer-91f5 (from-werkkamer 1 ) Up Bridged Call SIP/video1-e2a0 SIP/video1-e2a0 (pbx1202 1 ) Up Dial SIP/swissSIP/snomSIP/werkkamerIAX2/florianSIP/video1 4 active channel(s) Phone A now hears music on hold. Phone B and C can chat. Phone B now hits flash again. All phones end in a three-way conversation: *CLI show channels Channel (ContextExtensionPri ) State Appl. Data IAX2[172.28.8.8:4569]/7 ( s1 ) Up Bridged Call SIP/werkkamer-2507 SIP/werkkamer-2507 (pbx4307076 2 ) Up Dial IAX2/provider/4307076 SIP/werkkamer-91f5 (from-werkkamer 1 ) Up Bridged Call SIP/video1-e2a0 SIP/video1-e2a0 (pbx1202 1 ) Up Dial SIP/swissSIP/snomSIP/werkkamerIAX2/florianSIP/video1 4 active channel(s) Now the misery starts: If Phone B wants to back out of the conversation, it seems phones C and A are also disconnected. I've tried doing this with SIP firmwares, 2.15, 2.16, 3.0 and 3.1 and CVS HEAD as of today. Other people have claimed success: http://lists.digium.com/pipermail/asterisk-users/2003-August/018388.html Is this: http://lists.digium.com/pipermail/asterisk-users/2003-August/018414.html also related ? By the way, canreinvite=no as suggested by Mark in one of the slightly related conversations on bugs.digium.com does not help... I would really _love_ to know why this is and to see it fixed somehow. A bounty would be in order. Can anyone comment on this ?? On a related note: If the consultation ends in a failure (user unavailable or unable to talk) the way to back out is hitting flash once if the remote hung up (ata doesn't give any tone at that time??) or twice if you got voicemail. The remote (phone A) briefly hears this, as the first flash opens a three-way conversation with phones A, B and the voicemail. The second one then disconnects the voicemail again. Not really elegant (albeit useable). Is there a better way ? Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Seperate asterisk VM system possibility
I would like to move voicemail to a dedicated server but I can't figure out how to make the MWI work if the ATA doesn't register to the voicemail server. The main reason for this is redundancy. I have two SIP registrars running and in the case of a failure from the primary, both the gateways and the ATAs switch over to the secondary, but since the voicemail is on the primary, it also fails. Anyone have any suggestions? Thanks, Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mystery SIP channels
Has anyone seen this before? This channel is consistently present on both of my asterisk servers. Sometimes they disappear for a few seconds and then come back. It always has the same Call ID. voip1*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.102(None) df92fb1b-8a 00101/03059 0ms ms UNKN ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Mystery SIP channels
I don't actually know. All of the users are behind NAT, so the channel list doesn't match the peers list. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nik Martin Sent: Thursday, May 20, 2004 10:48 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Mystery SIP channels What address is that? Is it a phone (or address of a PC with a softphone?) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Dolloff Sent: Thursday, May 20, 2004 10:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Mystery SIP channels Has anyone seen this before? This channel is consistently present on both of my asterisk servers. Sometimes they disappear for a few seconds and then come back. It always has the same Call ID. voip1*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.102(None) df92fb1b-8a 00101/03059 0ms ms UNKN ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sipura fade to static
Get an RMA. I've had a few that did that as well. Stephen Dolloff DLS Internet Services 847-854-4799 x256 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Christopher J. Wolff Sent: Thursday, April 01, 2004 5:50 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] sipura fade to static Hello, One of the Sipura 2k's I'm using has a dialtone that occasionally fades to static when the user picks up the line. Are there any settings that I can check that would affect this? Regards, Christopher ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 and short delay before voice startsafter ring.
We have the same complaint here. The caller doesn't hear the receiver say hello and so no-one knows what's going on. Stephen -Original Message- From: James Sizemore [mailto:[EMAIL PROTECTED] Sent: Thursday, March 11, 2004 9:38 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice startsafter ring. exten = 6500,1,Answer exten = 6500,2,Wait,1 exten = 6500,3,VoicemailMain2 Or should I say, Me too! Is this the bug for the case in question? CSCed48311: Media takes 0.4 sec to be set up Thanks. -Andrew Yes the problem is that when making outgoing calls, there is enough of a delay in the call setup once the remote side picks up, that people that answer the phone hello will be heard saying o or if they talk fast enough not heard at all therefor leaving a very awkward silence at the start of a call. This is very annoying. A earlier person suggested answering the calls before dialing and playing a ringing sound till the start of the voice. That may be a work around of sorts for some, you will hear a ring then a congestion tone on call that can't connect, or a ring before a operator messages (say to dial one before the number) that most users may not be used to. I'll be playing with that ideal to see what odd effect a ring has before call setup causes. The work around may be less annoying then the problem. smile I'll see. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E911 support
I haven't looked into it, but either * or the AS5350 gateway that I use sees the Anonymous text and sets the appropriate flags. -Original Message- From: John Fraizer [mailto:[EMAIL PROTECTED] Sent: Thursday, February 26, 2004 3:32 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] E911 support Steve Dolloff wrote: I have the following in my sip.conf entries: callerid=Anonymous 8885551212 This still passes the number for 911, but flags the call as private. I believe this will meet your requirements. Stephen OK. I was under the impression that the PSAP got their information based on ALI/ANI and not from CLID. Are you telling me that they're looking at CLID? Also, at least in the testing I've done, the text portion of the CLID string is ignored by the telco. They only look at the number and generate the text based on what is in their database. IE; If I tell my asterisk server to set my callerID to test my home number and call someplace, What I get on the CLID display of the phone I dial is John Fraizer and my home number. Since Powell has stated that we must provide E911 services, I am wondering what precisely is going to have to be done to do so with Asterisk. Routing the call to the PSAP when someone dials 911 is the easy part. Sending all of the information they want/need (much more than just CLID and something that is regulated) is an alltogether different story. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, specificallyCLID priva cy
I have the following in my sip.conf entries: callerid=Anonymous 8885551212 This still passes the number for 911, but flags the call as private. I believe this will meet your requirements. Stephen -Original Message- From: Olle E. Johansson [mailto:[EMAIL PROTECTED] Sent: Thursday, February 26, 2004 10:17 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, specificallyCLID priva cy Low, Adam wrote: Hey All, I have a Cisco AS5300 running SIP against an Asterisk server with multiple C7940 phones. My issue is that from what I see in chan_sip.c there is no support for the Remote-Party-ID field in relation to withholding the calling partys number. This is a legal requirement for many countries and although it doesnt appear as an Impressed. Does some countries have laws on SIP implementations? Wow. ;-) Is this something planned to be added or perhaps a minor oversight ? If it's somethine planned to be added is really up to your (our someone else's) willingness to code... :-) Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=full Could you please point me in direction of standard documents, drafts or documentation of this? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Memory usage
Can anyone else share their memory use experiences? I am currently running * with about 100 sip.conf entries and 400 dialplans. The memory usage starts at around 10M and goes up every day. After 5 days, it is currently at 90M. Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip flow diagram?
http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_programm ing_reference_guide_book09186a0080080221.html Stephen Dolloff DLS Internet Services 847-854-4799 x256 [EMAIL PROTECTED] -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 11:45 AM To: Asterisk-a-users-list Subject: [Asterisk-Users] Sip flow diagram? Does anyone have a high level flow diagram showing acceptable sip messages exchanges? For exampe: Source Dest Invite - -Trying Ok - I'm specifically trying to debug an issue with various hangups, prior to call completion, after call completion, calling vs called party hold, etc, and getting rather confused watching the various packets flowing between sip devices with a sniffer (and no reference document). Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NO DTMF detection in the Outgoing call with GW Cisco5300
In terms of your dtmf settings, you need to make sure that the 5300 is configured with the same dtmf-relay method and codec as Asterisk. I am also trying to do this using SIP ATAs. It works fine for most calls, but certain ones do not. I have been working with Cisco on this and it appears that the problem is that some IVR systems do not correctly send the PRI Connect status message. Because of this, the AS5300 does not tell * that the call is connected and so SIP info messages are never sent from the ATA. My concern is that this also happens with inband even when the rtp stream is up. Asterisk people, pay attention here. If I connect either a Cisco or a Sipura device directly to the AS5350, it works fine. If I try to connect through *, the DTMF never appears to be sent. I have tried this with INFO, inband and rfc2833 modes and have changes the dtmfmode settings on all 3 devices for each. Stephen -Original Message- From: Areski [mailto:[EMAIL PROTECTED] Sent: Friday, January 16, 2004 10:59 AM To: Asterisk-Users Mailing-list Subject: [Asterisk-Users] NO DTMF detection in the Outgoing call with GW Cisco5300 Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK -- GW AS5300 -- PSTN But the DTMF is working correctly when it's an incoming call. PSTN - - GW AS5300 - - ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I disable ulaw the outgoing call doesn't work. disallow=all ;allow=ulaw allow=alaw debug message: File chan_sip.c, Line 5590 (sip_request): Asked to get a channel of unsupported format ULAW while capability is ALAW Why Asterisk doesn't use the SAME codec with outgoing incoming calls ? In my AS5300, dtmf is configured as dtmf-relay rtp-nte perhaps I should try with h245-signal or h245-alphanumeric ? ALL ideas will be really appreciated ! Cheers, Areski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Credit Card Terminal
Sipura recommended disabling the echo cancellation on the SPA-2000 for modem pass-through. It does help although still not 100% success rate. Stephen -Original Message- From: Christopher J. Wolff [mailto:[EMAIL PROTECTED] Sent: Thursday, January 15, 2004 12:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Credit Card Terminal Hello, I have a Hypercom T7P swipe card terminal sitting on a dedicated G711ulaw port. The Hypercom operates at either 1200 or 2400bps. I get about a 50% success rate when I try to authorize cards. On this same G711ulaw port, I have a fax machine with a 100% success rate operating at 9600bps. Any suggestions on how to change *, ATA186, or SIPURA SPA-2000 to enhance the card terminals ability to process would be appreciated. Regards, Christopher J. Wolff, VP CIO Broadband Laboratories, Inc. http://www.bblabs.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] USA dial plan
Some areas in the US also use 10 or 11 digital dialing for all calls, whether they are local, long, toll or non-toll. -Original Message- From: Eric Wieling [mailto:[EMAIL PROTECTED] Sent: Friday, January 09, 2004 1:53 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] USA dial plan Generally speaking, Yes. The usual dial plan in the USA is as follows: NXX- (Free Local Call to number in same Area Code) NXX-NXX- (Free Local Call to number in different Area Code) 1-NXX- (Toll Call to number in same Area Code) 1-NXX-NXX- (Toll Call to number in different Area Code) 1-800-NXX- (Toll Free Call) 1-855-NXX- (Toll Free Call) 1-866-NXX- (Toll Free Call) 1-877-NXX- (Toll Free Call) 1-888-NXX- (Toll Free Call) Yes, in most places in the USA local calls are totally free, no per min charge. Some parts of the USA have Local Toll Calls, that is calls that are dialed as NXX- that are not free, but have a very small per min cost. Los Angels is one of these places I think. On Fri, 2004-01-09 at 12:50, Senad Jordanovic wrote: Hi, Do the callers in USA dialling from USA Telco lines always have to prefix the CITY/AREA code with 1 in order To successfully make a call to other USA destinations? I have not been to USA (yet) :) Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk log messages
I have 2 questions regarding asterisk logs that I really hope someone can help me with. Jan 7 09:40:14 WARNING[1009517568]: File chan_iax.c, Line 3537 (registry_rerequest): Received unsolicited registry authenticate request from '209.242.15.34' I get this IAX message every minute or so. I have 2 asterisk servers that both register with each other. I can post the configuration again, but it should be in the list multiple times already. Jan 7 09:40:25 NOTICE[1011735552]: File chan_sip.c, Line 5355 (handle_request): Registration from 'Smith, John sip:[EMAIL PROTECTED]' failed for '209.242.0.1' How can I get more information on what is causing the failure? This same user authenticates fine most of the time, but I still get these types of messages much too frequently. Stephen Dolloff DLS Internet Services 847-854-4799 x256 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF via SIP not working for certain phone systems
I really hope that someone can help me with this one. DTMF tones are not working for certain places that I call, specifically 1-800-882-8880 which is the AA advantage line. It works for almost everyplace else. If I bypass asterisk, the call works fine. Network looks like: SPA-2000 --SIP-- ASTERISK --SIP-- AS5350 --PRI-- PSTN sip.conf entries [VGW01] (this is the AS5350) type=friend nat=no host=192.168.0.1 context=default [8475551212] (this is the SPA-2000) type=friend secret=XX nat=yes host=dynamic canreinvite=no qualify=yes mailbox=8475551212 context=unlimited SPA-2000 is using INFO AS5350 is using dtmf-relay rtp-nte Everything is using G711u. I have also tried setting the whole system to inband including the sip.conf entries. The weird part is that if I watch the network traffic, I don't see the SIP INFO messages for the dtmf when I'm connected to the number listed above. Normally, I do. I have also tried this with an ATA-186 with the same results. Any suggestions would be very appreciated. Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] one way choppy sound problem !
I am having the same problem, but only with one specific user, so I believe it is network related. Anyone that can point me in the specific direction of what would cause this? -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: Monday, January 05, 2004 10:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] one way choppy sound problem ! Michael Van Donselaar wrote: On Mon, 5 Jan 2004 13:29:06 +0100, Dawid Mielnik [EMAIL PROTECTED] wrote: Hi Again, Apart from X-lite client I have also tried eStara, diax phone, iaxcomm and some others. I have tried different codecs - GSM, aLAW uLAW. They all give the same result. In the direction PSTN user --- Softphone user the sound is crystal clear (also tried on a dial-up connection), in the other direction however the sound is a bit choppy. The chops occur at regular intervals of time, at about 1-2 seconds !? Are the PSTN interface and a network card sharing an interrupt? I had similar problems with my X100P and a thunderlan dual ethernet card shring IRQs (also would make one of the ethernet ports fails until reboot) Are you still using the P133? I tried using a P120, but it wouldn't do the trick with GSM conversion. DIAX and iaxComm, since they use the iaxclient library, need to use GSM. I have the same choppy sound problem on my server, my card is not sharing an interrupt and I am using G711 which is not hittng the P2 400 at all.. It seems there is a gremlin.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Backup Proxy Automatic Failover
I simply have 2 asterisk servers and have the clients point to a DNS SVR record for their proxy. The DNS record lists the primary and secondary with preference for the primary. This won't stop calls from being dropped if the primary goes down if you are routing them through the server, but it does ensure that calls placed while the primary is down will still go through. You could do some load management by putting multiple servers in the DNS record and use a DNS server that supports round robin responses. Stephen -Original Message- From: Adthrawn [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 12:50 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Backup Proxy Automatic Failover Hi, I read in the Asterisk Whitepaper, that you can run two cloned servers, one as a primary, one as a backup, and have them automatically failover to the other unit when it crashes, or when you need to restart it. The primary application of course, would be ensuring calls can be made when frequent updates are being handled, or when an update must be restarted on a busy network. The term TDM is banded around too, but from my knowledge, TDM is trunking (probably some clever acronym relating to trunking), and in Asterisk's case, using the IAX protocol. This leads me to the big question; Is there anyway of shifting the load of one Asterisk server to another without breaking or loosing a call? I know that with Survivable Routing (Cisco's big on this), the ISDN interface is actually a router; so the Proxy is just used to decide the destination and LCR functions, and then hands off to a router. This of course, if a Proxy went down, would just prevent new calls from being made, whilst existing calls can continue merrily - until someone switches the Router off, or corrupts the IOS settings :-) At least with Routers, you can configure them to load manager effectively, but how do you backup and load manage Asterisk?? I using SIP, and will be using a bit of SCCP too, so any suggestions would be most grateful!! Regards, Ad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Land line vs. VoIP provider.
Not all VoIP providers will have Vonage's 911 issues. It's perfectly possible for a VoIP provider to provide outbound caller information to the PSAPs if they spend the time and money to do so. Stephen Summary: if you're the only caller, calling only to the US, then you might be crazy to not use a land line, especially given the deals currently available and the 911 issue (but see http://www.vonage.com/features_911.php). Even then, if you already have broadband in house (or at home), VoIP amy be an attractive alternative, if only for the control it gives you over your phone service. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco Gateway Integration
Is there an appropriate place to post network diagrams, configuration files, system info, etc for future implementers? I would like to give back to the community, but I don't want to maintain a separate web site for it. Stephen Btw, I'm still looking for answers to a few questions that I posted yesterday to finish my implementation. -Original Message- From: Asterisk [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 17, 2003 10:15 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco Gateway Integration Hi, Where can I find more information on your setup. I would like to do something similar. Thanks, Seth - Original Message - From: Steve Dolloff [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 16, 2003 11:15 AM Subject: RE: [Asterisk-Users] Cisco Gateway Integration I am using it with the AS5350 via SIP and it works great. I was also using the ATA186 with SIP but I am switching to the SPA-2000 for a better feature set. Stephen -Original Message- From: Bruce Hedreen [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 16, 2003 8:29 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco Gateway Integration Did you use the h323 module on asterisk? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Tinchev Sent: Tuesday, December 16, 2003 12:37 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco Gateway Integration Bruce Hedreen wrote: Has anyone succesfully integrated * with a cisco voice gateway ? Works well with AS5350 and ATA186. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco Gateway Integration
I am using it with the AS5350 via SIP and it works great. I was also using the ATA186 with SIP but I am switching to the SPA-2000 for a better feature set. Stephen -Original Message- From: Bruce Hedreen [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 16, 2003 8:29 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco Gateway Integration Did you use the h323 module on asterisk? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Tinchev Sent: Tuesday, December 16, 2003 12:37 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco Gateway Integration Bruce Hedreen wrote: Has anyone succesfully integrated * with a cisco voice gateway ? Works well with AS5350 and ATA186. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Requesting advice from experienced * users/developers
Greetings, I have a couple of questions and figured I would put them all in one message to not spam the list as much as possible. I have searched voip-info, google and the list archives for all of these questions. If I have missed the correct response, please accept my apologies. I have been stuck on these for a long time and I am really hoping that the other users out there will be able to help me out. 1) VM attendant sounds scratchy? I am using the G711ulaw codec via SIP. The messages themselves sound fine, recorded in WAV49 format (chosen for least hd space). This is especially true when reading the numbers from a mailbox or the number of messages. I would guess that it's due to the attendant messages being recorded in gsm, but I have recorded a few messages using the Record App and they sound fine too. Is this something to do with the VM app? 2) Privacy manager/Zapateller not working correctly. I have posted the details of this problem in a previous post, but basically, they don't seem to recognize anonymous callers vs callers with no caller-id. 3) Call-waiting caller-id doesn't work. (but either by themselves do) I am not using zapta interfaces, but I have enabled the appropriate setting in that conf file in case it has some relevance. I didn't see a similar option in the sip.conf file. I have tested with both the ATA186 and the SPA-2000. 4) I am looking for a provider of caller-id name database services on a U.S. national basis for incoming calls using an AGI script. Willing and able to post all relevant config files. Thanks for any help you can provide. Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call-waiting caller-id
Are there any known issues with call-waiting caller-id for SIP? Caller-ID on the first call works fine, but when the second call comes in, I hear the interrupt tone, but the caller-id doesn't display anything. I have tried this with the Cisco ATA and the SPA-2000. I have also tried two different phones to verify that it wasn't something specific to the phone. Thanks, Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX error messages in log
I constantly get the following error messages in /var/log/asterisk/messages: Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 3324 (iax_ack_registry): Received unsolicited registry ack from '192.168.0.1' Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181 (socket_read): Registration failure Where 192.168.0.1 is another asterisk server. Below are the local and remote IAX configurations. Local server: register = [EMAIL PROTECTED] ; [voip2p] type=peer host=dynamic port=4569 trunk=no qualify=yes context=IAX Remote server: register = [EMAIL PROTECTED] ; [voip1p] type=peer host=dynamic port=4569 trunk=no qualify=yes context=IAX ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Port density: DS3 cards?
I would be seriously wary of putting a DS3's worth of voice traffic on a TNT. I don't believe they are rated to handle that much voice. The APX1000 would be a much better platform, but I don't know if you can find one used. Stephen -Original Message- From: Ernest W. Lessenger [mailto:[EMAIL PROTECTED] Sent: Thursday, December 04, 2003 4:51 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Port density: DS3 cards? At 02:34 PM 12/4/2003, you wrote: However, considering the traffic volumes that you are talking about, is it really true to say that the traditional telco cards are astronomically priced, given the amount of revenue that can be generated per month on a DS3? Eight quad-span T-1 cards from Digium: $8,970 Three reasonable-quality asterisk servers: $1,000 One T-1/DS-3 MUX: $5000 Total system cost: $14,970 That actually sounds quite reasonable to me. However, if I were doing this myself I would look hard at getting a MAX TNT with VoIP capability off eBay. The price would be equivalent or less, the interface would be more complicated, but all the DSP would be done by the MAX. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Issues with Privacy Manager and Zapateller
I am still having these same problems. Anyone with experience with
these apps that could point me in the right direction?
I am having issues with Privacy Manager and Zapateller.
If I set callerid= on a sip user zapateller sends the tones
If I set callerid=Anonymous 8475551212 zapateller doesn't send
the
tones
If I call from a phone after dialing *67 zapateller doesn't send the
tones
In the last 2 cases, the display on the phone shows -Blocked Call-
PrivacyManager always gives the following messages:
-- Executing PrivacyManager(SIP/8475551212-9ec4, ) in new
stack
-- CallerID Present: Skipping
Even when the phone shows -Blocked Call- and even when zapateller
sends
tones.
Here is the Dial-Plan for the extension
exten = _NXXNXX/,1,Zapateller
exten = _NXXNXX,1,NoOp
exten = 847666,2,PrivacyManager
exten = 847666,3,Dial(SIP/${EXTEN},,r)
exten = 847666,4,Hangup
Stephen
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RE: [Asterisk-Users] Issues with Privacy Manager and Zapateller
Anyone have any thoughts on this since last week?
I am having issues with Privacy Manager and Zapateller.
If I set callerid= on a sip user zapateller sends the tones
If I set callerid=Anonymous 8475551212 zapateller doesn't send the
tones
If I call from a phone after dialing *67 zapateller doesn't send the
tones
In the last 2 cases, the display on the phone shows -Blocked Call-
PrivacyManager always gives the following messages:
-- Executing PrivacyManager(SIP/8475551212-9ec4, ) in new
stack
-- CallerID Present: Skipping
Even when the phone shows -Blocked Call- and even when zapateller
sends
tones.
Here is the Dial-Plan for the extension
exten = _NXXNXX/,1,Zapateller
exten = _NXXNXX,1,NoOp
exten = 847666,2,PrivacyManager
exten = 847666,3,Dial(SIP/${EXTEN},,r)
exten = 847666,4,Hangup
Stephen
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RE: [Asterisk-Users] PREPAID APPLECATION
Speaking of voice prompts, could anyone tell me why the pre-recorded prompts sometimes sound garbled, but the voicemail messages themselves sound fine? Is it the format of the prompts? Stephen I would like to release prepaid application. But I have a small problem, we are using their Cisco prompts (nice lady voice) And I do not know if it is ok to release it. Bart I will agree with the comments of others on this topic. You should _not_ include the prompts from Cisco. That is almost certainly a copyright violation. For a very low price, you can have Allison Smith (http://www.theivrvoice.com/) re-record the prompts, as she is the person that did almost all the current Asterisk vocalizations (except for the tt-monkeys.gsm file: that was me.) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Issues with Privacy Manager and Zapateller
I am having issues with Privacy Manager and Zapateller.
If I set callerid= on a sip user zapateller sends the tones
If I set callerid=Anonymous 8475551212 zapateller doesn't send the
tones
If I call from a phone after dialing *67 zapateller doesn't send the
tones
In the last 2 cases, the display on the phone shows -Blocked Call-
PrivacyManager always gives the following messages:
-- Executing PrivacyManager(SIP/8475551212-9ec4, ) in new stack
-- CallerID Present: Skipping
Even when the phone shows -Blocked Call- and even when zapateller sends
tones.
Here is the Dial-Plan for the extension
exten = _NXXNXX/,1,Zapateller
exten = _NXXNXX,1,NoOp
exten = 847666,2,PrivacyManager
exten = 847666,3,Dial(SIP/${EXTEN},,r)
exten = 847666,4,Hangup
Stephen
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RE: [Asterisk-Users] IAX trunk monitoring
Creating a separate user and peer does allow me to call over the trunk
using this format:
exten = 8475551212,3,Dial(IAX/voip2/${EXTEN},31,r)
(this must be a bug. Why would the same format not work for a friend?)
It does not solve my original problem of failing the call if the trunk
is down. Calls now are always sent to the voip2 iax user regardless of
whether that user is connected.
Also, voip2 was created as a user and voip2peer was created as a peer.
If I use:
exten = 8475551212,3,Dial(IAX/voip2peer/${EXTEN},31,r)
or
exten = 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},31,r)
the call fails as unavailable regardless of whether or not the other
server is running.
The registry information looks wrong too:
voip1*CLI iax show registry
Host UsernamePerceived Refresh State
209.242.15.34:5036voip1peer Unregistered 60 Request
Sent
209.242.15.34:5036voip1 Unregistered 60
Rejected
-Original Message-
From: Philipp von Klitzing [mailto:[EMAIL PROTECTED]
aachen.de]
Sent: Friday, November 14, 2003 4:08 AM
To: Steve Dolloff
Subject: RE: [Asterisk-Users] IAX trunk monitoring
You might want to try this:
split the entries for voip1 and voip2, i.e. instead of type=friend
have
an entry for type=peer and type=user for each of the two machines.
Greetings, Philipp
I have modified the configuration for dynamic host and registered each
server with the other. The iax show users now lists the other iax
device as registered vs unavailable, but I still don't know how to keep
it from calling if the device becomes unavailable.
I changed the extensions file to:
exten = 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
also tried:
exten = 8475551212,3,Dial(IAX/voip2/${EXTEN},,r)
since voip2 is now a registered user, but it is not trying to call the
other server.
If I leave it as [EMAIL PROTECTED] it works as long as the trunk is up
but it doesn't check to make sure before sending the call there.
Any suggestions?
---
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] IAX trunk monitoring
I have an issue where * tries to route a call over IAX to another server
even if the server is down. I have included the relevant entries from
my iax.conf, extensions.conf, and some debug output. If someone could
tell me what I have configured incorrectly, I would appreciate it.
Thanks,
Stephen
---iax.conf on voip2--
[voip1]
type=friend
username=voip1
host=x.x.x.x (ip address of voip1)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX
--iax.conf on voip1---
[voip2]
type=friend
username=voip2
host=x.x.x.x (ip address of voip2)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX
-extensions.conf on voip1
exten = 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
-extensions.conf on voip2
exten = 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
voip1*CLI iax show users
Username Secret Authen Def.Context A/C
voip2 md5,plaintextIAX No
voip1*CLI iax show peers
Name/UsernameHost Mask Port Status
voip2/voip2 x.x.x.x (S) 255.255.255.255 5036 UNREACHABLE
voip2*CLI iax show users
Username Secret Authen Def.Context A/C
voip1 md5,plaintextIAX No
voip2*CLI iax show peers
Name/UsernameHost Mask Port Status
voip1/voip1 x.x.x.x (S) 255.255.255.255 5036 UNREACHABLE
voip1*CLI iax debug
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 01 Type: IAX Subclass: ACK
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: INVAL
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
voip1*CLI
voip2*CLI iax debug
IAX Debugging Enabled
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass
[Asterisk-Users] IAX trunk monitoring
I have an issue where * tries to route a call over IAX to another server
even if the server is down. I have included the relevant entries from
my iax.conf, extensions.conf, and some debug output. If someone could
tell me what I have configured incorrectly, I would appreciate it.
Thanks,
Stephen
---iax.conf on voip2--
[voip1]
type=friend
username=voip1
host=x.x.x.x (ip address of voip1)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX
--iax.conf on voip1---
[voip2]
type=friend
username=voip2
host=x.x.x.x (ip address of voip2)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX
-extensions.conf on voip1
exten = 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
-extensions.conf on voip2
exten = 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
voip1*CLI iax show users
Username Secret Authen Def.Context A/C
voip2 md5,plaintextIAX No
voip1*CLI iax show peers
Name/UsernameHost Mask Port Status
voip2/voip2 x.x.x.x (S) 255.255.255.255 5036 UNREACHABLE
voip2*CLI iax show users
Username Secret Authen Def.Context A/C
voip1 md5,plaintextIAX No
voip2*CLI iax show peers
Name/UsernameHost Mask Port Status
voip1/voip1 x.x.x.x (S) 255.255.255.255 5036 UNREACHABLE
voip1*CLI iax debug
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 01 Type: IAX Subclass: ACK
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: INVAL
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
voip1*CLI
voip2*CLI iax debug
IAX Debugging Enabled
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 01 Type: IAX Subclass: ACK
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: INVAL
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
voip2*CLI
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RE: [Asterisk-Users] IAX trunk monitoring
I have modified the configuration for dynamic host and registered each
server with the other. The iax show users now lists the other iax
device as registered vs unavailable, but I still don't know how to keep
it from calling if the device becomes unavailable.
I changed the extensions file to:
exten = 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
also tried:
exten = 8475551212,3,Dial(IAX/voip2/${EXTEN},,r)
since voip2 is now a registered user, but it is not trying to call the
other server.
If I leave it as [EMAIL PROTECTED] it works as long as the trunk is up
but it doesn't check to make sure before sending the call there.
Any suggestions?
---
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] IAX trunk monitoring
I have an issue where * tries to route a call over IAX to another server
even if the server is down. I have included the relevant entries from
my iax.conf, extensions.conf, and some debug output. If someone could
tell me what I have configured incorrectly, I would appreciate it.
Thanks,
Stephen
---iax.conf on voip2--
[voip1]
type=friend
username=voip1
host=x.x.x.x (ip address of voip1)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX
--iax.conf on voip1---
[voip2]
type=friend
username=voip2
host=x.x.x.x (ip address of voip2)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX
-extensions.conf on voip1
exten = 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
-extensions.conf on voip2
exten = 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
voip1*CLI iax show users
Username Secret Authen Def.Context A/C
voip2 md5,plaintextIAX No
voip1*CLI iax show peers
Name/UsernameHost Mask Port Status
voip2/voip2 x.x.x.x (S) 255.255.255.255 5036 UNREACHABLE
voip2*CLI iax show users
Username Secret Authen Def.Context A/C
voip1 md5,plaintextIAX No
voip2*CLI iax show peers
Name/UsernameHost Mask Port Status
voip1/voip1 x.x.x.x (S) 255.255.255.255 5036 UNREACHABLE
voip1*CLI iax debug
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 01 Type: IAX Subclass: ACK
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: INVAL
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
voip1*CLI
voip2*CLI iax debug
IAX Debugging Enabled
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 01 Type: IAX Subclass: ACK
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: INVAL
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
voip2*CLI
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RE: [Asterisk-Users] ReplayTV connecting through Asterisk box
Can someone point me to the echo cancellation settings for a pure sip setup? Thanks, Stephen Subject: Re: [Asterisk-Users] ReplayTV connecting through Asterisk box Has anyone had any luck getting a ReplayTV DVR box to connect through an Asterisk box? Mine seems to dial just fine, but can't negotiate a connection. I am using: I would suggest NOT using the agressive echo cancellor. I think it buggers up modems in a big way. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] passing digits for voicemail from sip gateway
I am seeing strange behavior that I don't understand. Voicemail2 and voicemailmain2 work fine if I call from a sip phone directly connected to *, but if I call either of them from an analog line on the other side of a sip gateway, voicemail seems to ignore digits. If I am recording a message and press #, nothing happens except that it records the tone onto the message and I can't specify a mailbox using digits either, it just hangs up on me. Is this a config problem on the gateway? Thanks, Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Context restrictions
Can someone please explain what I am doing wrong here? I only want the extensions listed in long-users to be able to access the longdistance context. If I do this, I get a congestion tone no matter what I dial. If I add a [default] context and include = longdistance, then the local callers can call the long distance number fine, which is not what I want, but I still want long-users to be able to call locally and I need long and local users to be able to call each other, and inbound calls need to be able to go to local and long users as well. I tried reading the handbook, but even though they say that you can restrict based on context, it never shows an example of how. [local-users] exten = 8478414198,1,Dial(SIP/8478414198) exten = 8478414198,2,Hangup [long-users] exten = 8478414199,1,Dial(SIP/8478414199) exten = 8478414199,2,Hangup [local] exten = _XX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _XX,2,Congestion include = local-users [long-distance] exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _1NXXNXX,2,Congestion include = local include = long-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Carrier
I am looking for a SIP carrier to handle wholesale residential traffic. Standard LEC services in the US. Anyone have any suggestions? Thanks, Stephen Stephen Dolloff DLS Internet Services 847-854-4799 x256 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No Ringing from PSTN
Here is my Configuration PSTN - Cisco AS5350 - SIP - ASTERISK - SIP - ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong? Thanks, Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No Ringing from PSTN
That does make a ringing sound, but any idea what's causing the problem? Stephen Subject: Re: [Asterisk-Users] No Ringing from PSTN You can send a fake ring by using something like: exten = 1234,1,Dial(SIP/[EMAIL PROTECTED],20,r) Assuming the ATA is in the sip.conf as [1234] However, this does NOT solve the underlying problem. On Thu, 2003-10-09 at 15:29, Steve Dolloff wrote: Here is my Configuration PSTN - Cisco AS5350 - SIP - ASTERISK - SIP - ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong? Thanks, Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] getting inbound caller-id from sip remote-party-id field
I am looking for examples or instructions on how to route calls to
voicemailmain based on remote-party-id.
I have the following entry in my extensions.conf file:
exten = 200,1,Voicemailmain(${CALLERIDNUM})
I am routing calls to * via SER and sending Remote-Party-ID in the SIP
headers. I am trying to find out how to map the SIP Remote-Party-ID
field to caller-id so that I can use it to map to the correct extension
for voicemail retrieval.
Thanks,
Stephen
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