RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem
Hi, What is your setup? By MS RTC do you mean Office Communicator? If you are using MS OC, do you use SER in between (to convert SIP UDP2TCP)? Please share some more details J Cheers, Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, June 14, 2006 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem It seems that Microsoft RTC has some problems with originated calls from Asterisk. If I execute Manager API originate application, with SIP channel as parameter, the Microsoft RTC softphone will start to ring after a couple of seconds delay, but nothing more happens after when I answer there is no second call to an extension. When I looked through the sip debug, I noticed that Microsoft RTC fails to properly respond to INVITE messages (I have attached the sip debug). Asterisk has to retransmit INVITE message for 6 times and even then the RTC still doesn't respond in a proper time. However, if I do direct call to that problematic Microsoft RTC based softphone, everything works fine, eventhough very same INVITE messages are being transmited to it from Asterisk. Does anyone have any ideas for a workaround? Regards, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voip to voip bridge
Hi, Check if reinvites are enabled, and that you dont use any parameter in the dial command that forces asterisk to stay in the loop. Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Baum Sent: Wednesday, June 14, 2006 5:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] voip to voip bridge Has anyone had any good experiences with a voip to voip bridge... where you have an incoming call on a voip line which is redirected out another voip line to a regular phone line? Whenever we do this, the connected call is kinda lagged and the quality isn't always that great. It seems to me this is just a problem with the inherent delay in the voip connections. But I was wondering if there's any special configurations that could make the situation better? Erick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem
Hmm.. Interesting, I didnt try to implement it this way... but, if its the same libraries used for Office communicator, than it supports only SIP over TCP or TLS, since asterisk doesnt support any of those its impossible to connect them directly... If udp works, maybe the registration part is problematic, try configuring asterisk with autocreatepeer (just for testing) to see if you can dial out without being registered. Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, June 14, 2006 11:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem Nope, it's just the Microsoft RTC Core 1.3 library ... more or less a single DLL J. And I'm almost sure there is no SER in between should there be one? It's pretty much a straightforward thing I have a few SIP clients defined in my sip.conf, like this: [general] context=default allowguest=yes realm=timd.si bindport=5060 bindaddr=0.0.0.0 srvlookup=yes domain=timd.si,from-sip domain=111.111.111.8,from-sip videosupport=yes disallow=all allow=alaw allow=ulaw musicclass=default rtptimeout=100 rtpholdtimeout=100 tos=0x18 canreinvite=yes [SIPClient001] username= SIPClient001 secret= mysecret type=friend host=dynamic context=from-sip disallow=all allow=alaw allow=ulaw qualify=yes [SIPClient002] username= SIPClient002 secret= mysecret type=friend host=dynamic context=from-sip disallow=all allow=alaw allow=ulaw qualify=yes And there is an MS RTC based Softphone, that I made, on the other side that registers to Asterisk, using this profile XML string: provision key=5B29C449-29EE-4fd8-9E3F-04AED077690E name=Asterisk user account=SIPClient001 uri=sip:[EMAIL PROTECTED] / sipsrv addr=111.111.111.8 protocol=udp auth=digest role=registrar session party=first type=pc2ph / /sipsrv /provision Now, doing an originate to CHANNEL=SIP/SIPClient002, and some extension, will randomly fail, for example (see OriginateFailure reponse as well): action: Originate actionid: 123 exten: 03020846051635424 channel: SIP/SIPClient002 timeout: 3 priority: 1 context: asttel async: true Event: OriginateFailure Privilege: call,all ActionID: 123 Channel: SIP/ SIPClient002 Context: asttel Exten: 03020846051635424 Reason: 1 Uniqueid: null From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 10:14 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem Hi, What is your setup? By MS RTC do you mean Office Communicator? If you are using MS OC, do you use SER in between (to convert SIP UDP2TCP)? Please share some more details J Cheers, Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, June 14, 2006 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem It seems that Microsoft RTC has some problems with originated calls from Asterisk. If I execute Manager API originate application, with SIP channel as parameter, the Microsoft RTC softphone will start to ring after a couple of seconds delay, but nothing more happens after when I answer there is no second call to an extension. When I looked through the sip debug, I noticed that Microsoft RTC fails to properly respond to INVITE messages (I have attached the sip debug). Asterisk has to retransmit INVITE message for 6 times and even then the RTC still doesn't respond in a proper time. However, if I do direct call to that problematic Microsoft RTC based softphone, everything works fine, eventhough very same INVITE messages are being transmited to it from Asterisk. Does anyone have any ideas for a workaround? Regards, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750
Hi, As long for HiPath 4000 callerID name doesn't work, only number -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pavel Jezek Sent: Thursday, May 25, 2006 9:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750 Hi, interoperability between asterisk and siemens interesting me too can you tell me, if caller id _name_ is fully working between asterisk and siemens, and what signaling do you use? currently I have Q.SIG signaling between siemens and ci$co voice gateway (with HDV-E1 module), but because ci$co can't decode caller id name from isdn/Q.SIG, I'm planning to replace ci$co gw with asterisk in near feature :-) PJ Josué Conti wrote: Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip ignores context definition?
Hi All, I've a very strange error. I've configured a Cisco gw with * and when an incoming call is arriving from the Cisco to * asterisk will always put the call in the default context (ignoring the part in the [Cisco]) I'm attaching my conf files: [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=alaw allow=gsm allow=ulaw context = from-trunk ; Send unknown SIP callers to this context callerid = Unknown [Cisco] type=user/friend/peer (tried all options) port=5060 host=myip context=from-Cisco disallow=all allow=alaw allow=ulaw qualify=yes autocreatepeer=yes (with and without this option, in here and in the general setting) nat=no canreinvite=no on Asterisk Console I see (with Verbose 9): Executing AbsoluteTimeout(SIP/myip-b6895f10, 15) in new stack -- Set Absolute Timeout to 15 -- Executing Congestion(SIP/myip-b6895f10, ) in new stack -- Executing AbsoluteTimeout(SIP/myip-b6895f10, 15) in new stack -- Set Absolute Timeout to 15 -- Executing Congestion(SIP/myip-b6895f10, ) in new stack which is my default context: [from-trunk] exten = _.,1,AbsoluteTimeout(15) exten = _.,2,Congestion exten = _.,3,Hangup [from-Cisco] exten = s,1,Answer exten = s,2,Dial($bla) exten = s,3,Hangup Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip ignores context definition?
Hi, Asterisk is 1.09, I've tried to change that like you suggested but no luck. When I'm doing sip debug, its look like it always go to the default sip context. I've a second sip host definition and that works, exactly the same configuration just different IP. Could that be a bug? How can I make sure, and if its a bug, how do I submit it? Thanks again, Ohad What version are you running, and is your [Cisco] definition the last one in the file? I have the same problem with 1.0.7, and the ugly fix I came up with was to add a dummy entry as the last sip entry. B. J. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Sent: Friday, November 11, 2005 4:48 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] sip ignores context definition? Hi All, I've a very strange error. I've configured a Cisco gw with * and when an incoming call is arriving from the Cisco to * asterisk will always put the call in the default context (ignoring the part in the [Cisco]) I'm attaching my conf files: [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=alaw allow=gsm allow=ulaw context = from-trunk ; Send unknown SIP callers to this context callerid = Unknown [Cisco] type=user/friend/peer (tried all options) port=5060 host=myip context=from-Cisco disallow=all allow=alaw allow=ulaw qualify=yes autocreatepeer=yes (with and without this option, in here and in the general setting) nat=no canreinvite=no on Asterisk Console I see (with Verbose 9): Executing AbsoluteTimeout(SIP/myip-b6895f10, 15) in new stack -- Set Absolute Timeout to 15 -- Executing Congestion(SIP/myip-b6895f10, ) in new stack -- Executing AbsoluteTimeout(SIP/myip-b6895f10, 15) in new stack -- Set Absolute Timeout to 15 -- Executing Congestion(SIP/myip-b6895f10, ) in new stack which is my default context: [from-trunk] exten = _.,1,AbsoluteTimeout(15) exten = _.,2,Congestion exten = _.,3,Hangup [from-Cisco] exten = s,1,Answer exten = s,2,Dial($bla) exten = s,3,Hangup Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * behind NAT, client behind NAT(handytone 286), very strange behavior
Hi All, I've an Asterisk Server behind a NAT. Using DNAT, I've opened port 5060 and all 1:2 udp. Sip configured with externalip and subnet. I've another site, also with NAT, where I map the rtp port (as defined in the client) to map to the local client (DNAT). Using Xlite, this configuration works, it requires using the quality=yes and NAT=yes/always in the sip ext configuration but works quite well. However, lately I've purchased a Grandstream ATA Handytone 286 and tried to apply the same settings but When doing an echo test, I can't hear myself, but I can hear the asterisk server (meaning asterisk can reach the client behind the NAT). When doing some tcpdump, it looks like some packets are coming from the client to asterisk, so the network setting looks ok. When calling to another sip device, with or without canreinvite (yes/no) the rtp stream is unable to establish it self, no matter where the second client is (inside/outside NAT). But! When calling using a zap channel (which is on the asterisk server) everything works! I can hear the person I'm talking to and he can hear me. I'm a bit confused.. How could it be that this works and echo test doesnt? Any help would be appreciated! Thanks, Ohad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk security issue
http://archives.neohapsis.com/archives/fulldisclosure/2005-06/0297.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] flash panel only works with IP address
Hi, It seems that my flash panel only works when I specify my ip address and not the host name. I've tried quite a few things (change host file, dns resolve, proxying.) but couldnt get it to work. Anyone knows how to solve this? Thanks, Ohad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] connecting Asterisk with Siemens HiPath4000
Hi All, Can anyone give me some pointers about what is the requirement at both sides? I already have OH323 support in Asterisk, but have no clue how to configure the HiPath. Thanks, Ohad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: HiPath 4000 and Asterisk
Hi,Could you please post your oh323.conf file and explain which changes are required at the HiPath? ( I have the HG3550 Card ) however I have no access to the HiPath system and I need to ask someone else to perform that changes therefore my ability to debug this issue is small.Thanks a lot!Ohad--- Ohad.Levy at infineon.com wrote: I'm trying to setup Asterisk trunk to Siemens HiPath 4000 V2.01i suppose you mean version 2.0 ;-) What would be the best way to do so? I am a bit confused because as far as I've understand this PBX doesn't support H323, but I saw somewhere someone who created a cornet trunk and it worked using H323.I have a HiPath4000 V1.0 interconnected to Asteriskusing a STMI board (HG3550) and oh323. Theinteroperability works well. The chan_cornet AFAIK isnot released by Steffen. The interconnection betweenH4kV2.0 and * is identical, use a HG3550 V2.0 for theH4k and oh323 for *. I've read some information about the cornet connectivity which is in development - does anyone knows the status of that?AFAIK not released :-( ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] capi dial in/out configuration
Hi
all,
I've
recentrly starting to play around with *, when all I wanted is to configure an
fritz ISDN card with [EMAIL PROTECTED]
Currently
I'm stuck at the phase of what do I do with capi after everything is installed.
I'm
trying to understand how to setup incoming and outgoing calls at [EMAIL PROTECTED] since I'm
getting a bit lost with the default dial plan.
It
seems that * answers but disconnect it directly, and I'm unable to setup
outgoing calls.
I
know this is a very general question, but if anyone could give me some pointers
about how to setup capi dial plan, and explain some terms like msn in the
capi.conf file.
My
capi.conf
[EMAIL PROTECTED] asterisk]# cat
capi.conf |grep -v ';'
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=50
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=demo
devices=2
I've
added these two lines the extensions_custom:
s,1,Dial,CAPI/@50:b${EXTEN}|30
always early B3
s,1,Dial,CAPI/@50:${EXTEN}|30|r
no early B3, fake ring indication
when
dialing out I get:
--
Executing Macro(SIP/200-3b6b, dialout-trunk|1|999) in
new stack
-- Executing
GotoIf(SIP/200-3b6b, fooOhad?4) in new stack
-- Executing
SetCallerID(SIP/200-3b6b, Ohad Levy) in new stack
-- Executing
Goto(SIP/200-3b6b, 6) in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing
SetGroup(SIP/200-3b6b, OUT_1) in new stack
-- Executing
CheckGroup(SIP/200-3b6b, ) in new stack
-- Executing
SetVar(SIP/200-3b6b, DIAL_NUMBER=999) in new stack
-- Executing
SetVar(SIP/200-3b6b, DIAL_TRUNK=1) in new stack
-- Executing
AGI(SIP/200-3b6b, fixlocalprefix) in new stack
-- Launched
AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script
fixlocalprefix completed, returning 0
-- Executing
Dial(SIP/200-3b6b, /999) in new stack
== Everyone is busy/congested
at this time
-- Executing
NoOp(SIP/200-3b6b, dial failed) in new stack
-- Executing
Macro(SIP/200-3b6b, outisbusy) in new stack
-- Executing
Playback(SIP/200-3b6b, allison7/all-circuits-busy-now) in
new stack
-- Playing 'allison7/all-circuits-busy-now'
(language 'en')
== Spawn extension (macro-outisbusy,
s, 1) exited non-zero on 'SIP/200-3b6b' in macro 'outisbusy'
== Spawn extension (from-internal,
, 2) exited non-zero on 'SIP/200-3b6b'
-- Executing
Macro(SIP/200-3b6b, hangupcall) in new stack
-- Executing
ResetCDR(SIP/200-3b6b, w) in new stack == Starting CAPI[contr1/8856224]/0
at demo,8856224,1 failed so falling back to exten 's'
== Starting CAPI[contr1/8856224]/0
at demo,s,1 still failed so falling back to context 'default'
-- Executing
Playback(CAPI[contr1/8856224]/0, vm-goodbye) in new
stack
-- started
pbx on channel (callgroup=0)!
-- Playing 'vm-goodbye'
(language 'en')
-- Executing
Macro(CAPI[contr1/8856224]/0, hangupcall) in new stack
-- Executing
ResetCDR(CAPI[contr1/8856224]/0, w) in new stack
-- Executing
NoCDR(CAPI[contr1/8856224]/0, ) in new stack
-- Executing
Wait(CAPI[contr1/8856224]/0, 5) in new stack
-- Executing
Hangup(CAPI[contr1/8856224]/0, ) in new stack
== Spawn extension (macro-hangupcall,
s, 4) exited non-zero on 'CAPI[contr1/8856224]/0' in macro 'hangupcall'
== Spawn extension (default,
s, 2) exited non-zero on 'CAPI[contr1/8856224]/0'
-- Executing
NoCDR(SIP/200-3b6b, ) in new stack
-- Executing
Wait(SIP/200-3b6b, 5) in new stack
== Spawn extension (macro-hangupcall,
s, 3) exited non-zero on 'SIP/200-3b6b' in macro 'hangupcall'
== Spawn extension (from-internal,
h, 1) exited non-zero on 'SIP/200-3b6b'
When
receiving a call:
==
Starting CAPI[contr1/myisdn#]/0 at demo, myisdn#,1 failed so
falling back to exten 's'
== Starting CAPI[contr1/myisdn#]/0
at demo,s,1 still failed so falling back to context 'default'
-- Executing
Playback(CAPI[contr1/myisdn#]/0, vm-goodbye) in
new stack
-- started
pbx on channel (callgroup=0)!
-- Playing 'vm-goodbye'
(language 'en')
-- Executing
Macro(CAPI[contr1/myisdn#]/0, hangupcall) in
new stack
-- Executing
ResetCDR(CAPI[contr1/myisdn#]/0, w) in new
stack
-- Executing
NoCDR(CAPI[contr1/myisdn#]/0, ) in new stack
-- Executing
Wait(CAPI[contr1/myisdn#]/0, 5) in new stack
-- Executing
Hangup(CAPI[contr1/myisdn#]/0, ) in new stack
== Spawn extension (macro-hangupcall,
s, 4) exited non-zero on 'CAPI[contr1/myisdn#]/0' in macro 'hangupcall'
== Spawn extension (default,
s, 2) exited non-zero on 'CAPI[contr1/myisdn#]/0'
Thanks
a lot,
Ohad
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HiPath 4000 and Asterisk
Hi all, I'm trying to setup Asterisk trunk to Siemens HiPath 4000 V2.01 What would be the best way to do so? I am a bit confused because as far as I've understand this PBX doesnt support H323, but I saw somewhere someone who created a cornet trunk and it worked using H323. So if anyone knows what I need to configure I would appreciate it. I've read some information about the cornet connectivity which is in development - does anyone knows the status of that? Thanks! Ohad ___ 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
