RE: [Asterisk-Users] IAX2 one side loses audio
I am afraid i do not have a solution for you, but we also had this problem occur, exactly the same. It happened overnight, with no changes to the server. With help from our IAX provider, we did many tests, no solution, we then moved to a SIP connection to our provider, problem solved. Our * server is a beast with 2GB DDR ram and no load, QOS, 2MB leased line.. Ask your provider if you can try SIP with them. From: [EMAIL PROTECTED] on behalf of Trevor Peirce Sent: Sun 16/01/2005 00:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IAX2 one side loses audio It seems to never fail - after 3 to 5 minutes SIP - IAX calls drop audio on one side. I place a call out through voipjet, and call quality is flawless. However a few minutes later the person who I'm talking to can no longer hear me. I can still hear them. What should I look for to resolve this? Has anyone else had this problem? Using last night's CVS this problem still exists. Thanks, Trevor Peirce ___ 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 winmail.dat___ 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] MeetMe does not compile with Asterisk
Two Asterisk machines, different CVS, both say no application MeetMe, show application does not show MeetMe, when I browse to /asterisk/apps/ I notice that it is the only app that has not installed? Do I need to install ZAPRTC first then try to install the MeetMe application? I do not have a timing device, would this be the reason why meetme does not install when installing asterisk? Any ideas. ___ 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] OT: SIP Aware Firewall with Asterisk
We are on the lookout for a Firewall which is SIP aware, to pass the voice stream to Asterisk. We have looked at the Ingate Products, but they are very expensive. Can anyone point us to a well priced Enterprise SIP aware Firewall? SIP Phones - Firewall - Asterisk Thanks ___ 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] Call Monitor Fails after Transfer
I have a problem with incoming calls being recorded after a supervised transfer. Incoming is CAPI BRI - Asterisk - Supervised Transfer - SIP. Call comes in, receptionist answers, caller put on hold, Asterisk Monitor is recording, caller is on Hold, Callee picks up the call, Asterisk Monitor Stops. All recorded calls are named CallerID to Exten. Receptionist sees the incoming PSTN callerID, yet when we get a transfer from the receptionist, we see her SIP callerID, not the incoming callerID from the PSTN? Which rules out, putting a Monitor line into our macro-stdexten, it will record, but the filename will be local SIP CallerID's, and we end up with two files for the one call. We use Cisco 79XX. Is there a way to continue the same recording after a transfer? Is there a way to pass on the Incoming callerID from PSTN to the SIP phones that have the call transferred to? TA. ___ 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] CallerID after Supervised Transfer
Is there a way to keep the incoming CallerID from the PSTN and pass it onto the sip phone receiving the supervised call transfer? The receptionist receives the PSTN callerID, performs a supervised transfer, we get her local SIP callerID, not the original callers. The main reason we would like the true callerID is for asterisk monitor to name the file correctly for call records. Is this possible with Asterisk? ___ 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] CAPI, BRI and grouping B channels
Hi, I have the exact same problem, we have two Eicon DIVA Cards (BRI UK), using chan_capi by Junghann. The cards have been tested and work perfectly, if we make two outgoing calls simultaneously, and someone calls us, they get a busy tone or call failed, yet capi info says 2 channels are still free??? My capi.conf says to use 2 controllers etc yet Chan_capi does not seem to work correctly. There is basically no information on the website on how to configure for the above or no examples on how to use the capi features. We tried contact via email - no response. So decided to move over to using Cisco Routers and Bri modules, which actually works as it should. Using Callgroup doesn't make any difference: [interfaces] msn=123456789 incomingmsn=* controller=1,2 softdtmf=1 accountcode= context=incoming ;echosquelch=1 ;echocancel=yes ;echotail=64 callgroup=1 ;deflect=12345678 devices=2 In sip.conf [**] type=friend username=** secret=20 canreinvite=no host=dynamic context=test callgroup=1 pickupgroup=1 Maybe that may help. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Smith Sent: 08 December 2004 23:57 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] CAPI, BRI and grouping B channels Dear All, I have a working asterisk installation in the UK on BRI point-to-point. I am using Redhat8 with one Eicon Diva Server 2.0 card with chan_capi-0.3.5 and Asterisk 1.0.1. I have got to the stage where I can make and receive calls over ISDN. My question: How do I group the 2 B channels so that when one channel is in use, the other channel is availble to receive[make] an incoming[outgoing] call ? At present, when only one channel is used, any attempt to dial in from outside is met with a busy tone. I think the 'group' directive is what I am looking for, but I don't know if it can be used in /etc/asterisk/capi.conf or even if that is the correct file to place it? modem.conf ? extensions.conf ? Any help gratefully received, and thank you to all for this excellent software. John. ___ Win a castle for NYE with your mates and Yahoo! Messenger http://uk.messenger.yahoo.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] CAPI, BRI and grouping B channels
Thanks for the info, unfortunately that still doesn't work for me. Making two outgoing using ISDN. Contr1: 2 B channels total, 0 B channels free. Contr2: 2 B channels total, 2 B channels free. *CLI capi info Contr1: 2 B channels total, 0 B channels free. Contr2: 2 B channels total, 2 B channels free. If I try to dial incoming, I still get call failed on my Mobile phone. When CAPI should push the call to the other card which has two channels free. What is your hunt group settings on your DIVA card? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Smith Sent: 09 December 2004 10:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] CAPI, BRI and grouping B channels Hi All, I have just this minute found a solution that works for us. The problem is not with the asterisk configuration but with the configuration of the Eicon Card. I use the Eicon-supplied http server on port 10005 to configure the Eicon card. On the hardware configuration page, set: CAPI Call distribution = Standard behaviour After this change, I could have two simultaneous channels open for calls initiated in either direction. Notes on my setup: hardware IBM PC 500Mhz Pentium III, 64MB ram Digium X100P PCI card Eicon Diva Server 2.0 PCI card software RedHat 8.0 Asterisk 1.0.1 Eicon Linux driver package v 7.5 Chan_capi-0.3.5 Line British Telecom ISDN2e line, configured point-to-point, 6-digit presentation, and no ddi numbers. Best wishes, John. --- Craig Waddington [EMAIL PROTECTED] wrote: Hi, I have the exact same problem, we have two Eicon DIVA Cards (BRI UK), using chan_capi by Junghann. The cards have been tested and work perfectly, if we make two outgoing calls simultaneously, and someone calls us, they get a busy tone or call failed, yet capi info says 2 channels are still free??? My capi.conf says to use 2 controllers etc yet Chan_capi does not seem to work correctly. There is basically no information on the website on how to configure for the above or no examples on how to use the capi features. We tried contact via email - no response. So decided to move over to using Cisco Routers and Bri modules, which actually works as it should. Using Callgroup doesn't make any difference: [interfaces] msn=123456789 incomingmsn=* controller=1,2 softdtmf=1 accountcode= context=incoming ;echosquelch=1 ;echocancel=yes ;echotail=64 callgroup=1 ;deflect=12345678 devices=2 In sip.conf [**] type=friend username=** secret=20 canreinvite=no host=dynamic context=test callgroup=1 pickupgroup=1 Maybe that may help. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Smith Sent: 08 December 2004 23:57 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] CAPI, BRI and grouping B channels Dear All, I have a working asterisk installation in the UK on BRI point-to-point. I am using Redhat8 with one Eicon Diva Server 2.0 card with chan_capi-0.3.5 and Asterisk 1.0.1. I have got to the stage where I can make and receive calls over ISDN. My question: How do I group the 2 B channels so that when one channel is in use, the other channel is availble to receive[make] an incoming[outgoing] call ? At present, when only one channel is used, any attempt to dial in from outside is met with a busy tone. I think the 'group' directive is what I am looking for, but I don't know if it can be used in /etc/asterisk/capi.conf or even if that is the correct file to place it? modem.conf ? extensions.conf ? Any help gratefully received, and thank you to all for this excellent software. John. ___ Win a castle for NYE with your mates and Yahoo! Messenger http://uk.messenger.yahoo.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 ___ Win a castle for NYE with your mates and Yahoo! Messenger http://uk.messenger.yahoo.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
RE: [Asterisk-Users] A waning console error
Try this: http://lists.digium.com/pipermail/asterisk-users/2004-March/039819.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ismaelg Sent: 09 December 2004 12:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] A waning console error Hello, I am getting this kind of Warning in the Asterisk console, but i don't know why. WARNING[8200]: chan_sip.c:681 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Could you give some clue to solve this problem? Thanks in advice. Ismael. ___ 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 Monitor after Call Transfer failing to record the call
I have a problem with incoming calls being recorded after a supervised transfer. Call comes in, receptionist answers, caller put on hold, Asterisk Monitor is recording, caller is on Hold, Callee picks up the call, Asterisk Monitor Stops. All recorded calls are named CallerID to Exten. Receptionist sees the incoming PSTN callerID, yet when we get a transfer from the receptionist, we see her SIP callerID, not the incoming callerID from the PSTN? Which rules out, putting a Monitor line into our macro-stdexten, it will record, but the filename will be local SIP CallerID's, and we end up with two files for the one call. We use Cisco 79XX. Incoming is CAPI BRI - Asterisk - Supervised Transfer - SIP. Is there a way to continue the same recording after a transfer? Is there a way to pass on the Incoming callerID to the SIP phones that have the call transferred to? ___ 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] Firewall traversal anomalies - AJA
It's the RTP Stream Asterisk by default uses ports UDP 10,000 to 20,000 RTP = Audio Open them on your firewall. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Aken Sent: 07 December 2004 15:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Firewall traversal anomalies - AJA I'm trying to setup a Cisco ATA 186 which has a public IP address but sits behind a firewall and connects to an Asterisk server with a NAT IP address sitting behind a BSD firewall. The Cisco registers with the Asterisk server without any problems, and I can place calls without any problems and the phone on the other end rings correctly. However, I cannot hear anything through the Cisco after the connection is made. Where should I begin looking for the problem? This is the sip.conf entry for the Cisco: [6184341501] callerid=GlobalEyes 6184341501 canreinvite=no context=from-internal dtmfmode=rfc2833 host=dynamic mailbox=x nat=yes port=5060 secret=xxx type=friend username=x allow=all ___ 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 Call Monitor and soxmix error
Asterisk Monitor seems to be working fine. Though the
problem I am having is the two files (in out) muxing.
I added ,m to the string, yet the call records two files
still, and I get the resulting error, at the bottom.
monitor executing ( nice -n 19 soxmix
/var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:48:23-in.gsm
/var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:48:23-out.gsm
/var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:48:23.wav
rm -f
/var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:48:23-*
)
nice: soxmix: No such file or directory
soxmix exists
exten = _8.,2,Monitor(gsm,${CALLFILENAME},m)
Path to soxmix = /usr/bin/soxmix
Asterisk seems to be looking in the wrong place for it?
Is there a command line for soxmix to test muxing two .gsm files ?
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RE: [Asterisk-Users] Asterisk Call Monitor and soxmix error
Hi
I am pretty sure it exists
[EMAIL PROTECTED] asterisk]# whereis nice
nice: /bin/nice
/usr/share/man/man1/nice.1.gz /usr/share/man/man2/nice.2.gz
It seems asterisk cant find soxmix to me,
maybe my config is wrong?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of E. Versaevel
Sent: 01 December 2004 15:18
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
Asterisk Call Monitor and soxmix error
Have you checked if nice
allso exists?
It tries to move the soxmix to the
background
Van:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Craig Waddington
Verzonden: woensdag 1 december
2004 15:56
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users]
Asterisk Call Monitor and soxmix error
Asterisk Monitor seems to be working fine. Though the
problem I am having is the two files (in out) muxing.
I added ,m to the string, yet the call records two files
still, and I get the resulting error, at the bottom.
monitor executing ( nice -n 19 soxmix
/var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:48:23-in.gsm
/var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:48:23-out.gsm
/var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:48:23.wav
rm -f
/var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:48:23-*
)
nice: soxmix: No such file or directory
soxmix exists
exten = _8.,2,Monitor(gsm,${CALLFILENAME},m)
Path to soxmix = /usr/bin/soxmix
Asterisk seems to be looking in the wrong place for it?
Is there a command line for soxmix to test muxing two .gsm files ?
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[Asterisk-Users] Audio Drops out at Random - one way
Have a strange problem. 2 different asterisk servers, running different CVS. One behind Firewall, one not. Cisco 7940 phones. Over the past two weeks, users have had a problem with one way audio, after about 2 minutes into a call, they can hear the other person, but the other person cannot hear them, this happens for about 3-5 seconds, then all is fine again. It doesnt happen on every call, about one in 5. Hardware is good, 2mb Connection, QOS enabled. If it was only one Asterisk server I would be ok, but it happens on two completely different places. I cannot work out what is causing it, can anyone offer anyone offer a solution or a method to track this down. Thanks. ___ 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] Audio Drops out at Random - one way
Sorry forgot to mention this is with IAX2 only, SIP works fine. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Waddington Sent: 29 November 2004 10:46 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Audio Drops out at Random - one way Have a strange problem. 2 different asterisk servers, running different CVS. One behind Firewall, one not. Cisco 7940 phones. Over the past two weeks, users have had a problem with one way audio, after about 2 minutes into a call, they can hear the other person, but the other person cannot hear them, this happens for about 3-5 seconds, then all is fine again. It doesnt happen on every call, about one in 5. Hardware is good, 2mb Connection, QOS enabled. If it was only one Asterisk server I would be ok, but it happens on two completely different places. I cannot work out what is causing it, can anyone offer anyone offer a solution or a method to track this down. Thanks. ___ 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] Audio Drops out at Random - one way
I found this: http://lists.digium.com/pipermail/asterisk-dev/2003-May/000764.html But it is old, and I am sure lots of changes have been made to the source, since then. Where and how do you set absolutetimeout=0, would this help? A test I want to perform is, we make a call, and say nothing for 20 seconds, and see if that's why the audio stream is being dropped. ??? What I am doing currently is running debug IAX2 when users make a call, to try pinpoint the issue, but I don't know what I am looking for in the output. Are you using Cisco Phones? If so, what firmware, that is the only common thing at my end. This install worked fine for months, the audio issue has just started occurring. The quality is perfect, except this loss of Audio for a few seconds. Is your problem purely outgoing? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: 29 November 2004 13:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Audio Drops out at Random - one way Craig Waddington wrote: Have a strange problem. 2 different asterisk servers, running different CVS. One behind Firewall, one not. Cisco 7940 phones. Over the past two weeks, users have had a problem with one way audio, after about 2 minutes into a call, they can hear the other person, but the other person cannot hear them, this happens for about 3-5 seconds, then all is fine again. It doesn't happen on every call, about one in 5. Hardware is good, 2mb Connection, QOS enabled. If it was only one Asterisk server I would be ok, but it happens on two completely different places. I cannot work out what is causing it, can anyone offer anyone offer a solution or a method to track this down. Thanks. --- - I have a similar problem with my IAX connection to my termination provider.. No one seems to be able to help and I have replaced or reinstalled just about every component in the chain except the internet itself and the termination provider.. Have updated Asterisk to 1.0.2, have added a switch to my network (was using a hub), have changes to a different firewall, have setup port mapping through the NAT, have tried different DSL routers and put in a high quality microfilter.. So the only things I think it can be are a) my termination provider (but they service many people and I am sure others would have brought it up if it was a problem), b) Asterisk itself or c) my DSL line or ISP.. Unfortunately these are all hard to check and the debug logging on Asterisk didn't help much when I tried looking at it.. I know this doesn't help much but if you come up with anything please let me know.. Its driving us crazy having calls drop on us especially when talking to customers.. Later.. ___ 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] IAX2 Warnings - chan_iax2.c:1464 attempt_transmit
I am getting quite a few of these warnings lately, and audio is sometimes dropping to one way. Is this some way related? Latency to my IAX provider is minimal, and no major packet loss. Nov 29 15:04:00 WARNING[1095035200]: chan_iax2.c:1464 attempt_transmit: Max retries exceeded to host X.X.X.X on IAX2/marg/2 (type = 6, subclass = 2, ts=120009, seqno=36) Nov 29 15:04:00 WARNING[1095035200]: chan_iax2.c:1464 attempt_transmit: Max retries exceeded to host X.X.X.X on IAX2/marg/2 (type = 6, subclass = 11, ts=120012, seqno=37) Nov 29 15:10:42 WARNING[1089370688]: chan_sip.c:675 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) Nov 29 15:10:46 WARNING[1089370688]: chan_sip.c:675 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Can anyone bring some light to this? Using cisco phones. ___ 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] Random Audio Drop out one side
On say 2 out of 10 calls, when on a call, the Audio at our end will drop for about 5 seconds, we can hear them, they cant hear us. It doesnt happen every call, random, which is making it very hard to trouble shoot, I am guessing it has something to do with RTP stream? Nothing has changed this end, yet this has just started happening. Seems to happen at about 2-3mins into a call. Anyone had this happen to them, any advice on a fix? ___ 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] Call failover and redundancy
Recently our provider had an issue, so we couldnt make
VOIP calls.
We currently have a BRI which we use for incoming calls, at
the moment I have the below in my dialplan, so if our VOIP provider or our
internet drops, the outgoing calls are sent through the ISDN Bri.
The problem is, it takes 30 seconds of trying the IAX
account, before it uses the BRI, is there a timeout I can insert somehow, so if
a call fails on VOIP, a few seconds later it switches to the ISDN outgoing?
My current Extensions.conf
exten = _[68]X,1,Dial(IAX2/user:[EMAIL PROTECTED]/44${EXTEN})
exten = _[68]X,2,Dial(${ISDN1}:${EXTEN})
exten = _[68]X,102,Congestion
exten = _[68]X,103,Busy
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[Asterisk-Users] Cisco 1751-V SIP Gateway for Asterisk
I have a 1751 with a BRI Wic, I would like it to pass
incoming calls to Asterisk.
After spending a lot of time on this, I cannot get it to
work. I can see the incoming call and the callerID, yet the router
doesnt seem to pass the call to asterisk.
In SIP.conf
[213.137.185.150]
context=incoming
type=friend
host=213.137.185.150
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
In extensions.conf incoming context:
123456789 is an example of our phone number.
exten = 123456789,1,Wait(1)
exten = 123456789,2,Dial(SIP/5011,15)
exten = 123456789,3,VoiceMail(u${5011})
exten = 123456789,4,Congestion
exten = 123456789,102,Hangup
Can anyone provide me a working config with BRI and a 1751.
We are in UK.
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[Asterisk-Users] Cisco 1751-V as SIP Gateway for Asterisk
I have a 1751 with a BRI Wic, I would like it to pass
incoming calls to Asterisk.
After spending a lot of time on this, I cannot get it to
work. I can see the incoming call and the callerID, yet the router doesnt
seem to pass the call to asterisk.
In SIP.conf
[213.137.185.150]
context=incoming
type=friend
host=213.137.185.150
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
In extensions.conf incoming context:
123456789 is an example of our phone number.
exten = 123456789,1,Wait(1)
exten = 123456789,2,Dial(SIP/5011,15)
exten = 123456789,3,VoiceMail(u${5011})
exten = 123456789,4,Congestion
exten = 123456789,102,Hangup
Can anyone provide me a working config with BRI and a 1751.
We are in UK.
Does the full number get used?
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[Asterisk-Users] Cisco PIX and Asterisk
I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) Internet PIX (NAT) Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the incoming to work? Has anyone managed to get this to work or am I wasting my time on this? Ta. ___ 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 PIX and Asterisk
Thats Great news. Thanks for the information. What version of the PIX IOS you running? Do you have sip fixup protocol enabled? I have found a workaround, install onDo sip server on a machine behind the PIX. The phones register to that, on the pix port forward to the onDo sip server. But I would much rather get it working without having to do that. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hagler Sent: 25 September 2004 19:59 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Cisco PIX and Asterisk It works fine for me. I have a handful of Cisco 7960s behind a PIX firewall and they register to a Asterisk server outside of the PIX with no trouble at all. I didnt do anything special to the PIX (i.e. no access list entries). The tricks I found to make it work generally apply to any setup where the clients are behind NAT. I also run the tftp server for the phones to get configs inside the firewall, and the SIPDefault.cnf file specifies the proxy address outside of the firewall. In the Cisco phone config I have these NAT settings: nat_enable: 1 ; 0-Disabled (default), 1-Enabled nat_address: ; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060) start_media_port: 16384 ; Start RTP range for media (default - 16384) end_media_port: 32766 ; End RTP range for media (default - 32766) nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled And the sip.conf entry for this peer is: [7000] type=friend nat=yes qualify=yes context= secret= callerid= host=dynamic canreinvite=no dtmfmode=rfc2833 timer_register_expires: 120 Setting the registry timer to 120 seconds causes the phone to send out a packet at least every 2 minutes which will open a UDP xlate on the PIX for the session. Then the trick is to use both nat=yes and qualify=yes so Asterisk chats with the phone pretty often. The interval of OPTIONS or REGISTER messages between Asterisk and phone definitely needs to be shorter than the PIXs UDP xlate timeout or the PIX will close the xlate and you wont be able to pass packets into the phone for an incoming call. Note that you can put a numeric value after qualify= instead of yes to fine-tine the interval at which it sends a OPTIONS message. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Waddington Sent: Saturday, September 25, 2004 8:17 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco PIX and Asterisk I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) Internet PIX (NAT) Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the incoming to work? Has anyone managed to get this to work or am I wasting my time on this? Ta. ___ 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_capi module
Go into modules.conf
Comment out chan_modem.so=yes
Make it look like this:
[global]
chan_capi.so=yes
chan_modem.so=yes
;space here
Hope that helps
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 13 September 2004 21:22
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] chan_capi module
Hi!
I am trying to start Asterisk 1.0-RC1 with chan_capi.
Here the error:
---
Parsing '/etc/asterisk/modem.conf': Found
== Loading modem driver chan_modem_chan_capi.soSep 13 22:14:08
WARNING[1076968064]: loader.c:242 ast_load_resource:
/usr/lib/asterisk/modules/chan_modem_chan_capi.so: cannot open shared
object file: No such file or directorySep 13 22:14:08
ERROR[1076968064]: chan_modem.c:954 load_module: Failed to
load driver chan_modem_chan_capi.so == Unregistered channel type
'Modem'
Sep 13 22:14:08 WARNING[1076968064]: loader.c:328 ast_load_resource:
chan_modem.so: load_module failed, returning -1 == Unregistered channel
type 'Modem'
Sep 13 22:14:08 WARNING[1076968064]: loader.c:374 load_modules: Loading
module chan_modem.so failed!---
I dont have a chan_modem_chan_capi.so module, only a chan_modem.o.
I am using chan_capi from:
http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.5.tar.gz
My modem.conf:
---
[interfaces]
context=remote
driver=chan_capi
stripmsd=0
dialtype=tone
mode=immediate
group=1
Capiinfo:
---
02 (49.18)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
internal controller supported
DTMF supported
Supplementary Services supported
channel allocation supported (leased lines)
B1 protocols support: 0x411f
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
V.110 asynconous operation with start/stop byte framing
V.110 synconous operation with HDLC framing
T.30 modem for fax group 3
Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x0b1b
ISO 7776 (X.75 SLP)
Transparent
LAPD with Q.921 for D channel X.25 (SAPI 16)
T.30 for fax group 3
ISO 7776 (X.75 SLP) with V.42bis compression
V.120 asyncronous mode
V.120 bit-transparent mode
B3 protocols support: 0x80bf
Transparent
T.90NL, T.70NL, T.90
ISO 8208 (X.25 DTE-DTE)
X.25 DCE
T.30 for fax group 3
T.30 for fax group 3 with extensions
Modem
0100
0200
3900
1f010040
1b0b
bf80
0101 0002
Supplementary services support: 0x03ff
Hold / Retrieve
Terminal Portability
ECT
3PTY
Call Forwarding
Call Deflection
MCID
CCBS
Any help/hints/tips would be great!
Thanks!
Mario
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RE: [Asterisk-Users] cisco phone and parked calls
In my sip extensions context I have
include = parkedcalls
In extensions.conf I have
[parkedcalls]
Exten = 2000,1,Answer
In parking.conf I have the same.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe
Antkowiak
Sent: 29 June 2004 22:56
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] cisco phone and parked calls
sent this before, but it bounced back and didn't show up on the list.
If it did get sent, I apologize.
-- Forwarded message --
From: Joe Antkowiak [EMAIL PROTECTED]
Date: Tue, 29 Jun 2004 14:55:25 -0400
Subject: cisco phone and parked calls
To: [EMAIL PROTECTED]
So, I can't figure out how to get the parkandannounce application to
work the way I want it to... I have cisco 7960 IP phones using SIP,
and this is what I have in my extensions.conf:
exten =
700,1,ParkAndAnnounce(pbx-transfer:PARKED|90|SIP/${EXTEN:1}|internal,${E
XTEN:1},1)
exten = 700,2,Hangup
and in my parking.conf:
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls
on
context = parkedcalls ; Which context parked calls are
in
parkingtime = 180
In order for the person parking the call to hear what parked extension
the call is on, they have to do the transfer by pressing # and dialing
700. When the user uses the transfer function on the cisco phone, it
still correctly parks the call, but never tells the person what
extension its parked on.
Also, for some reason, I had to create that 700 extension, it always
complains that it can't find 700 when I don't do that, even though
parkedcalls is included in the internal context...
Any suggestions?
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RE: [Asterisk-Users] Chan_Capi Down
I am also having the same problem. Latest CVS Latest Capi When it does work and you pick up the phone, CAPI disconnects the call. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ePyron Felix Deierlein Sent: 28 June 2004 18:34 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Chan_Capi Down Hi all, * was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a Today chan_capi stopped working, without any changings at the system. It seems, that not * is the reason, because isdn-log also shows no calls. If I try to call * from outside via capi, I only get a busy. That is the try from inside to outside: stern01*CLI -- data = @89930:0107901723168212 -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/2 == CAPI Call CAPI[contr1/89930]/2 -- CONNECT_CONF ID=003 #0x000d LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_IND ID=003 #0x0002 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, 01723168212, 2) exited non-zero on 'SIP/ePfd-7515' -- data = @89930:01079h -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/3 == CAPI Call CAPI[contr1/89930]/3 -- CONNECT_CONF ID=003 #0x000e LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_CONF ID=003 #0x000f LEN=0014 Controller/PLCI/NCCI= 0x Info= 0x2002 -- DISCONNECT_IND ID=003 #0x0003 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, h, 1) exited non-zero on 'SIP/ePfd-7515' dmesg shows: isdn_dc2minor: di(0) ch(-1072539760) invalid capidrv-1: now up (2 B channels) capidrv-1: D2 trace enabled capi: controller 1 up kcapi: notify up contr 2 capidrv: controller 2 up isdn_dc2minor: di(1) ch(-1072539760) invalid capidrv-2: now up (2 B channels) capidrv-2: D2 trace enabled capi: controller 2 up kcapi: notify up contr 3 capidrv: controller 3 up isdn_dc2minor: di(2) ch(-1072539760) invalid capidrv-3: now up (2 B channels) capidrv-3: D2 trace enabled capi: controller 3 up kcapi: notify up contr 4 capidrv: controller 4 up isdn_dc2minor: di(3) ch(-1072539760) invalid capidrv-4: now up (2 B channels) capidrv-4: D2 trace enabled capi: controller 4 up I hope, that you could help me... Thanks Felix Deierlein ___ 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] RE: Chan_Capi Down
Thanks I will give that a try. Looks like this may need a bug report? We are all getting the same errors. Outgoing is fine for me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas Anderson Sent: 28 June 2004 23:26 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE: Chan_Capi Down Same here :-( asterisk show's this error in the same moment i'm trying to pick up an incoming call: Jun 23 13:14:03 ERROR[-1284076624]: chan_capi.c:881 capi_write: dont know how to write subclass 64 This problem starts with cvs update -D 6/21/04 21:00:00 CET If i revert back to cvs update -D 6/21/04 18:00:00 CET the problem is gone. -- original message -- I am also having the same problem. Latest CVS Latest Capi When it does work and you pick up the phone, CAPI disconnects the call. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ePyron Felix Deierlein Sent: 28 June 2004 18:34 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Chan_Capi Down Hi all, * was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a Today chan_capi stopped working, without any changings at the system. It seems, that not * is the reason, because isdn-log also shows no calls. If I try to call * from outside via capi, I only get a busy. That is the try from inside to outside: stern01*CLI -- data = @89930:0107901723168212 -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/2 == CAPI Call CAPI[contr1/89930]/2 -- CONNECT_CONF ID=003 #0x000d LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_IND ID=003 #0x0002 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, 01723168212, 2) exited non-zero on 'SIP/ePfd-7515' -- data = @89930:01079h -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/3 == CAPI Call CAPI[contr1/89930]/3 -- CONNECT_CONF ID=003 #0x000e LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_CONF ID=003 #0x000f LEN=0014 Controller/PLCI/NCCI= 0x Info= 0x2002 -- DISCONNECT_IND ID=003 #0x0003 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, h, 1) exited non-zero on 'SIP/ePfd-7515' dmesg shows: isdn_dc2minor: di(0) ch(-1072539760) invalid capidrv-1: now up (2 B channels) capidrv-1: D2 trace enabled capi: controller 1 up kcapi: notify up contr 2 capidrv: controller 2 up isdn_dc2minor: di(1) ch(-1072539760) invalid capidrv-2: now up (2 B channels) capidrv-2: D2 trace enabled capi: controller 2 up kcapi: notify up contr 3 capidrv: controller 3 up isdn_dc2minor: di(2) ch(-1072539760) invalid capidrv-3: now up (2 B channels) capidrv-3: D2 trace enabled capi: controller 3 up kcapi: notify up contr 4 capidrv: controller 4 up isdn_dc2minor: di(3) ch(-1072539760) invalid capidrv-4: now up (2 B channels) capidrv-4: D2 trace enabled capi: controller 4 up I hope, that you could help me... Thanks Felix Deierlein _ Listen to music online with the Xtra Broadband Channel http://xtra.co.nz/broadband ___ 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] SetVar - bellcode and cisco phone
I am trying to have the ring types different for internal and external incoming calls. I have followed the guide on the wiki, the SetVar executes, in extensions.conf I have it as s,1, Yet it doesnt work? When the phone rings, the ring type is the one I chose on the phone, it rings same tone for both when I test. Using Asterisk Stable. Anyone got this working and can point me in the right direction? Ouput of both internal and external incoming calls. -- Executing Macro(SIP/20-5722, stdexten|SIP/22) in new stack -- Executing SetVar(SIP/20-5722, ALERT_INFO=Bellcode-dr2) in new stack -- Executing Dial(SIP/20-5722, SIP/22|25|tr) in new stack -- Called 22 -- SIP/22-080c is ringing == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'SIP/20-5722' in macro 'stdexten' == Spawn extension (sip, 22, 1) exited non-zero on 'SIP/20-5722' -- Executing SetVar(CAPI[contr1/s]/0, ALERT_INFO=Bellcode-dr5) in new stack -- Executing Dial(CAPI[contr1/s]/0, SIP/22|35|t) in new stack -- Called 22 -- started pbx on channel (callgroup=2)! -- SIP/22-e97c is ringing == Spawn extension (incoming, s, 2) exited non-zero on 'CAPI[contr1/s]/0' -- CAPI Hangingup
[Asterisk-Users] Chan CAPI and Latest CVS wont compile
When I saw the update for Cisco Phone RTP issue I thought I would try it. Unfortunately chan_capi wont compile on this update. Can anyone recommend a good * release for Capi, Bri ISDN and Cisco 7940s SIP 6.3. Or will CHAN_CAPI also be updated ? Running Eicon Diva Bri Cards. Error: chan_capi.c:1187: too many arguments to function ast_dsp_process
RE: [Asterisk-Users] Chan CAPI and Latest CVS wont compile
Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chrétien Wetemans Sent: 22 May 2004 12:19 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Chan CAPI and Latest CVS wont compile http://lists.digium.com/pipermail/asterisk-users/2004-April/044125.html - Original Message - From: Craig Waddington [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 22, 2004 12:24 PM Subject: [Asterisk-Users] Chan CAPI and Latest CVS wont compile When I saw the update for Cisco Phone RTP issue I thought I would try it. Unfortunately chan_capi wont compile on this update. Can anyone recommend a good * release for Capi, Bri ISDN and Cisco 7940's SIP 6.3. Or will CHAN_CAPI also be updated ? Running Eicon Diva Bri Cards. Error: chan_capi.c:1187: too many arguments to function 'ast_dsp_process' ___ 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] SIP in the UK
Voiptalk provide an excellent service and great support. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Potkin Sent: 10 May 2004 23:51 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP in the UK On Mon, May 10, 2004 at 08:58:23AM +0100, Gavin Hamill wrote: http://www.voiptalk.org/ - this is the service-side of TelAppliant, official UK Digium resellers. I've written to VoIPTalk a couple of times and never got any response from them, and their outbound calling rates aren't fantastic. I would be concerned about their quality of customer service were I to be considering using them for business use. The comment on VoIPTalk's calling rates is interesting as I came to a different conclusion. To instance the two main destinations I call, the UK and Spain, as an example, the rates are 1.6p and 2p per minute respectively. This appears to me to be very competitive with other offerings. Brian. ___ 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] CAPI Gain
I am using ISDN with CAPI and Eicon Diva card. On ISDN calls in and out, some people are saying they find it hard to hear us. Its only the odd few though, not everyone. We can hear them no problem. Do I just increase the txgain? What is the limit for txgain, or are there any gotchas for turning it up? If you use the same what are your settings? I have: rxgain=0.4 txgain=1.5 Thanks.
[Asterisk-Users] Help choosing a UK IAX provider
Hi, Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? Ta.
RE: [Asterisk-Users] Help choosing a UK IAX provider
Yes, but, I am talking about this world. Ive got 2mb up/down with qos, just need another (good) provider. If I can try a few and see which is best. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: 21 April 2004 15:32 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Help choosing a UK IAX provider On Wed, Apr 21, 2004 at 03:24:06PM +0100, Craig Waddington wrote: Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? That's the trouble with running VoIP over contended public Internet. Find someone who can offer you connectivity with QoS and then has QoS across their network for VoIP traffic. Or find someone with infinite bandwidth. Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ 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] Help choosing a UK IAX provider
Thanks Tan. I will look into it my end. Unfortunately it isn't happening from just one location, and a variety of phones. The quality used to be perfect, the odd call would be a little jittery/choppy, but now most are like that, I am running asterisk stable with eicon diva cards. 3.0Ghz dell 2GB ram. 1 1 ms 2 ms 1 ms 10.5.0.1 217 ms14 ms14 ms 195.10.119.94 317 ms14 ms14 ms 195.10.119.158 422 ms14 ms15 ms 217.23.160.1 515 ms15 ms31 ms 217.23.162.122 617 ms15 ms14 ms 217.23.160.85 719 ms18 ms14 ms 217.23.160.186 830 ms26 ms29 ms tier1-1.BUD2.psie.net [154.14.68.113] 931 ms39 ms29 ms linx1.teleglobe.net [195.66.224.51] 1026 ms28 ms30 ms if-0-0-0.bb2.London.Teleglobe.net [195.219.96.81 ] 1159 ms87 ms 108 ms ix-3-1-0-822.bb2.London.Teleglobe.net [195.219.2 .34] 1276 ms54 ms54 ms wi2.westloc.com [82.145.32.2] 13 229 ms 239 ms 187 ms wc3-10.westloc.com [82.145.32.73] Trace complete. I don't know if asterisk is reporting this right, but all day on the console I am seeing voiptalk unreachable, then 5 secs later reachable? IAX.conf allow=ulaw allow=alaw jitterbuffer=500 maxexcessbuffer=300 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 21 April 2004 16:01 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Help choosing a UK IAX provider Craig, 2mb up/down with QoS doesn't mean anything, especially when you hit the Internet. What is better is to look at the exact route of your calls and then determine whether maybe there are some other issues. For instance, we had a customer with Ciscos who was reporting choppy audio. However, this was down to a bug in asterisk (http://bugs.digium.com/bug_view_page.php?bug_id=0001374) and cvs updating fixed the problem. Tan Telappliant.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Waddington Sent: 21 April 2004 15:38 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Help choosing a UK IAX provider Yes, but, I am talking about this world. Ive got 2mb up/down with qos, just need another (good) provider. If I can try a few and see which is best. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: 21 April 2004 15:32 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Help choosing a UK IAX provider On Wed, Apr 21, 2004 at 03:24:06PM +0100, Craig Waddington wrote: Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? That's the trouble with running VoIP over contended public Internet. Find someone who can offer you connectivity with QoS and then has QoS across their network for VoIP traffic. Or find someone with infinite bandwidth. Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ 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 ___ 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] Help choosing a UK IAX provider
Hahahhaaa your right there Tan. List, don't get me wrong, voiptalk are very good, service, support, price, I am just having some issues which may be my end. I was just wanting to try some iax providers out to see what worked best for us. Hopefully will get sorted. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 21 April 2004 16:02 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Help choosing a UK IAX provider In the UK, with the sort of equipment that BT has in its network, you're lucky to even get adsl going through! ISPs can only provide QoS up to a certain boundary. After that it is out of their control! Tan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: 21 April 2004 15:44 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Help choosing a UK IAX provider On Wed, 2004-04-21 at 09:32, Steve Kennedy wrote: That's the trouble with running VoIP over contended public Internet. Find someone who can offer you connectivity with QoS and then has QoS across their network for VoIP traffic. LOL! I've not found any providers that offer QoS on their network other than a small regional ISP that put QoS on their network when we waved enough money at them. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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] Cisco 7940 no audio
Title: Message Thanks for the help, I am currently running the latest sip image, it seems to have fixed a lot of bugs.. I did a full rebuild of the server and used the stable cvs, all is working perfectly now. I am actually amazed at the quality of the call using the diva card/capi through isdn. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of AstGrp Sent: 18 April 2004 04:44 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7940 no audio Try upgrading to SIP 6.3. I heard from someone on the IRC Channel about this problem and 6.3 resolved it -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Waddington Posted At: Friday, April 16, 2004 1:04 PM Posted To: Asterisk User Group Conversation: Cisco 7940 no audio Subject: [Asterisk-Users] Cisco 7940 no audio When we receive or make a call to the outside they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi - Asterisk network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this problem, and has a fix? Thanks.
[Asterisk-Users] Capi MSN routing.
Kudos to the CAPI developers. I would like to have multiple MSNs on my ISDN Bri lines. I see all the cool features but cannot find any examples or guides to build from. Currently running Diva Eicon Cards with CAPI from http://www.junghanns.net I would like to route calls to sip phones via msn. Set up callgroups etc. Can anyone share some some examples I can build from. I want to use some of the features the capi drivers support.
[Asterisk-Users] Cisco 7940 no audio
When we receive or make a call to the outside they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi - Asterisk network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this problem, and has a fix? Thanks.
RE: [Asterisk-Users] Cisco 7940 no audio
I will try disallow=all, thanks, Nat is off. Sip.conf below. If I call from my mobile to the isdn number, the Cisco phone rings, I pick up, they can hear me, I can not hear them, if I transfer it to the Sipura I get AUDIO!! It is also happening over IAX with the Cisco phones. I followed a lot of the examples on loligo.com, which were a great help, but this is so hard to troubleshoot as I cannot see any errors in debug, asterisk thinks a good call is in progress. Anything internal is perfect. The CAPI works fine. Its just the audio from the other end. Every now and then I can hear a quick bit of sound. One in 20 calls may work. [general] port=5060 ; Port to bind to bindaddr=0.0.0.0; Address to bind to allow=ulaw allow=alaw tos=lowdelay [20] type=friend username=20 secret=20 canreinvite=no host=dynamic mailbox=20 callerid=Cisco Phone 20 accountcode=20 qualify=yes context=sip Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Cuthie Sent: 16 April 2004 18:37 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 no audio Craig Waddington wrote: When we receive or make a call to the outside - they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi - Asterisk à network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this problem, and has a fix? Thanks. Make sure you don't have the Cisco phone set to do NAT. -brian ___ 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 7940 no audio
Yes MOH etc work fine for the receiving end, dialing from outside. I have run X-lite and GS phones on the network on a test machine before this one, and it worked great. Though I haven't had a chance to see if they work or not. I will definatley check my Firewall logs, that's a good point, but the sipura works. It seems codec to me, but I have tried many different confs in sip. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Laird Sent: 16 April 2004 18:32 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 no audio On Apr 16, 2004, at 10:04 AM, Craig Waddington wrote: When we receive or make a call to the outside - they can hear us, but we cant hear them. With SIP, missing audio is *usually* either a firewall or NAT issue. Check firewall logs and make sure that you aren't seeing packets being lost. Do you have more then one 7940? If so, can they call each other? Also, when people call into your system, do they get audio from asterisk? Does voicemail work? Scott ___ 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 7940 no audio - sip debug
Of Tracy R Reed Sent: 16 April 2004 19:20 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 no audio On Fri, Apr 16, 2004 at 06:04:04PM +0100, Craig Waddington spake thusly: When we receive or make a call to the outside - they can hear us, but we cant hear them. I have had this problem several times and so far no resolution. However for me it has always been with IAX. I have been told that IAX is supposed to be NAT-safe but that does not seem to be the case for me. For example: SIP (grandstream, snom) -Asterisk-NAT-Asterisk-SIP (grandstream, snom) He can hear me but I can't hear him. In another case I had: IAXclient (soft phone)-NAT-Asterisk-Snom And I could hear him but he could not hear me. Same phone system and settings as above. However as soon as I switched the first users phone to talk directly to my Asterisk box with SIP it worked perfectly. And when I switched the user in the second case to a SIP based soft phone it also worked just fine. SIP has worked better through NAT than IAX (with nat=yes in sip.conf) which is bizarre and contrary to what I have read where IAX should be NAT-safe and SIP not. I have dreams of a world fully converted to IPv6 where NAT no longer exists. Alas, it is but a dream. -- Tracy Reed The attachment is a digital signature. http://copilotconsulting.com More info: http://copilotconsulting.com/sig ___ 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] IPC5000 - Wireless Sip phone
I am looking to buy a wireless sip phone, probably the IPC5000, I have looked at Wisip phone and read tons of posts regarding that phone. Do any * admins have any feedback on this phone? Is there any major differences between the phones, besides looks? The site has very limited information regarding prices etc. Ta.
RE: [Asterisk-Users] IPC5000 - Wireless Sip phone
Thanks for the info. Sounds good. Does that mean I can contact them for a test unit also, to try before I buy? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Devenijn Sent: 11 March 2004 18:25 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] IPC5000 - Wireless Sip phone I ordered a test unit and will recieve it this week (already shipped from sweden), i will post some comments on this list when it is tested .. I hope it will do his job !! ... the mail they sent to : Hello Michael, Hope you are well. Your sample is on the way and pls find attached delivery note for your reference. Ps. frieght charge was USD10 lower, so we own you USD10 that we will pretty reduced it with your next order or we transfer it to your bank account. I'll the coming days send you updated information about the handset and its new design i.e. it has L2 roaming feature now. The handoff time is 200 ~ 300ms between the AP. We aim to short it to 100 ~ 200ms. The implementation of Web Authentication(web-login) what we call HTTPS(SSL)is ongoing and should be releasedon June. It can be software upgrade. Best Regards, Mohammed Fahd -Oorspronkelijk bericht- Van: [EMAIL PROTECTED]namensCraig Waddington Verzonden: do 11/03/2004 19:15 Aan: [EMAIL PROTECTED] CC: Onderwerp: [Asterisk-Users] IPC5000 - Wireless Sip phone I am looking to buy a wireless sip phone, probably the IPC5000, I have looked at Wisip phone and read tons of posts regarding that phone. Do any * admins have any feedback on this phone? Is there any major differences between the phones, besides looks? The site has very limited information regarding prices etc. Ta.
RE: [Asterisk-Users] Ahead SIPPS and Asterisk
What sort of phone are you trying to call? I use SIPPS and * and it works fine, it just wont work when you call Windows Messenger for some reason. I can call X-lite, POTS, GS phones no problems. I use the same config as X-lite, in SIPPS if you click on the spanner or press F9 to go into configuration and click on network. You should see aquired? Click modify and ensure Gateway is ticked and you have the IP of your server in there, for some reason it defaults as redirect. On mine I leave Dial prefix blank. Click modify again then Ok, which will take you to the config section for the user, choose use authentication, fill in the blanks, I found also putting the realm in made it work correctly. Hope that helps in some way. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlton J. O'Riley Sent: 05 March 2004 15:17 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Ahead SIPPS and Asterisk Has anyone gotten Ahead's SIPPS softphone to work with Asterisk? I get it to register, but when I dial into a number as soon as voice is to be connected I get a Warning 399 SDP body missing message and then a BYE disconnecting the call. The setup I have works great with Xten's x-pro, but can't get it to work with SIPPS. Any hints? ___ 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] Ahead SIPPS and Asterisk
Your problem is what I experience with Messenger, when I call it. Unfortunately I never bothered trying to work out the problem. I like the SIPPS phone features, but it is ugly. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlton J. O'Riley Sent: 05 March 2004 18:48 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Ahead SIPPS and Asterisk I'll try to look over my config again. Not sure I put a realm in, but everything else seemed fine. I get the acquired message and I see the SIP messages flowing on the Asterisk server, but once any sound needs to be sent, it dies. I'm using g711ulaw and I was calling into an announcement menu to test that I have setup on the server. The entry in my sip.conf is exactly what I use for X-Pro, so that is why I am confused by the missing SDP message. -Original Message- Message: 8 Subject: RE: [Asterisk-Users] Ahead SIPPS and Asterisk Date: Fri, 5 Mar 2004 18:25:05 - From: Craig Waddington [EMAIL PROTECTED] To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] What sort of phone are you trying to call? I use SIPPS and * and it works fine, it just wont work when you call Windows Messenger for some reason. I can call X-lite, POTS, GS phones no problems. I use the same config as X-lite, in SIPPS if you click on the spanner or press F9 to go into configuration and click on network. You should see aquired?=20 Click modify and ensure Gateway is ticked and you have the IP of your server in there, for some reason it defaults as redirect. On mine I leave Dial prefix blank. Click modify again then Ok, which will take you to the config section for the user, choose use authentication, fill in the blanks, I found also putting the realm in made it work correctly. Hope that helps in some way. ___ 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] Simple * status
Nice one thanks for sharing, I look forward to it. This will be very handy for SIP call transfers. At the moment I blindly transfer on sip. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer Sent: 05 March 2004 19:49 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Simple * status On Fri, Mar 05, 2004 at 08:27:59PM +0100, [EMAIL PROTECTED] wrote: Tim, It looks interesting.. Are you willing to release the source code? Sure. let me clean it up a bit... OK, a LOT... and finish the comments, and I'll have a download link for it sometime this weekend. I'll keep the downloadable stuff up-to-date with the running version. Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ 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] New Wiki page: Dimensioning an Asterisk system
Title: Re: [Asterisk-Users] New Wiki page: Dimensioning an Asterisk system Is the article correct in saying: g729 codecs licenses can be purchased for Asterisk (not for SCSI systems!) I thought people had this working on SCSI now? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cheesman Sent: 23 February 2004 04:48 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] New Wiki page: Dimensioning an Asterisk system Hi all Sorry for the last post! Not enough sleep combined with inattention caused me to reply to the wrong message. Sean -Original Message- From: Anton Tinchev [mailto:[EMAIL PROTECTED] Sent: Mon 2/23/2004 12:25 AM To: [EMAIL PROTECTED] Cc: Subject: Re: [Asterisk-Users] New Wiki page: Dimensioning an Asterisk system hmm, this pages must be fixed. Looks terrible on all NGlayout based browsers Philipp von Klitzing wrote: Hi there, please comment and adjust or enhance as you find appropriate: http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning Typical questions asked on the mailing asterisk-users are: How fast/big must my machine be in order to serve my needs? How many simultaneous calls can Asterisk handle? Unfortunately there are no simple answers. You'll need work through the following checklist to at least get nearer to an answer or be able to post a meaningful question to asterisk-users: [...] Cheers, Philipp ___ 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] Wifi Phones
Those phones look good, but, only have 10 milliwatt output. Have you looked at these: http://www.spectralink.com/products/nl-wts.html 100mw output. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Miguel Cavazos Sent: 15 February 2004 12:39 To: Asterisk Users Subject: [Asterisk-Users] Wifi Phones Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details you can find it here http://mike.calle69.net/IPC5000-brochure.PDF, and its almost the same price as Wisip. But when I ask if this phone will work with asterisk I got this answer We didn't tested on Asteriskt but on IX66, hotsip, Cisoc etc... However, The IPC5000 should work on other SIP platform without any problem as it is standard based. I just dont want to spend 290 USD for a phone that wont work and that no one seems to use here. So I would like to know if anyone of you guys had try out this model or seen it working, sorry about the unnesesary traffic to the list, my question is simple would this work against asterisk if anyone knows any other Wifi phones besides Wisip and Ciscos expensive toy please tell me. Miguel Cavazos ___ 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] fwd settings
SIP.CONF
[general]
; Codecs your choice
disallow=all
;allow=gsm
allow=ulaw
allow=alaw
;allow=ilbc
;allow=spx
allow=g723
allow=g729
register=1234:[EMAIL PROTECTED]/5000
[fwd.pulver.com]
type=friend
secret=password
username=1234
host=fwd.pulver.com
context=sip
nat=yes
;ext for free world dial up
fromuser=1234
fromdomain=fwd.pulver.com
reinvite=no
canreinvite=no
EXTENSIONS.CONF
[globals]
FWDPHONE=SIP/5000
FWDUSERID=1234
FWDPASSWORD=password
FWDUSERNAME=CalleID Name
[default]
; context which is in zapata.conf
include = fwd-out
[sip]
exten = 5000,1,Dial(${FWDPHONE},30,t)
exten = 5000,2,Hangup
[fwd-out]
exten =
_7.,1,SetCallerID(${FWDUSERID})
exten =
_7.,2,SetCIDName(${FWDUSERNAME})
exten =
_7.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _7.,4,Playback(invalid)
exten = _7.,5,Hangup
www.ntfs.org
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Francisco Perez-Landaeta
Sent: 05 February 2004 19:20
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] fwd
settings
Hi, i finally was able to getdialtone on my fxs board.
!! however, i think i am missing something in the fwd setting to make work my
account.
i am getting an error authenticating my account
could someone send me the exact settings to put on sip.conf
? to make it work ?
i have my own account, password but i am getting it wrong.
thanks,
Francisco
RE: [Asterisk-Users] talking clock
You can add: ; Say Current Date and Time;exten = 13,1,DateTime()exten = 13,2,Wait(1)exten = 13,3,DateTime()exten = 13,4,Hangup into exten. maybe that helps http//www.ntfs.org From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deepakumar JV Sent: 04 February 2004 14:20 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] talking clock Hello I am looking for a AGI application that can say the current time with seconds, but i don't need the day/year. Has anyone got this already? Thanks in advance Deepak
RE: [Asterisk-Users] Music on Hold Warnings
Tilghman Thanks for the help. You were spot on, yup the bitrate was screwed. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread: Request to schedule in the past?!?! And the machine does seem to be heavily underload - Asterisk = 100% CPU. MOH is working great now. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: 30 January 2004 16:39 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Music on Hold Warnings On Friday 30 January 2004 04:33, Craig Waddington wrote: 1.Warning, flexibel rate not heavily tested! You're using variable rate mp3's. If you want to avoid the error, recode your mp3s to a static rate. 2. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread: Request to schedule in the past?!?! Is your machine heavily loaded? This could indicate that a thread was unable to complete a task because it was interrupted and did not resume for a fairly long time (as processor time goes). It could also indicate clock drift (sync your time with NTP servers more often). -Tilghman ___ 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] Music on Hold Warnings
Hi. I am having the following warning when using music on hold. It works from X-Lite to Grandstream. I get a lot of errors and warnings. 1.Warning, flexibel rate not heavily tested! 2. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread: Request to schedule in the past?!?! Thanks for any help. Full Output below: Jan 30 10:24:55 WARNING[1133718080]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 909 (Response) Jan 30 10:24:55 WARNING[1217602880]: file.c:521 ast_readaudio_callback: Failed to write frame == Spawn extension (sip, 5001, 2) exited non-zero on 'SIP/5002-0922' -- SIP/5001-6a4d answered SIP/5002-d365 -- Attempting native bridge of SIP/5002-d365 and SIP/5001-6a4d -- Started music on hold, class 'default', on SIP/5001-6a4d Warning, flexibel rate not heavily tested! Jan 30 10:25:14 NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread: Request to schedule in the past?!?! -- Stopped music on hold on SIP/5001-6a4d == Spawn extension (sip, 5001, 1) exited non-zero on 'SIP/5002-d365' -- Executing Dial(SIP/5002-a28b, SIP/5001|20) in new stack -- Called 5001 -- SIP/5001-87f7 is ringing -- SIP/5001-87f7 answered SIP/5002-a28b -- Attempting native bridge of SIP/5002-a28b and SIP/5001-87f7 -- Started music on hold, class 'default', on SIP/5002-a28b Warning, flexibel rate not heavily tested! Jan 30 10:26:40 NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread: Request to schedule in the past?!?! Jan 30 10:26:50 NOTICE[1234379840]: rtp.c:264 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible -- Stopped music on hold on SIP/5002-a28b == Spawn extension (sip, 5001, 1) exited non-zero on 'SIP/5002-a28b'
RE: [Asterisk-Users] 100% of cpu in an out of the box *
Me too :( 100% CPU. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of F.G.Testa Sent: 14 January 2004 20:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 100% of cpu in an out of the box * Hi all! I'm newbie, so here goes my situation: I have succefully compiled the cvs version as shown in asterisk website in some linux distros: Debian (2.4.22), Conectiva, Fedora Core 1 and in all of them, * starts and consumes all the cpu (on top). Does anybody know this issue? Thanks! Testa ___ 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] FS/OS Telephony Summit 2004
Hi I am attending the tutorial day, i am looking forward to it. See you there. Craig. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus-Peter Junghanns Sent: 13 January 2004 10:31 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: [Asterisk-Users] FS/OS Telephony Summit 2004 Hello * world, i will be attending the FS/OS Telephony Summit 2004 in Geilenkirchen from the 16th til 20th january. Together with Christian Richter i will be speaking about * on monday. And we will give an * tutorial on tuesday. I will be presenting some ISDN stuff there, including the quadBRI cards. If you will be there too and want to meet, just let me know. :) Details on the summit can be found at: http://www.guug.de/veranstaltungen/telephony-summit-2004/ best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ 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] Asterisk behind NAT How to do it.
Balaji. I just left rtf.conf at default. Though I guess it wouldn't hurt to change it to test. Does it currently work for you with the settings I provided? Craig. www.ntfs.org -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Balaji NJL Sent: 11 January 2004 10:35 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk behind NAT How to do it. i like the idea of not requiring to open 1 ports in the firewall. Do i need to change rtf.conf to from 1 - 2 to 16384 and 16394. thanks, -B - Original Message - From: Craig Waddington [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, December 27, 2003 3:43 AM Subject: RE: [Asterisk-Users] Asterisk behind NAT How to do it. Hi I have SIP working on NAT using X-lite phones. On my Cisco 827H ADSL router I forwarded ports 5060, 16394, 16384 to my * (10.1.0.0). 16394,16384 being RTP. In X-lite set the RTP port to use 16394 instead of the default 8000. Works great over the internet. Didn't need patches or anything else. I hope that helps you. -C www.ntfs.org -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Balaji NJL Sent: 27 December 2003 08:34 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk behind NAT How to do it. Hi All, i tried to apply this patch and i got the following error. The chan_sip.c version i hv is 1.265 hv any one tried this patch on this latest chan_sip version. thanks, -B chan_sip.o: In function `load_module': chan_sip.o(.text+0x15ebf): undefined reference to `ast_rtp_proto_register' chan_sip.o(.text+0x15ee0): undefined reference to `ast_register_application' chan_sip.o: In function `delete_users': chan_sip.o(.text+0x15fc1): undefined reference to `ast_free_ha' chan_sip.o(.text+0x1604d): undefined reference to `ast_sched_del' chan_sip.o: In function `prune_peers': chan_sip.o(.text+0x16167): undefined reference to `ast_sched_del' chan_sip.o(.text+0x1618d): undefined reference to `ast_sched_del' chan_sip.o: In function `unload_module': chan_sip.o(.text+0x162bd): undefined reference to `ast_channel_unregister' chan_sip.o(.text+0x162ce): undefined reference to `ast_unregister_application' chan_sip.o(.text+0x16337): undefined reference to `ast_softhangup' chan_sip.o(.text+0x1636c): undefined reference to `ast_log' chan_sip.o(.text+0x163ab): undefined reference to `pthread_cancel' chan_sip.o(.text+0x163be): undefined reference to `pthread_kill' chan_sip.o(.text+0x163d1): undefined reference to `pthread_join' chan_sip.o(.text+0x16418): undefined reference to `ast_log' chan_sip.o(.text+0x164b8): undefined reference to `ast_log' collect2: ld returned 1 exit status make: *** [chan_sip.so] Error 1 - Original Message - From: listas iPfone [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 09, 2003 2:10 AM Subject: Re: [Asterisk-Users] Asterisk behind NAT How to do it. Hi The version 1.260 of chan_sip.c already have that patch?: http://bugs.digium.com/file_download.php?file_id=430type=bug thanks! Miklos - Original Message - From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, November 28, 2003 2:10 AM Subject: [Asterisk-Users] Asterisk behind NAT How to do it. Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c version 1.249 (the version the patch was written for) and the latest as of today 1.258. Both work great. Open ports 5060 and your RTP range (found in /etc/asterisk/rtp.conf). Default is 1 - 2 Forward ports 5060 and your RTP range to your internal Asterisk box. For your sip.conf, you need to add three lines: ; sip.conf snippet [general] port=5060 ; make sure you have this line :) inside_net=192.168.1.100; this is the internal ip address of the; asterisk server inside_mask=255.255.255.0 ; internal ip mask. /24 as this example outside_addr=216.239.33.100 ; this can also be a FQDN! ie. ; my.domain.com ; ... plus whatever else you have in your sip.conf Download the patch at: http://bugs.digium.com/file_download.php?file_id=430type=bug Either update your Asterisk or verify you have at least version 1.249 of chan_sip.c: cd /usr/src/asterisk/channels/ cvs status chan_sip.c === File: chan_sip.c
RE: [Asterisk-Users] Cisco 79xx Ringtones
Customizing the Cisco SIP IP Phone Ring Types The Cisco SIP IP phone ships with two ring types: Chirp1 and Chirp2. By default, your ring type options will be those two choices. However, using the RINGLIST.DAT file, you can customize the ring types that are available to the Cisco SIP IP phone users. Step 1 Create a pulse code modulation (PCM) file of the desired ring types and store the PCM files in the root directory of your TFTP server. PCM files must contain no header information and comply with the following format guidelines: 8000 Hz sampling rate 8 bits per sample ulaw compression Step 2 Using a ASCII editor, open the RINGLIST.DAT file and for each of the ring types you are adding, specify the name as you want it to display on the Ring Type menu, press Tab, and then specify the filename of the ring type. For example, the format of a pointer in your RINGLIST.DAT file should appear similar to the following: Ring Type 1 ringer1.pcm Step 3 After defining pointers for each of the ring types you are adding, save your modifications and close the RINGLIST.DAT file. http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_administr ation_guide_chapter09186a0080087511.html#1042487 -C www.ntfs.org -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adthrawn Sent: 11 January 2004 16:27 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 79xx Ringtones Hi, I'm after two very specific ringtones for the 79xx's... A dog barking, and a horse either galloping or neighing. I've tried making the sounds, but for some bizarre reason they're not working. I used to make quite a few ringtones for the 79xx's, but I seem to have forgotten how to do it! And to top things off, I can't even find the documentation on Cisco's site for making new ringtones. I do recall, you had to set the sample length to a divisible, something like 800? And there was a maximum sample length too... Best, Ad. ___ 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] OT: Anyone going to Open Source Telephony Summit in Geilenkirchen from North Germany?
Thanks for the info. I would like to go. Is it in German or English? I only speak English. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peer Oliver schmidt Sent: 04 January 2004 18:10 To: Asterisk User List Subject: [Asterisk-Users] OT: Anyone going to Open Source Telephony Summit in Geilenkirchen from North Germany? Hello, anyone from northern germany planning to go to http://www.guug.de/veranstaltungen/telephony-summit-2004/ If yes, could you contact me off list. Maybe we can save some money by car-pooling?! -- Best regards Peer Oliver Schmidt the internet company ___ 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] Asterisk behind NAT How to do it.
Hi I have SIP working on NAT using X-lite phones. On my Cisco 827H ADSL router I forwarded ports 5060, 16394, 16384 to my * (10.1.0.0). 16394,16384 being RTP. In X-lite set the RTP port to use 16394 instead of the default 8000. Works great over the internet. Didn't need patches or anything else. I hope that helps you. -C www.ntfs.org -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Balaji NJL Sent: 27 December 2003 08:34 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk behind NAT How to do it. Hi All, i tried to apply this patch and i got the following error. The chan_sip.c version i hv is 1.265 hv any one tried this patch on this latest chan_sip version. thanks, -B chan_sip.o: In function `load_module': chan_sip.o(.text+0x15ebf): undefined reference to `ast_rtp_proto_register' chan_sip.o(.text+0x15ee0): undefined reference to `ast_register_application' chan_sip.o: In function `delete_users': chan_sip.o(.text+0x15fc1): undefined reference to `ast_free_ha' chan_sip.o(.text+0x1604d): undefined reference to `ast_sched_del' chan_sip.o: In function `prune_peers': chan_sip.o(.text+0x16167): undefined reference to `ast_sched_del' chan_sip.o(.text+0x1618d): undefined reference to `ast_sched_del' chan_sip.o: In function `unload_module': chan_sip.o(.text+0x162bd): undefined reference to `ast_channel_unregister' chan_sip.o(.text+0x162ce): undefined reference to `ast_unregister_application' chan_sip.o(.text+0x16337): undefined reference to `ast_softhangup' chan_sip.o(.text+0x1636c): undefined reference to `ast_log' chan_sip.o(.text+0x163ab): undefined reference to `pthread_cancel' chan_sip.o(.text+0x163be): undefined reference to `pthread_kill' chan_sip.o(.text+0x163d1): undefined reference to `pthread_join' chan_sip.o(.text+0x16418): undefined reference to `ast_log' chan_sip.o(.text+0x164b8): undefined reference to `ast_log' collect2: ld returned 1 exit status make: *** [chan_sip.so] Error 1 - Original Message - From: listas iPfone [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 09, 2003 2:10 AM Subject: Re: [Asterisk-Users] Asterisk behind NAT How to do it. Hi The version 1.260 of chan_sip.c already have that patch?: http://bugs.digium.com/file_download.php?file_id=430type=bug thanks! Miklos - Original Message - From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, November 28, 2003 2:10 AM Subject: [Asterisk-Users] Asterisk behind NAT How to do it. Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c version 1.249 (the version the patch was written for) and the latest as of today 1.258. Both work great. Open ports 5060 and your RTP range (found in /etc/asterisk/rtp.conf). Default is 1 - 2 Forward ports 5060 and your RTP range to your internal Asterisk box. For your sip.conf, you need to add three lines: ; sip.conf snippet [general] port=5060 ; make sure you have this line :) inside_net=192.168.1.100; this is the internal ip address of the; asterisk server inside_mask=255.255.255.0 ; internal ip mask. /24 as this example outside_addr=216.239.33.100 ; this can also be a FQDN! ie. ; my.domain.com ; ... plus whatever else you have in your sip.conf Download the patch at: http://bugs.digium.com/file_download.php?file_id=430type=bug Either update your Asterisk or verify you have at least version 1.249 of chan_sip.c: cd /usr/src/asterisk/channels/ cvs status chan_sip.c === File: chan_sip.cStatus: Locally Modified Working revision:1.258 Repository revision: 1.258 /usr/cvsroot/asterisk/channels/chan_sip.c,v While in pwd /usr/src/asterisk/channels/ patch -p0 /path/to/patch Nothing should fail. cd /usr/src/asterisk/ make cp /usr/src/asterisk/channels/chan_sip.so /usr/lib/asterisk/modules/ Restart your Asterisk and try it. If you want to call a NAT'd Asterisk box, my Free World Dialup number is 18924. Currently online. -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com ___ 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 __ Do you
RE: [Asterisk-Users] MSN to GS - Call drops in 10 secs
Balaji, I also have the same issue. Works fine on any phone except GS for me. After a bit of research I found a post saying set the phone to offer only one codec set. It looks like we have to set the phone to use one codec GSM I am concerned that you cant use passwords when logging in to * using Messenger. Craig. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Balaji NJL Sent: 23 December 2003 23:04 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MSN to GS - Call drops in 10 secs resending. Can anyone help me in trying to understand what would be the problem. appreciate ur time. i need to get this working. thanks a lot, -B - Original Message - From: Balaji NJL To: [EMAIL PROTECTED] Sent: Monday, December 22, 2003 8:15 PM Subject: [Asterisk-Users] MSN to GS - Call drops in 10 secs Hi All, i dont know what changes i made recently but i am unable to hold the call for more 10 secs between MSN and GS. PSTN to MSN and PSTN to GS and vice versa works fine. i am not behind NAT.Also MSN to MSN works fine too. my SIP details [general] port = 5060 bindaddr = 0.0.0.0 context = bogon-calls ;context = default disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm ;My SIP phone - GS [2000] type=friend username=2000 secret=qweqwe host=dynamic context=from-sip mailbox=2000 dtmfmode=inband ;MSN Msgr [2002] type=friend host=dynamic insecure=yes dtmfmode=inband ;dtmfmode=rfc2833 context=from-sip mailbox=2002 ;auth=plaintext i did a SIP trace it says Format=UKN CSeq=BYE thanks for the help, -Balaji Do you Yahoo!? Yahoo! Photos - Get your photo on the big screen in Times Square Do you Yahoo!? Yahoo! Photos - Get your photo on the big screen in Times Square
[Asterisk-Users] MSN messenger and *
Sorry for the late reply. I try port 5060 and it just knocks me back straight away, I cant see it even try to authenticate in the CLI. X-lite works both inside the LAN and outside using SIP. Messenger version = 4.7 John I will try your suggestion with sip.conf thanks for the help. I notice a few differences, I seem to be missing some bits.. Its like it is trying to authenticate with the Linux box and not asterisk. Sip.conf [general] port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind to context=sip ; Default for incoming calls allow=ulaw allow=alaw allow=gsm allow=ilbc [3001] type=friend username=3001 fromuser=Craig1 secret=secret host=dynamic mailbox=3001 context=sip dtmfmode=info I found 3 guides and each one seems to be a bit different and use different ports. I am using the X100P, it is a home system, to reduce call charges for my family overseas. If I can get Messengger working it will be easier to talk them through the setup.
[Asterisk-Users] MSN messenger and *
I have read the guides on using Messenger to connect via SIP. I just cant get it to connect, even inside the LAN. I enter local ip address:5036, it trys to sign in, but times out and says Service Unavailable. Do I need anything extra in my configs for Messenger to work? Have * admins managed to get this to work? Any help welcome. Thanks
