search for: sipdebug

...is dependent on implementors? The problem is chan_sip.c cannot correctly translate received DTMF digits, especially #,*. At least with my Antek EGW-804 gateway. Looking into chan_sip.c, I found this code: line 3982 if (p->owner) { if (strlen(buf)) { if (sipdebug) ast_verbose("DTMF received: '%c'\n", buf[0]); event = atoi(buf); << WHY? if (event < 10) { resp = '0' + event; } else if (event...

...8B393A24F60@192.168.1.24 CSeq: 30931 ACK Max-Forwards: 70 Content-Length: 0 Here is my default in SIP.conf. Each SIP config has canreinvite=no [general] disallow=all allow=gsm allow=ulaw nat=no canreinvite=no externip=(real external IP is here) localnet=192.168.1.195/255.255.255.0 srvlookup=yes sipdebug=yes I have tried nat=no and nat=yes -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051017/72472e87/attachment.htm

...xt part -------------- [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes domain=proxy.myhostname disallow=all allow=alaw sipdebug = yes recordhistory=yes dumphistory=yes register => <authstuff>@sip.externalpeer.com externhost=proxy.myhostname localnet=192.168.0.0/255.255.0.0 localnet=10.0.0.0/255.0.0.0 localnet=172.16.0.0/12 localnet=169.254.0.0/255...

...= no localnet = internal subnet maxcallbitrate = 384 maxexpiry = 3600 minexpiry = 60 mohinterpret = default nat = yes notifyringing = yes pedantic = no progressinband = never promiscredir = no realm = asterisk recordhistory = no registerattempts = 0 registertimeout = 20 relaxdtmf = no sendrpid = no sipdebug = no t1min = 100 t38pt_udptl = no tos_audio = none tos_sip = none tos_video = none trustrpid = no useragent = Asterisk PBX usereqphone = no videosupport = no disallow = all allow = ulaw,gsm subscribecontext = device-hints register => 6087294351:<sip password>@sip.broadvoice.com [trunk_1]...

...uth section to is 63e8aedc77335879c93123055d21211d Would this value match what chan_sip would pass as the md5 credentials? Our sip.conf looks like the following... [general] context = NECTEST bindaddr = 0.0.0.0 bindport = 5060 websocket_enabled = false srvlookup = no allowguest = yes debug = yes sipdebug = yes defaultexpiry = 480 deny = 0.0.0.0/24 permit = 10.100.102.0/24 permit = 192.168.9.0/24 canreinvite = yes callcounter = yes register = 3016:3016 at 10.100.102.82:5060/3016 [3016] type = friend qualify = no nat = no host = 10.100.102.82:5060 defaultuser = 3016 secret = 3016 incominglimit = 24...

Hallo Group! My Name is Guenther Sohler and I registred to this group, because I think asterisk could be interesting for me. I have got a small server at home running linux. It does NAT and a Firewall. There is an intranet with my home PC and a hardware SIP phone. This SIP phone registers at mujtelefon.cz Now I got another account at sipgate.at My idea is following: I want to be reachable at

This may be slight off topic, but perhaps it has relevance: My iconnecthere account no longer works for "inbound" calls through NAT using the standard configuration that they provide on their website. I have sent them a message, but I believe it will be flushed down the toilet by the first-tier support people. When I call my iconnect number, it goes directly to voicemail. There

Is there a way to configure the old SIP channel to stay in sip set debug all the time, without human intervention and also at boot time, by default? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20181205/d0ee9297/attachment.html>

...isabled and SIP port is set to 5061. As part of the asterisk configuration in 10.0.0.1, the following entries have been made in sip.conf: [general] context=default udpbindaddr=0.0.0.0 bindport=5060 srvlookup=no language=en contactpermit=127.0.0.1/255.255.255.0 contactpermit=10.0.0.2/255.255.255.0 sipdebug=yes allowsubscribe=no localnet=10.0.0.1/255.255.255.0 localnet=10.0.0.2/255.255.255.0 nat=never allowexternaldomains=no domain=10.0.0.1 matchexterniplocally=yes autodomain=yes directmedia=yes disallow=all allow=gsm allow=ulaw allow=alaw ;entry for phones [100] type=friend context=phones host=dynam...

...allow=ulaw jitterbuffer=yes maxjitterbuffer=1500 ;allow=ilbc ;musicclass=default ;language=en ;relaxdtmf=yes rtptimeout=60 ;rtpholdtimeout=300 ;trustrpid = no ;sendrpid = yes ;progressinband=never ;useragent=Asterisk PBX ;promiscredir = no ;usereqphone = no dtmfmode = rfc2833 ;compactheaders = yes ;sipdebug = yes ;subscribecontext = default ;notifyringing = yes And these are the extensions: [xxxx] type=friend host=dynamic dtmfmode=rfc2833 username=xxxx secret=xxxx [xxxx2] type=friend host=dynamic dtmfmode=rfc2833 username=xxxx secret=xxxx As you can see I put the jitterbuffer, maxjitterbu...

Can you also post information such as: Type of phone (model Number would be idela) How is it conencted, SIP, ZAP, IAX, Channel Bank. Corresponding config files would also help. Help us help you. >>-----Original Message----- >>From: asterisk-users-bounces@lists.digium.com >>[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of >>Paul A Brown >>Sent:

...alify = yes qualifysmoothing = yes realm = 206.b.c.d ; realm = metaswitch regcontext = test secret = metaswitch sipdebug = yes type = friend ; type = peer ; type = user username = metaswitch Here's the console SIP debug messages: <-- SIP read from 172.b....

Hi, I still have the same problem trying to configure ITSP failover in extensions.conf for a connected PRI. Any comments thoughts or direction would be greatly appreciated. I sympathize with wanting inbound DID failover. If we have a client with multiple DIDs we will spread them across two or three ITSPs so that all inbound connectivity will not be lost if one of them has an issue. I

...r REDIR to non-local SIP address > ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains > dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 > ;compactheaders = yes ; send compact sip headers. > ;sipdebug = yes ; Turn on SIP debugging by default, from > ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests > ;notifyringing = yes ; Notify subscriptions on RINGING state > ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER...

...: ; info : SIP INFO messages ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw) ; auto : Use rfc2833 if offered, inband otherwise ;compactheaders = yes ; send compact sip headers. ;sipdebug = yes ; Turn on SIP debugging by default, from ; the moment the channel loads this configuration ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests ; Useful to limit subscriptions to local...

Trying to get SIP realtime working here... I'm connected to the database... *CLI> realtime mysql status Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds. I can get information for the extension in question... *CLI> realtime load sipusers name 2944093 Column Name Column Value

What models? Jeff Noxon (jeff-asterisk at planetfall.com) wrote*: > >I just purchased a bunch of Nortel Meridian POTS phones that support >intercom on the 3rd pair. I intend to get it working with Asterisk. >The phones support MWI, have a 3-line display, callerID, call waiting >callerID, 2 lines...very nice. > >On Thu, Feb 27, 2003 at 01:07:19AM -1000, James H. Thompson