similar to: Possible fix for grandstream outgoing

Here is the diff from chan_sip.c 15 days ago and 16 days ago. 15 days ago is the point outgoing calls made via grandstream budgetone stopped working. Any help on why it breaks? Any possible fix? /tmp# diff asterisk/channels/chan_sip.c asterisk.works/channels/chan_sip.c 289d288 < int capability; 3921,3922d3919 < p->capability = user->capability;

Greetings. Attempting to connect Asterisk to LDAP database using res_config_ldap module. While trying to register sip client (Ekiga softphone), according to slapd.log, asterisk connects to LDAP server, asks for some attributes to modify (they do exist, and asterisk user has all permissions to do that, etc). And then asterisk application just crashes. Without ldap (using just static users'

Hi, I'm working with asterisk 1.4.35 and found an issue regarding codecs negotiation when T38 is enabled (t38pt_udptl=yes). In particular if the INVITE sdp contains no allowed codec the call is not rejected with "488 - Not acceptable here" but it goes through and the 200 OK SDP is as follows: v=0 o=root 27285 27285 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0

Hi, I would like to prepare postlogin a script that allow imap connection to roundcube for all but restrict imap access for selected users. My question is that: Is possible in condition IF use IP addresses as range or with mask (because I've more than one web servers) ? My script: #!/bin/sh if [ "$IP" = "172.11.0.28" ] ; then printf "* [ALERT] Access allowed from

Thx, prips works as I expected, gr8 tool, not available in Gentoo repository but after compilation Dovecot doing what I wanted. Regards, Jack 2017-11-09 21:19 GMT+01:00 Gedalya <gedalya at gedalya.net>: > A bit clunky but perhaps you could find another command. > > https://packages.debian.org/stretch/netmask > > $ IP=172.11.0.28 > $ if [ "$(netmask -n $IP/24)"

non-video mode, it is never able to add video to the channel? Is this correct, or am I missing something? It looks as if the codec 'jointcapability' is calculated at the start of the call, and can never be added to (with exceptions for T.38 fax) as any SDP update is masked using the existing 'jointcapability' and knocks out the newly requested codec. Is that right? Thanks, Steve

A bit clunky but perhaps you could find another command. https://packages.debian.org/stretch/netmask $ IP=172.11.0.28 $ if [ "$(netmask -n $IP/24)" == "???? 172.11.0.0/24" ]; then echo OK; fi OK $ IP=172.12.0.11 $ if [ "$(netmask -n $IP/24)" == "???? 172.11.0.0/24" ]; then echo OK; fi $ Range: https://packages.debian.org/stretch/prips $ IP=172.11.0.28 $

Dovecot v.2.2.32 and I have configured two imap post-login scripts and it seems like after successfully login scripts are not closed (dovecot_node/imap imap-postlogin : multiple processes are running still) and after some times there are too many processes and the limit is reached (imap proces_limit 1500): 1) #!/bin/sh case $IP in 10.10.1[1-2][0-7].*) exec "$@" ;;

Hi, I'm can't make this work, I don't had very much problems configuring Linux to access the printer, but in solaris i don't know to do it. Can soneone help me i there any HOWTO... Thanks....

My legacy PBX accepts CID number, but not name. My old PRI vendor never sent the name, so there was never an issue. I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI - asterisk - PRI - Legacy. Any calls from asterisk (sip and iax extensions) which have callerID set, will not connect. The legacy PBX hangs up, but asterisk thinks that it is still ringing. I have added

Thanks Paradise, this seems to have worked a treat!!! I commented out the: exten => 110,hint,SIP/110 lines which were in extensions_additional.conf for each sip extension I had. This seems to have stopped the crashes which were previously 3-5 times a day, now: System uptime: 1 day, 18 hours, 10 minutes, 3 seconds Interestingly it had the knock on effect of fixing another problem I had

I am not having any luck placing out going calls I dial the number 08 82420173 ( our outside line ) But all I get is engaged signal and log this. Oct 14 08:40:14 DEBUG[16401]: File pbx_wilcalu.c, Line 65 (autodial): Entered Wil-Calu fd=20 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 548

Hi all. I'm trying to get a simple configuration working so I can later expand it to something more interesting. I'm using kphone to call an extension on the * server. When I try to connect, I get this error: DEBUG[81926]: File chan_sip.c, Line 3562 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on

Hello there! I'm working on some modifications on Asterisk to adapt it to our needs considering some particular demandings of the infraestructure we want to provide. Two of these modifications are: 1- A proprietary configuration driver that will communicate with a server that will be the source of information for the entire infraestructure; and, 2- A call control application that will be

I have just setup asterisk on a debian sarge box. I am running Asterisk 1.21 with AMP and chan_capi_cm 0.6.1 using a BT Speedway (AVM Fritz) ISDN card, connected to a BT ISDN2e line. Currently we have 6 extensions (SIP) configured all using CounterPath(Xten) eyebeam softphone. After many hours of Googling I have finally got it all setup and working. We can transfer calls internally and make and

The Asterisk Development Team has announced the third release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 10.0.0-rc3 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the

The Asterisk Development Team has announced the third release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 10.0.0-rc3 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the

The Asterisk Development Team has announced the release of Asterisk 1.8.14.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.14.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

I have a sipura 3000 that I am using just to send calls to my mac asterisk server. When you call the phone it rings, answers, and then goes right to a busy signal. Any ideas? Thanks for your help! Jane At the console in verbose mode I get: *CLI> DEBUG[8501248]: File chan_sip.c, Line 663 (create_addr): Setting NAT on RTP to 0 DEBUG[8501248]: File chan_sip.c, Line 554 (__sip_ack): Stopping

I've been looking at how our budgetone's have been failing and have found the following: A quick layout -- Latest CVS as of tonight. Sip phone behind NAT. * server with public IP address. -------from sip.conf for my phone: [1747xxxxxxx] username=xxxxx secret=xxxxx host=dynamic type=friend nat=yes ------- -------from the * log messages Nov 6 01:50:07 DEBUG[4101]: File chan_sip.c,