similar to: canreinvite and codec negotations... and NAT

Okay, now on to my problem.. I have people who will be using ulaw, and I have people who will be using g729.. I want to set it up so that canreinivte will work.. I have a single cisco gateway.. Asterisks isn't handling the negotation between the 2 devices very well.. For example.. [gateway] type=friend host=1.2.3.4 canreinvite=yes qualify=200 dtmfmode=rfc2833 context=default disallow=all

I'm stuck running a old copy of asterisk because something strange is going on in later versions of the CVS.. When I call in from a PSTN into my cisco 2610XM gateway which then routes the call to my asterisk box via sip.. Asterisk can no longer process DTMF tones generated by the calling party. This affects DISA, prompts and menus.. Has anyone else had this problem?? and use.. I DO have

The lastest CVS's versions (both stable and head), the 'r' option in app_dial doesn't work with SIP and Re-invites. I've heard reports that it's not working with IAX2 either.. I'm using Cisco gateway and cisco ATA's and I am doing re-invites, and it's worked up till this point.. What's going on? Thanks, Billy

I've been getting this Warning message for a while now.. Apr 27 13:56:45 WARNING[1142106560]: chan_sip.c:5775 sipsock_read: Recv error: Resource temporarily unavailable and from what I can tell, this warning coinsides with a dropped call.. I'm running Cisco Gateways with Cisco ATA's (running 3.1 firmware) and I am doing Re-invites with NAT & STUN (and in some cases RTP aware

I'm new to Asterisk, but, Managed to get it working for outound calls from my ATA --> Asterisk --> Cisco 2620 using SIP. However, I'm having problems with Inbound calls from the Cisco.. Cisco 2620 --> Asterisk --> ATA .. In fact, voice mail won't even work.. This is a snippet of what I'm getting when I try to call the ATA -- Executing

Dear User, Anybody could dial these sip uri : sip:info@nxs.yi.org (french) sip:music@nxs.yi.org (music 60s) sip:support@nxs.yi.org (french) Thanks for help ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur

Hi, I have configured asterisk now with ENUM lookups which are working really perfect. Now I want to play a small soundfile before dial the number to inform the caller which protocl is used (SIP, IAX2 or ISDN). How can I do this? With Playback it doesn't seems to work: [iax2-sipport-out] ; with leading 3 using IAX-sipport exten => s,1,NoOp(Dialing ${DIALSTR} with iax2-sipport-out) exten

Tzafrir, How did you set sip:tzafrir@local.xorcom.com I use ser----asterisk look at my sip.conf and extensions.conf Regards Harry //////////////////////////////////////////////////// [general] context=sip realm=nxs.yi.org bindport=5050 bindaddr=nxs.yi.org srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=1000 allow=all musicclass=default language=fr insecure=very allowguest=yes

Tzafrir, How did you set sip:tzafrir@local.xorcom.com I use ser----asterisk look at my sip.conf and extensions.conf Regards Harry //////////////////////////////////////////////////// [general] context=sip realm=nxs.yi.org bindport=5050 bindaddr=nxs.yi.org srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=1000 allow=all musicclass=default language=fr insecure=very allowguest=yes

Hello, I posted a lot of mails may be asterisk is not able to accept sip calls from internet !? My english is not fluent i try my best ! My problem I use ser+asterisk. For local calls there are no problem (PSTN or IP) Now i wish to receive calls from other internet domain but asterisk ask for authentication 407. IS IT possible to Disable authentication for incoming calls to my sip uri ?

I'm trying to connect a Cisco 7942 to my Asterisk box. I have a 7960 and 7912 currently connected and functioning. I'm trying to use the recommendations from here: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP I have created a "XMLDefault.cnf.xml" and it took the latest image but the phone states it's unprovisioned? Any

continuing discussions of 79xx issues. i've seen referenced and am experiencing difficulty getting a 7970 to work behind NAT to a public asterisk server. i am successful with 7960s. 1. SIP load is 70.8-3-3SR2S 2. config works fine if 7970 is connecting to an asterisk server a local LAN (same subnet) 3. when debugging it in a NAT'd environment I see the register and

Hi, I have been using Cisco 7960's with Asterisk for years. I am trying get a 7961 working and have a problem. In my configuration, not all of my line appearances register to the same Asterisk SIP server. I have an Asterisk server at home and another at work. My Line 1 button registers to the home server and my Line 2 button registers to the work server. This has worked for years

On Friday 13 June 2008 22:46:41 Mark McLoughlin wrote: > lguest (in rusty's use-tun-ringfd patch) assumes that the > guest has updated its feature bits before setting its status > to VIRTIO_CONFIG_S_DRIVER_OK. > > That's pretty reasonable, so let's make it so. Applied. That's a bug, I'll send to Linus immediately (I screwed up in the feature changes which are

From: Mark McLoughlin <markmc at redhat.com> lguest (in rusty's use-tun-ringfd patch) assumes that the guest has updated its feature bits before setting its status to VIRTIO_CONFIG_S_DRIVER_OK. That's pretty reasonable, so let's make it so. Signed-off-by: Mark McLoughlin <markmc at redhat.com> Signed-off-by: Rusty Russell <rusty at rustcorp.com.au> ---

On Friday 13 June 2008 22:46:41 Mark McLoughlin wrote: > lguest (in rusty's use-tun-ringfd patch) assumes that the > guest has updated its feature bits before setting its status > to VIRTIO_CONFIG_S_DRIVER_OK. > > That's pretty reasonable, so let's make it so. Applied. That's a bug, I'll send to Linus immediately (I screwed up in the feature changes which are

From: Mark McLoughlin <markmc at redhat.com> lguest (in rusty's use-tun-ringfd patch) assumes that the guest has updated its feature bits before setting its status to VIRTIO_CONFIG_S_DRIVER_OK. That's pretty reasonable, so let's make it so. Signed-off-by: Mark McLoughlin <markmc at redhat.com> Signed-off-by: Rusty Russell <rusty at rustcorp.com.au> ---

Hi all, I've compiled 3.0.7 on a test box which also has an LDAP server running on it. This is the first time for me trying to use Samba with LDAP. I copied the IdealX scripts into /usr/local/sbin and edited the _config.pm file. It is attached at the bottom, stripped of comments. I then edited my smb.conf to the effect of the following: passdb backend = ldapsam:ldap://localhost ldap

Hi, I am trying to set up a linux box with 5 N/W cards of which one is 10/100/1000T and the others are 10/100T. I connected all the cards and turned on the machine. I wanted to force eth0 to be the 1000T cards but the cards get allotted eth0 to eth4 randomly. Is there some way wherein I can force my 1000T card to be eth0. Also I want to turn off auto negotation & flow control on the 1000T

Hello G?tz, could you do a closer look at your smbd (maybe with truss or strace) to see what exactly happens? or maybe it would be enough to watch traffic between your client and server (and traffic between your server and nameserver, domaincotroler (if involved)) best regards, chris -----Urspr?ngliche Nachricht----- Von: samba-bounces+christian.masopust=siemens.com@lists.samba.org