similar to: SIP re-INVITES problem

Could some please confirm that this behavior is incorrect. I am seeing issues where it appears that asterisk is not following the Record-route on it's reply messages. Please let me know if you require any other information. Thanks Example: xxx.yyy.154.243(PSTN-GW) <--sip--> xxx.yyy.77.23(Asterisk) <--sip--> xxx.yyy.91.74(SNOM or SER proxy) <--sip---->

Trouble getting chan_sip2 to compile below is what I have done -download acl.c.patch,acl.h.patch,chan2s_sip.c to /root/software cp /root/software/chan_sip2s.c /usr/src/asterisk/channels cd /usr/src/asterisk/ patch -p0 acl.c /root/software/acl.c.patch cd /usr/src/asterisk/include/asterisk patch -p0 acl.h /root/software/acl.h.patch - added the follow to /usr/src/asterisk/channels/Makefile

After setting the pridialplan=unknown I seeing the Called Number TON change to Unknown Number Type but not the Calling Number TON. Should both be following this parameter or not. If not is their another option to change the Calling Number TON? > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) > Ext: 1 Progress

SDP question if * recieves "a=rtpmap:103 telephone-event/8000" it shouldn't it send out the same "a=rtpmap:103 telephone-event/8000" to the other side of the connection? and not something like "a=rtpmap:101 telephone-event/8000"? Thanks

I download app_valetparking.c from http://www.loligo.com/asterisk/misc/apps/app_valetparking.c and followed the directions on http://www.loligo.com/asterisk/misc/apps/app_valetparking.README I am using asterisk-1.0.0 any suggestions [root@localhost asterisk]# astxs -install apps/app_valetparking.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude

Hi, I have the following situation I use asterisk as o gateway between networks. What is the reason for such response? What are the criteria for such evaluation? SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.129.74:5160;branch=z9hG4bK17c3.17db29e7.0;received= 192.168.129.74 Via: SIP/2.0/UDP 192.168.129.74;branch=z9hG4bK17c3.23083974.0 Via: SIP/2.0/UDP

Symptom: Phone after about 15mins will stop functioning Problem: DHCP lease renewed but default route dropped Fix: Assign a static ip and problem is resolved. Upgrade to new firmware once it is released It turn's out that these phones have a few issue in 1.0.3.81 firmware. The phone may stop transmitting packets if configured with DHCP, if DHCP is being provided by certain devices. Netopia

Hi, Has anyone had a problem setting up two registrations (on the same Asterisk server) on one Polycom phone? When the user tries to make a call on the 2nd line, it works fine. But when they try the first line, the CLI says:- Using INVITE request as basis request - 9f5fe9a5-215d0f3a-b2fbe6b7 at 192.168.1.138 Found peer client _202' <--- Which is incorrect, it should be client_201. And

I have been doing some testing and have found issue with certain devices and negotiating codecs in doing this I have noticed something that seems peculiar to me. It seems that including allow=all yields different results than having no disallow or allows in the sip.conf. Could someone please explain why that is true? Thanks -------------- next part -------------- An HTML attachment was

Just woundering if the intentend functionality of leastrecent and fewestcalls it to continually dial only the first chosen ext. in the queue. In other words if a memeber is logged into the queue but doesn't answer the call the call never moves on in my configuration from that ext. This could be really bad!!!! Thanks [support] announce-frequency=45 strategy=leastrecent music=default

Has any had any experiences with Watchguard Firebox 1000 and Asterisk. I have asterisk on public side and phones on the private side. I am able to get the phones to register and make outbound calls but the inbound calls are intermittent. I have NAT enable in asterisk and on the Cisco 7960. Any insight would be appreciated. Thanks

I am seeing an issue with getting certain sip devices to be recognized as defined SIP clients host= in the sip.conf and the only deference that I can find btw sources that work and don't work is that devices that send packets with an Initial Via header of themselves appears to work and pick the context correctly but those that don't have the Via just get dropped in the context of the

Hi David: In looking at your original post it is a bit difficult to ascertain exactly what your null hypothesis was. That is, you want to assess whether there is a treatment effect at time 3, but compared to what. I think your second post clears this up. You should refer to pages 224- 225 of Pinhiero and Bates for your answer. This shows how to specify contrasts. > -----Original Message-----

Hi everybody, I wanted to remind you to a) To request invites to the Xen Hackathon of you have not done so yet b) If you have an invite sign up at http://www.regonline.com/Register/Checkin.aspx?EventID=1211624 (I noticed that at least 5 people who have been granted an invite have not actually signed up) c) Note that about half of the available invites have gone d) Please also add

Hmmm... Folks, I beg you pardon, if I'm telling something which was said before, but actually I have not found this anywhere, neither on Voip-info.org or in several Asterisk's docs. So, here is the statement: If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them will ALWAYS go via Asterisk. I.e. Asterisk WILL NOT issue Re-INVITE even if: 1. Both UAs have

I currently have OpenSIPS set up with users and most of my call handling. OpenSIPS won't be able to handle things like Call Park, Hunt Groups, ACD, etc. So I want to send these types of requests to Asterisk. I also want to set Asterisk up as Multi Tenant. So my question is How can I send requests to Asterisk and have them funnel into the specific context for that specific Tenant? So if

I am using asterisk 1.4.18 (server A ) and have it store records in mysql database . One of my client uses predictive dialer ( asterisk 1.2.26 based and server B ) which makes many calls . B registers with A over sip and there is no nat involved If i re-invite rtp from server B to my carrier ( server A in between ) I saw many calls having duration of 0,1 or 2 seconds on server A's cdr but

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Many of my dialplan scenarios involve transferring incoming calls back out to other numbers. For reasons of call quality and bandwidth, I would like for the calls to be reinvite'd to bypass my server with the audio channel. What I am seeing is that my server does indeed send the reinvites, and I get OK responses, but the audio never stops passing through my server. I've been fooling

Hi All, This is probably a stupid question, but I'm trying to figure out if I Asterisk is in the middle of the media stream or not... Is there a command or something that indicates weather of not the two endpoints are talking directly? I am seeing messages such as : -- SIP/200-eb90 answered SIP/208-f0d6 -- Attempting native bridge of SIP/208-f0d6 and SIP/200-eb90 ...but not