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Hi, I have been out of touch with asterisk for quit some time and needed some recommendations. I am looking for SIP hardphone that works well with asterisk server. Pls suggest. cheers /ag -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110212/ccd9d985/attachment.htm>

Hi, I'm running Asterisk 1.2.10 on gentoo linux and facing strange kind of issue. 1. chan_sip.so takes about 10 secs to load up when asterisk starts. 2. When I dialout using SIP it takes 20 secs to output " -- Called SIP 123@1.1.1.1" and get ring back from B party... Is there any config that I can check to reduce both delays? -ag

Hi, Anyone who has deployed Cisco 7945G phone with asterisk, kindly share your experience. /ag -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110216/3405b087/attachment.htm>

Hi, I have made a fresh install of asterisk-1.6.2.10 and when I register my soft phone it gives following error. Rest are default configurations. 32.454370 MY_IP -> ASTERISK_SERVER_IP SIP Request: REGISTER sip:ASTERISK_SERVER_IP 32.454505 67.19.43.202 -> MY_IP SIP Status: 401 Unauthorized (0 bindings) 36.454814 MY_IP -> ASTERISK_SERVER_IP SIP Request: REGISTER

Hi, I have downloaded iaxcomm version iaxcomm-lin-1.0rc3, when I try to execute it it gives following error. # ./iaxcomm Error wxWindows Fatal Error : Couldn't Initialize IAX Client . any idea what's going wrong ? -ag

Hi, I would like to know what is "Asterisk Dynamic Loader". Let me explain what I'm about to ask. I have three Asterisk servers running my in-house built app_xyz.so application. Now what I do to save time is compile application on one server and scp app_xyz.so on rest of servers. All servers have same OS, H/W specs. Today I checked the logs and observed that at the time when

Hi, It's really weired issue,I'm facing with asterisk-1.2.10 version. I see SIP call sessions stuck in asterisk for hours and then somehow get released. There happens to be an issue with BYE/CANCEL release msgs b/w sip entities. Has anyone faced this issue before also rtptimeout option given in sip.conf is not helping out. Any suggestions? -AG x post to *-dev, *-users

Hi, Just need to confirm whether dial() command provided options h: Allow the callee to hang up by dialing * H: Allow the caller to hang up by dialing * work for SIP channels as well ? -ag

Hi, Let me explain what I'm looking for a solution using asterisk. I have one third party SIP based server (A) and on Asterisk server (B). 1. Extension-1 --> Server A calls Server B. 2. Server B does some processing and calls/sends back to Server A ---> Extension-2 3. SIP session has been established b/w two Extension-1 and Extension-2. Now is there any config that I can do in

Hi, I have installed asterisk real time and sip buddies information is being stored in DB. Now I have a question, Asterisk Realtime Server -A Third party SIP server-B Question: Is there any configuration in * RT that it can register with defined sip user on Server-B I was only able to find sip users information in DB not about user registration on other server. -ag -------------- next part

Hi, I have a scenario that * Server A ( behaving as client) has sip peers, P1, P2, P3 with different contexts. Peers register to another * or any other SIP server. Using realtime * I am able to create a peer entry in sip buddies table and a register statement in sip.conf on client side and it's respected entry in SIP server. Peer is being registered... Server A receives and sends back the

Hi, I have register a sip user to sip server. I can see after registration * is sending periodic "SIP Request: OPTIONS" messages to server. but it's not getting back any response that should be SIP 200/OK as the documents say. 3130.299707 192.168.2.113 -> 58.ab.cd.ef SIP Request: OPTIONS sip: sipserver.net 3131.299513 192.168.2.113 -> 58.ab.cd.ef SIP Request: OPTIONS sip:

Hi, I am trying to install asterisk-1.2.34 but facing following issue. I have gone through it and found that there are files in /usr/lib libpq.a libpq.so libpq.so.4 libpq.so.4.1 make[1]: Entering directory `/usr/src/asterisk-1.2.34/cdr' gcc -shared -Xlinker -x -o cdr_pgsql.so cdr_pgsql.o -lpq -lz -L/usr/lib/pgsql /usr/bin/ld: skipping incompatible /usr/lib/libpq.so when

Hi, I am using codec g729 on two asterisk machines, but when call is forwarded from asterisk server 1 (32bit) to server 2(64bit). server 1 outputs following error and there is no audio. Also the IVRs being played have choppy voice. "Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = '')" It is running fine when codec gsm is in RTP traffic. Also I

Hi, Can any one please guide how do I handle the SIP phone-context URI parameter. I got following traces.. 9.191690 IP_A -> IP_B SIP/SDP Request: INVITE sip:12599;phone-context=private at IP_B:5060;user=phone, with session description 9.191942 IP_B -> IP_A SIP Status: 404 Not Found 9.195656 IP_A -> IP_B SIP Request: ACK sip:12599;phone-context=private at

Hi, I'm trying to call a SIP server while providing the SIP server username/password in dial string but it's not working ... Dial(SIP/gs102:test at 192.168.2.81); User on sip server (192.168.2.81): [gs102] disallow=all allow=ulaw allow=alaw type=friend username=gs102 secret=test host=dynamic dtmfmode=inband defaultip=192.168.2.1 qualify=1000 mailbox=102 context=context-gs102

Hi, I'm using zaptel-1.2.22.1 with asterisk-1.2.10 and following steps to make zaptel working... OS is gentoo linux 2006.1 Hardware: --------- 0000:05:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 8085:0003 Flags: bus master, medium devsel, latency 32, IRQ 22 I/O ports at b400 Memory at ff900000