sr-users September 2006

sr-users@lists.kamailio.org

184 participants
436 discussions

by mukesh

Hi friends, I have successfully installed SER with MySQL support on my linux machine (having public ip) and have added to users leo and mukesh. I have exported SIP_DOMAIN as # export SIP_DOMAIN=ims.myserver.com and added "export SIP_DOMAIN=ims.myserver.com" to /etc/profile as well First Problem --------------------------- If I try below... # /usr/local/sbin/serctl ping sip:mukesh@ims.myserver.com [1]+ Done cat <$path | filter_fl 500 Below is error log extracted from /var/log/messages Sep 15 10:37:30 oradb1 ser: 2(3911) ERROR: mk_proxy: could not resolve hostname: "ims.myserver.com" Sep 15 10:37:30 oradb1 ser: 2(3911) ERROR: uri2proxy: bad host name in URI <sip:mukesh@ims.myserver.com> Sep 15 10:37:30 oradb1 ser: 2(3911) ERROR: uri2sock: Can't create a dst proxy Sep 15 10:37:30 oradb1 ser: 2(3911) ERROR: get_hfblock: send_sock failed Sep 15 10:37:30 oradb1 ser: 2(3911) ERROR: fifo_uac_error: no mem for hf block ----------------------------- Second Problem -------------- How can I configure my Nokia6680 as UA (It has SIP support)? any kind of help/suggestion is highly appreciated. Regards Mukesh Kumar

17 years, 10 months

[Serusers] which linux is best for SER

by vijay tiwari

i want to install SER on my linux machine can any one tell me the best which linux version suite SER 9.3 Vijay Tiwari Network Admin Sachitel Communications Noida(U.P.) M- 09871633711 --------------------------------- Get your email and more, right on the new Yahoo.com

17 years, 10 months

[Serusers] allow_trusted"trusted table"

by ravi reddy

Hi , When i tried to make a incoming call from pstn to SER its asking for proxy authentication required, I had entered the pstngw ip address in trusted table even then the same problem , is any thing else i have to do ;-) how i can cross this problem. Thank You. Regards, Ravi.

17 years, 10 months

[Users] mediaproxy problem

by Klaus Darilion

Hi! I'm using openser 1.0 and mediaproxy 1.4.2. I've detected a problem with reINVITEs: 1. cisco7960 calls eyebeam all fine 2. eyebeam puts call on hold all fine 3. eyebeam put call off hold problem: mediaproxy sends packets received from cisco back to cisco (instead of forwarding to eyebeam) I've reviewed the changelog of mediaproxy module and mediaproxy, but this bug fix is not mentioned. I want to avoid unnecessary updating - thus a question to the developers: Where might this bug be located? in the mediaproxy module or in the media proxy? thanks klaus

17 years, 10 months

[Serusers] ERROR: receive_msg: no mem for sip_msg

by Kristian Larsson

Hey, I'm having a problem with my sip proxy. "All of a sudden" I'm getting this error message: ERROR: receive_msg: no mem for sip_msg I'm not sure when it started but now as soon as I start it and I get some REGISTER messages it generates the error message. This is a lab machine, I have 5 entries in my subscriber table and not more in any other. I have no idea what triggered this. There's not too much memory on the machine I'm running it (only 256MB) but that shouldn't matter really, there's still quite a lot free. I've tried setting debug=9 but I don't get any more debugging info to work with. Any suggestions on how to proceed? Regards, Kristian. -- Kristian Larsson KLL-RIPE Network Engineer Net At Once [AS35706] email: kristian(a)netatonce.se irc: kll@ircnet phone: +46 470 592717 cell: +46 704 910401

17 years, 10 months

Re: [Users] sip trunking between carriers

by vallimamod abdullah

Hello Raviprakash, On 9/14/06, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> wrote: > Hi , > OpenSER is having authenticate feature... > For better Authentication and authorizing, USE Radius...... > Thank for your quick answer. I guess you are talking about the authentification module. My precise need is for proxy-to-proxy authentification as I can't sent back the "407 - Proxy Authentification Required" answer to the PSTN gateway. What I would like is something like this: pstn-gw === INVITE ===> local-openser === INVITE ===> remote-carrier <=== 407 === === INVITE ===> <=== 200 OK === <=== 200 OK === The UAC module can do this but the problem for me is that it can only accept one static (user,password,domain) tuple for all calls. And what I would like is to retrieve a specific tuple from db for each relayed call (depending on the From and R-URI for example.) Any idea on how to achieve this ? Regards, -vma .

17 years, 10 months

[Users] sip trunking between carriers

by vallimamod abdullah

Hello list, I have an installation with openser connected to a PSTN gateway and I am offering the possibility to my customers to rent a PSTN trunk to forward long distance calls to a remote SIP carrier via my proxy. My customer usually gets a sip normal account on from remote SIP carrier and my proxy must use to authentificate all invites relayed to this carrier. So my question is: how can I authentificate my proxy to the remote carrier with the credentials given by my customer ? Has anybody already implemented a similar solution ? I have checked the UAC module but it seems to give the ability to set only a static credential on FAILURE_ROUTE. Thanks, -vma .

17 years, 10 months

Re: [Users] Openser not able to place calls to sipxezphone, no port mentioned in Contact URI

by raviprakash sunkara

Hi vamsi, I'm 4M Hyderabad, first i need to why using SipXtapi, for me its new... I'm using only openSER, And what is SipXtapi, in VOIP protocal of SIP default having 5060-5064 port, and SIP-proxy forwards 1024-65000, I'm using OpeSER, is best routing calls in voip technologies On 9/14/06, Vamsi Pottangi <vamsipottangi(a)gmail.com> wrote: > > Hi Ravi, > I'm aware of the deafult port 5060, there is some stange problem with > sipxtapi accepting calls on 5060 port. > > I'm from Bangalore, how about you? > > > > On 9/14/06, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> > wrote: > > > > Hi Vasmi > > Nice to meet U.... > > U using the Openser as Proxy.... > > it forwards that ports only... > > I'm not using the SipXtapi , any how , > > sip .. forwards these port only.. > > check the configuring file of sipXtapi..... > > > > I'm using OpenSER , is the Best routing the voip calls... > > > by actually, where r u from > > > > > > > > On 9/14/06, Vamsi Pottangi <vamsipottangi(a)gmail.com > wrote: > > > > > > Hi bogdan and ravi, > > > > > > Thaks for your replies, I narrowed the problem to be sipXtapi related > > > and not with OpenSER. > > > > > > Regards, > > > ~Vamsi > > > > > > > > > On 9/14/06, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com > > > > wrote: > > > > > > > > Hello Vamsi > > > > > > > > When U using the Behind the NAT or inside, > > > > Default port no is 5060, > > > > For UAC or UAS's it may b/w include the RTP port 1024 to 65000 , > > > > openser is forwarding these ports, > > > > > > > > > > > > > > > > On 9/12/06, Vamsi Pottangi < vamsipottangi(a)gmail.com > wrote: > > > > > > > > > Hi All, > > > > > > > > > > Need help on resolving the incomign calls problems with > > > > > sipxezphone which openser as proxy > > > > > > > > > > sizezphone sends "Contact:" URI as > > > > > sip:<usernamename>@<ipadd>;LINEID=<lineid> rather than > > > > > sip:<username>@<ipadd>:<port> > > > > > For the same reason OpenSER is not able to route the incoming > > > > > calls to sipxezphone. > > > > > > > > > > How can I make the sipxezphone(sipxtapi) to send Contact URI with > > > > > port number mentioned? > > > > > > > > > > ~~~Contact(0xb60deb38)~~~ > > > > > domain : 'location' > > > > > aor : 'vamsi' > > > > > Contact : ' sip:vamsi@192.168.101.21;LINEID=a9bd64e174dc' > > > > > Expires : 232 > > > > > q : > > > > > Call-ID : '5dd7235bc92046fd79bb68788a7f5f55' > > > > > CSeq : 1 > > > > > User-Agent: 'sipXezPhone' > > > > > received : '' > > > > > Path : '' > > > > > State : CS_SYNC > > > > > Flags : 0 > > > > > Sock : 192.168.101.35:5060 (0x8145be8) > > > > > Methods : 4294967295 > > > > > next : (nil) > > > > > prev : (nil) > > > > > ~~~/Contact~~~~ > > > > > > > > > > > > > > > > > > > > When I try to do NAT fixing on OpenSER, it records the port from > > > > > which the REGISTER message had recieved and tries to send the INVITE message > > > > > to that port, in this case 1617 rather than 5060. Actually sipzexphone is > > > > > listening on 1617 and 5060 as well which can be found from the below netsta > > > > > -na output > > > > > > > > > > When I snooped the messages, I found that sipzezphone is sending > > > > > register message using port 1617, though the DEFAULT_PORT is 5060 > > > > > > > > > > ~~~Contact(0xb60deb38)~~~ > > > > > domain : 'location' > > > > > aor : 'vamsi' > > > > > Contact : ' sip:vamsi@192.168.101.21 ;LINEID=a9bd64e174dc' > > > > > Expires : 171 > > > > > q : > > > > > Call-ID : 'e1102f18dca4dcb81363d75c605632fa' > > > > > CSeq : 7 > > > > > User-Agent: '' > > > > > received : 'sip: 192.168.101.21:1617' > > > > > Path : '' > > > > > State : CS_SYNC > > > > > Flags : 0 > > > > > Sock : 192.168.101.35:5060 (0x8145be8) > > > > > Methods : 4294967295 > > > > > next : (nil) > > > > > prev : (nil) > > > > > ~~~/Contact~~~~ > > > > > > > > > > C:\Documents and Settings\Administrator>netstat -na > > > > > > > > > > Active Connections > > > > > > > > > > Proto Local Address Foreign Address State > > > > > TCP 0.0.0.0:135 0.0.0.0:0 LISTENING > > > > > TCP 0.0.0.0:445 0.0.0.0:0 LISTENING > > > > > TCP 0.0.0.0:2701 0.0.0.0:0 LISTENING > > > > > TCP 0.0.0.0:2702 0.0.0.0:0 LISTENING > > > > > TCP 0.0.0.0:3389 0.0.0.0:0 LISTENING > > > > > TCP 0.0.0.0:42510 0.0.0.0:0 LISTENING > > > > > TCP 127.0.0.1:19989 0.0.0.0:0 LISTENING > > > > > TCP 192.168.101.21:139 0.0.0.0:0 LISTENING > > > > > TCP 192.168.101.21:1501 192.168.101.35:22 ESTABLISHED > > > > > TCP 192.168.101.21:1554 192.168.101.21:1555 TIME_WAIT > > > > > TCP 192.168.101.21:1615 192.168.101.21:1616 > > > > > ESTABLISHED > > > > > TCP 192.168.101.21:1616 192.168.101.21:1615 ESTABLISHED > > > > > TCP 192.168.101.21:1621 192.168.101.35:5060 ESTABLISHED > > > > > TCP 192.168.101.21:5060 0.0.0.0:0 LISTENING > > > > > UDP 0.0.0.0:445 *:* > > > > > UDP 0.0.0.0:500 *:* > > > > > UDP 0.0.0.0:1036 *:* > > > > > UDP 0.0.0.0:1037 *:* > > > > > UDP 0.0.0.0:1100 *:* > > > > > UDP 0.0.0.0:1508 *:* > > > > > UDP 0.0.0.0:4500 *:* > > > > > UDP 0.0.0.0:4573 *:* > > > > > UDP 0.0.0.0:4588 *:* > > > > > UDP 0.0.0.0:6060 *:* > > > > > UDP 127.0.0.1:123 *:* > > > > > UDP 127.0.0.1:1038 *:* > > > > > UDP 127.0.0.1:1056 *:* > > > > > UDP 127.0.0.1:1075 *:* > > > > > UDP 127.0.0.1:1105 *:* > > > > > UDP 192.168.101.21:123 *:* > > > > > UDP 192.168.101.21:137 *:* > > > > > UDP 192.168.101.21:138 *:* > > > > > UDP 192.168.101.21:1617 *:* > > > > > UDP 192.168.101.21:1618 *:* > > > > > UDP 192.168.101.21:1619 *:* > > > > > UDP 192.168.101.21:1620 *:* > > > > > UDP 192.168.101.21:5060 *:* > > > > > > > > > > C:\Documents and Settings\Administrator> > > > > > > > > > > > > > _______________________________________________ > > > > Users mailing list > > > > Users(a)openser.org > > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > > > > > > -- > > > > Thanks and Regards with cheers > > > > Sunkara Ravi Prakash (Voip Developer) > > > > Hyperion Technology > > > > www.hyperion-tech.com > > > > +91-9985077535 > > > > > > > > > > > > > > > > -- > > Thanks and Regards with cheers > > Sunkara Ravi Prakash (Voip Developer) > > Hyperion Technology > > www.hyperion-tech.com > > +91-9985077535 > > > > -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

17 years, 10 months

[Users] Openser not able to place calls to sipxezphone, no port mentioned in Contact URI

by Vamsi Pottangi

Hi All, Need help on resolving the incomign calls problems with sipxezphone which openser as proxy sizezphone sends "Contact:" URI as sip:<usernamename>@<ipadd>;LINEID=<lineid> rather than sip:<username>@<ipadd>:<port> For the same reason OpenSER is not able to route the incoming calls to sipxezphone. How can I make the sipxezphone(sipxtapi) to send Contact URI with port number mentioned? ~~~Contact(0xb60deb38)~~~ domain : 'location' aor : 'vamsi' Contact : 'sip:vamsi@192.168.101.21;LINEID=a9bd64e174dc' Expires : 232 q : Call-ID : '5dd7235bc92046fd79bb68788a7f5f55' CSeq : 1 User-Agent: 'sipXezPhone' received : '' Path : '' State : CS_SYNC Flags : 0 Sock : 192.168.101.35:5060 (0x8145be8) Methods : 4294967295 next : (nil) prev : (nil) ~~~/Contact~~~~ When I try to do NAT fixing on OpenSER, it records the port from which the REGISTER message had recieved and tries to send the INVITE message to that port, in this case 1617 rather than 5060. Actually sipzexphone is listening on 1617 and 5060 as well which can be found from the below netsta -na output When I snooped the messages, I found that sipzezphone is sending register message using port 1617, though the DEFAULT_PORT is 5060 ~~~Contact(0xb60deb38)~~~ domain : 'location' aor : 'vamsi' Contact : 'sip:vamsi@192.168.101.21;LINEID=a9bd64e174dc' Expires : 171 q : Call-ID : 'e1102f18dca4dcb81363d75c605632fa' CSeq : 7 User-Agent: '' received : 'sip:192.168.101.21:1617' Path : '' State : CS_SYNC Flags : 0 Sock : 192.168.101.35:5060 (0x8145be8) Methods : 4294967295 next : (nil) prev : (nil) ~~~/Contact~~~~ C:\Documents and Settings\Administrator>netstat -na Active Connections Proto Local Address Foreign Address State TCP 0.0.0.0:135 0.0.0.0:0 LISTENING TCP 0.0.0.0:445 0.0.0.0:0 LISTENING TCP 0.0.0.0:2701 0.0.0.0:0 LISTENING TCP 0.0.0.0:2702 0.0.0.0:0 LISTENING TCP 0.0.0.0:3389 0.0.0.0:0 LISTENING TCP 0.0.0.0:42510 0.0.0.0:0 LISTENING TCP 127.0.0.1:19989 0.0.0.0:0 LISTENING TCP 192.168.101.21:139 0.0.0.0:0 LISTENING TCP 192.168.101.21:1501 192.168.101.35:22 ESTABLISHED TCP 192.168.101.21:1554 192.168.101.21:1555 TIME_WAIT TCP 192.168.101.21:1615 192.168.101.21:1616 ESTABLISHED TCP 192.168.101.21:1616 192.168.101.21:1615 ESTABLISHED TCP 192.168.101.21:1621 192.168.101.35:5060 ESTABLISHED TCP 192.168.101.21:5060 0.0.0.0:0 LISTENING UDP 0.0.0.0:445 *:* UDP 0.0.0.0:500 *:* UDP 0.0.0.0:1036 *:* UDP 0.0.0.0:1037 *:* UDP 0.0.0.0:1100 *:* UDP 0.0.0.0:1508 *:* UDP 0.0.0.0:4500 *:* UDP 0.0.0.0:4573 *:* UDP 0.0.0.0:4588 *:* UDP 0.0.0.0:6060 *:* UDP 127.0.0.1:123 *:* UDP 127.0.0.1:1038 *:* UDP 127.0.0.1:1056 *:* UDP 127.0.0.1:1075 *:* UDP 127.0.0.1:1105 *:* UDP 192.168.101.21:123 *:* UDP 192.168.101.21:137 *:* UDP 192.168.101.21:138 *:* UDP 192.168.101.21:1617 *:* UDP 192.168.101.21:1618 *:* UDP 192.168.101.21:1619 *:* UDP 192.168.101.21:1620 *:* UDP 192.168.101.21:5060 *:* C:\Documents and Settings\Administrator>

17 years, 10 months

[Serusers] (no subject)

by prabhakar g

hi everybody hey i am trying to installing ser+mysql+nathelper+rtpproxy i am using ser-0.9.6_src.tar. i have intsalled centos without any packages . i installed the ser-0.9.6_src.tar and then then i installed the mysql-databes, but mysql.so module file not loading i head that there a make file in source folder to to make changes before installing the ser. can anybody help abt this topic -- Prabhakar.g Network Engineer Northgate Technologies Banjarahils,Hyderabad-34

17 years, 10 months

← Newer
1
...
21
22
23
24
25
26
27
...
44
Older →

Jump to page:

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users September 2006